Jonathan Leong
2010-Aug-27 14:04 UTC
[asterisk-users] asterisk-users Digest, Vol 73, Issue 58
On 8/27/10, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote:> Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request at lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner at lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. CDR on Transfer... (Carlos Chavez) > 2. Re: Asterisk 1.6.1.17 ACK/BYE question (Trevor Benson) > 3. Re: Use of AGISIGHUP (Danny Nicholas) > 4. double DTMF digits (M S) > 5. Re: double DTMF digits (Andres) > 6. Re: Use of AGISIGHUP (Steve Edwards) > 7. Re: Use of AGISIGHUP (Danny Nicholas) > 8. Re: Use of AGISIGHUP (Steve Edwards) > 9. Re: double DTMF digits (Matt Desbiens) > 10. Asterisk 1.6 Displaying in Debug that it is playing a ulaw > file using BackGround() but no audio is heard from the phone > (Joe Wood) > 11. Re: double DTMF digits (M S) > 12. Re: Use of AGISIGHUP (Lee Archer) > 13. dynamic MeetMe, min. digits (Xavier) > 14. Re: dynamic MeetMe, min. digits (Doug Lytle) > 15. Re: dynamic MeetMe, min. digits (Xavier D.) > 16. music on hold in blind transfer (Tino) > 17. queue agent and blind transfer (Tino) > 18. Call Forwarding (Dan Journo) > 19. Re: music on hold in blind transfer (Paul Belanger) > 20. Re: Call Forwarding (Stefan Schmidt) > 21. Duplicate channel variables after transfer (Alex Hermann) > 22. Re: CDR on Transfer... (Andra?) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 26 Aug 2010 12:25:07 -0500 > From: Carlos Chavez <cursor at telecomabmex.com> > Subject: [asterisk-users] CDR on Transfer... > To: Asterisk <asterisk-users at lists.digium.com> > Message-ID: <1282843507.2830.13.camel at cursor.telecomabmex.com> > Content-Type: text/plain; charset="utf-8" > > I have searched for some time but I have not found an asnwer on how to > fix the CDR when a call is transferred. The problem is that if someone > dials a cell phone and then transfers the call to another extensi?n the > CDR for the cell call stops and there is no way to track that the call > was transferred so we can bill correctly. Many people have asked this > question but there is no answer, only a mention that it should be fixed > in 1.6 which it is not (at least on 1.6.2.11). > > Any tips oh how to correct this problem? A lot of customers give me > grief about this because they cannot properly bill people for their cell > calls. > > -- > Telecomunicaciones Abiertas de M?xico S.A. de C.V. > Carlos Ch?vez Prats > Director de Tecnolog?a > +52-55-91169161 ext 2001 > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 198 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/f970d9ee/attachment-0001.pgp > > ------------------------------ > > Message: 2 > Date: Thu, 26 Aug 2010 10:30:16 -0700 > From: Trevor Benson <tbenson at a-1networks.com> > Subject: Re: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <65F20266-3E46-4DCC-A17D-D181F8E4A6AE at a-1networks.com> > Content-Type: text/plain; charset="windows-1252" > > We have a box running 1.6.2.11 on CentOS 5 using the RPM's from the Digium > CentOS repository. We just left a 60 second voicemail on the system and had > the full audio as well in the inbox. Not sure how your SIP configuration > ties your SBC in, but native "users" created via users.conf and sip.conf > appears to be working for me. Wouldnt be able to test more without knowing > what settings you had between Asterisk and the SBC. > > > -- > Trevor Benson > dCAP, LPIC-1, CLA, Network+, MCP, CNA > A1 Networks - Network Engineer > DID (707)703-1041 > FAX (707)703-1983 > > > > > > > On Aug 26, 2010, at 8:47 AM, Steven C. Blair wrote: > >> >> As a test we built Asterisk v1.6.2.11 on a new server. This version of >> Asterisk exhibits the same behavior. From ngrep?s perspective we see an >> ACK followed immediately by a BYE message. The user hears the recording >> being played, begins to leave a message and is disconnected about 10 >> seconds into the call. >> >> >> >> From: asterisk-users-bounces at lists.digium.com >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven C. >> Blair >> Sent: Wednesday, August 25, 2010 2:08 PM >> To: asterisk-users at lists.digium.com >> Subject: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question >> >> >> We?re running Asterisk 1.6.1.17 for our campus voicemail server and >> Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are >> diverted to voicemail using a 302 redirect when the called party doesn?t >> answer. In this case the caller is able to hear the greetings and begin to >> leave a message only to have Asterisk terminate the call mid-recording. >> >> We?re uncertain why this is happening and this is where we are hoping you >> can help. In our environment the caller is any set on the PSTN. They call >> one of our IP phones which no one answers. Our proxy, SER, responds to the >> SBC with a 302 redirect and the call is diverted to Asterisk. The caller >> hears the unavailable greeting for 6-4050, begins to leave a message and >> is cut-off after about 10 seconds. In an ngrep trace we see Asterisk >> receive an ACK from the SBC and it immediately responds with a BYE message >> for that call. >> >> Has anyone else experienced this type of issue? >> >> >> --- >> >> ISC Networking & Telecommunications >> 3401 Walnut Street, Suite 221A >> Philadelphia, PA 19104 >> 215-573-8396 >> 215-898-9348 (fax) >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/59a22a5a/attachment-0001.htm > > ------------------------------ > > Message: 3 > Date: Thu, 26 Aug 2010 12:58:46 -0500 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] Use of AGISIGHUP > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <201008261730.o7QHULt4029526 at mail.debsinc.com> > Content-Type: text/plain; charset="us-ascii" > > Can you post the CLI output showing the hangup/script failure? > > > > _____ > > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Lee Archer > Sent: Thursday, August 26, 2010 11:39 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Use of AGISIGHUP > > > > Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but it > doesn't seem to be doing anything as the script is still exiting on a hangup > and not completing properly. I am using 1.4.35 and have tried various > combinations. Can anyone shed any light on this? > > Regards > > Lee > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/7a77e4e1/attachment-0001.htm > > ------------------------------ > > Message: 4 > Date: Thu, 26 Aug 2010 14:55:50 -0400 > From: M S <101mcs at gmail.com> > Subject: [asterisk-users] double DTMF digits > To: asterisk-users at lists.digium.com > Message-ID: > <AANLkTinoT4+HrPBHdKAx6WOU-vtyUPq3KcH9vRAE=E53 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > I've been getting complaints lately that callers to my IVR are pressing a > digit once but the system is responding as if they pressed it twice (once > for each of two consecutive menus). > I'm using an AGI script and logging all DTMF entries - and to the script, at > least, it looks like the digit is being pressed twice. The TN being called > is a VOIP number (provided by Flowroute) and being forwarded via SIP to my > asterisk 1.6.2.4 server. The dtmfmode is set to rfc28333 in sip.conf. > > The first time this happened, I figured the caller pressed the number twice > without realizing it. It's happening to too many people for that to be > plausible anymore. I also experienced it once myself, months ago, when I > entered my tn as 1234567890 and had it read back to me as 1122334455. > > Can anyone give me some pointers where to start troubleshooting? Can > overloading a system cause such an error? > > Thanks, > Mira > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/c9db8975/attachment-0001.htm > > ------------------------------ > > Message: 5 > Date: Thu, 26 Aug 2010 15:23:44 -0400 > From: Andres <andres at telesip.net> > Subject: Re: [asterisk-users] double DTMF digits > To: asterisk-users at lists.digium.com > Message-ID: <4C76BF40.20204 at telesip.net> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 8/26/2010 2:55 PM, M S wrote: >> Hi, >> >> I've been getting complaints lately that callers to my IVR are >> pressing a digit once but the system is responding as if they pressed >> it twice (once for each of two consecutive menus). >> I'm using an AGI script and logging all DTMF entries - and to the >> script, at least, it looks like the digit is being pressed twice. The >> TN being called is a VOIP number (provided by Flowroute) and being >> forwarded via SIP to my asterisk 1.6.2.4 server. The dtmfmode is set >> to rfc28333 in sip.conf. >> >> The first time this happened, I figured the caller pressed the number >> twice without realizing it. It's happening to too many people for >> that to be plausible anymore. I also experienced it once myself, >> months ago, when I entered my tn as 1234567890 and had it read back to >> me as 1122334455. >> >> Can anyone give me some pointers where to start troubleshooting? Can >> overloading a system cause such an error? >> >> Thanks, > I have seen this before. Upon careful analisys we saw that the far end > was sending the digits in RFC2833 plus SIP INFO (or Inband, I can't > remember). Thus Asterisk detected double digits. The solution was to > ask the remote end to only send RFC2833. > > Andres > http://www.telesip.net > > > > ------------------------------ > > Message: 6 > Date: Thu, 26 Aug 2010 12:41:25 -0700 (PDT) > From: Steve Edwards <asterisk.org at sedwards.com> > Subject: Re: [asterisk-users] Use of AGISIGHUP > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <alpine.DEB.2.00.1008261233160.25774 at localhost.localdomain> > Content-Type: text/plain; charset="iso-8859-7" > > On Thu, 26 Aug 2010, Lee Archer wrote: > >> Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but >> it doesn?t seem to be doing anything as the script is still exiting on a >> hangup and not completing properly.? I am using 1.4.35 and have tried >> various combinations.? Can anyone shed any light on this? > > I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's a > bad idea to protect lazy programmers :) > > Program defensively! > > Trap the HUP and do what is appropriate for your script -- even if that is > to ignore it. > > If the successful execution of your system depends on a setting, how long > will it take the next guy to figure out when the setting disappears > unexpectedly? > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > ------------------------------ > > Message: 7 > Date: Thu, 26 Aug 2010 14:52:53 -0500 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] Use of AGISIGHUP > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <201008261924.o7QJOREn030652 at mail.debsinc.com> > Content-Type: text/plain; charset="iso-8859-1" > >>From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Edwards >>Subject: Re: [asterisk-users] Use of AGISIGHUP > >>On Thu, 26 Aug 2010, Lee Archer wrote: > >>> Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but >>> it doesn?t seem to be doing anything as the script is still exiting on a >>> hangup and not completing properly.? I am using 1.4.35 and have tried >>> various combinations.? Can anyone shed any light on this? > >>I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's a > bad idea to protect lazy programmers :) > >>Program defensively! > >>Trap the HUP and do what is appropriate for your script -- even if that is > to ignore it. > >>If the successful execution of your system depends on a setting, how long > will it take the next guy to figure out when the setting disappears > unexpectedly? > > As usual, Steve is right. Here's a one-liner that should "fix" the problem > > local $SIG{HUP} = 'IGNORE'; > > Does that make me lazy? > > TIA. > > > > > ------------------------------ > > Message: 8 > Date: Thu, 26 Aug 2010 13:02:20 -0700 (PDT) > From: Steve Edwards <asterisk.org at sedwards.com> > Subject: Re: [asterisk-users] Use of AGISIGHUP > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <alpine.DEB.2.00.1008261257210.15301 at localhost.localdomain> > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > >>> On Thu, 26 Aug 2010, Lee Archer wrote: >> >>>> I am setting AGISIGHUP=no before running a Perl script via AGI but it >>>> doesn?t seem to be doing anything as the script is still exiting on a >>>> hangup and not completing properly. > >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve >> Edwards >> >>> I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's a >>> bad idea to protect lazy programmers :) > > On Thu, 26 Aug 2010, Danny Nicholas wrote: > >> Here's a one-liner that should "fix" the problem >> >> local $SIG{HUP} = 'IGNORE'; >> >> Does that make me lazy? > > Not at all. If that is the correct "response" for your program, it's > perfect: > > 1) The program will execute correctly on your system, my system, any > system regardless of the configuration. > > 2) It tells the next guy explicitly what you intended to happen upon > receiving the signal. > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > > > ------------------------------ > > Message: 9 > Date: Thu, 26 Aug 2010 16:51:04 -0400 > From: Matt Desbiens <desbiensm at gmail.com> > Subject: Re: [asterisk-users] double DTMF digits > To: andres at telesip.net, Asterisk Users Mailing List - Non-Commercial > Discussion <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTinfR_xKtGAnOUn4TMfPm6Afhq=GWLAyFf5ZA-_d at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > We've actually had issues with Flowroute in the past where DTMF was a > constant issue. My best suggestion for course of action is find another > provider. NexVortex is pretty solid all around. They also had the quickest > recourse for when GNAPS went bottoms up last month and sent pretty much all > VoIP traffic in New England into a tailspin. > > --Matt > > On Thu, Aug 26, 2010 at 3:23 PM, Andres <andres at telesip.net> wrote: > >> On 8/26/2010 2:55 PM, M S wrote: >> > Hi, >> > >> > I've been getting complaints lately that callers to my IVR are >> > pressing a digit once but the system is responding as if they pressed >> > it twice (once for each of two consecutive menus). >> > I'm using an AGI script and logging all DTMF entries - and to the >> > script, at least, it looks like the digit is being pressed twice. The >> > TN being called is a VOIP number (provided by Flowroute) and being >> > forwarded via SIP to my asterisk 1.6.2.4 server. The dtmfmode is set >> > to rfc28333 in sip.conf. >> > >> > The first time this happened, I figured the caller pressed the number >> > twice without realizing it. It's happening to too many people for >> > that to be plausible anymore. I also experienced it once myself, >> > months ago, when I entered my tn as 1234567890 and had it read back to >> > me as 1122334455. >> > >> > Can anyone give me some pointers where to start troubleshooting? Can >> > overloading a system cause such an error? >> > >> > Thanks, >> I have seen this before. Upon careful analisys we saw that the far end >> was sending the digits in RFC2833 plus SIP INFO (or Inband, I can't >> remember). Thus Asterisk detected double digits. The solution was to >> ask the remote end to only send RFC2833. >> >> Andres >> http://www.telesip.net >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/2d8b5f7d/attachment-0001.htm > > ------------------------------ > > Message: 10 > Date: Thu, 26 Aug 2010 18:58:31 -0700 > From: Joe Wood <schmoe at gmail.com> > Subject: [asterisk-users] Asterisk 1.6 Displaying in Debug that it is > playing a ulaw file using BackGround() but no audio is heard from the > phone > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTin5Axv=J7MOmCGoxeLQPbdNnouKXKyyJT=zdZcN at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > First off, let me first say that this is not a one-way audio problem. > Sometimes I can get 'her' to speak to me, other times I can't for a > long time. > > I'm just using a very simple system to dump people into MeetMe(). > Nothing fancy. > > I do have the following in my modules.conf: > > preload => format_mp3.so > preload => codec_ulaw.so > preload => format_pcm.so > > My extensions.conf looks like: > > [general] > autofallthrough=yes > static=no > writeprotect=no > extenpatternmatchnew=yes > clearglobalvars=no > > > [conference-calls] > exten => s,1,Answer() > exten => s,n,Background(welcome) > exten => s,n,Background(and) > exten => s,n,Background(thank-you-for-calling) > exten => s,n,Background(conference-reservations) > exten => s,n,Wait(2) > exten => s,n,Background(enter-conf-pin-number) > exten => s,n,WaitExten(10) > exten => i,1,Playback(pbx-invalid) > exten => i,n,Goto(conference-calls,9000,1) > exten => t,1,Playback(vm-goodbye) > exten => t,n,Hangup() > > exten => ${EXTEN},1,Meetme(${EXTEN}) > > > == Using SIP RTP CoS mark 5 > -- Executing [s at conference-calls:1] > Answer("SIP/2063161626-00000001", "") in new stack > == Using SIP RTP CoS mark 5 > -- Executing [s at conference-calls:1] > Answer("SIP/2063161626-00000002", "") in new stack > -- Executing [s at conference-calls:2] > BackGround("SIP/2063161626-00000001", "welcome") in new stack > -- <SIP/2063161626-00000001> Playing 'welcome.ulaw' (language 'en') > -- Executing [s at conference-calls:2] > BackGround("SIP/2063161626-00000002", "welcome") in new stack > -- <SIP/2063161626-00000002> Playing 'welcome.ulaw' (language 'en') > -- Executing [s at conference-calls:3] > BackGround("SIP/2063161626-00000001", "and") in new stack > -- <SIP/2063161626-00000001> Playing 'and.ulaw' (language 'en') > -- Executing [s at conference-calls:3] > BackGround("SIP/2063161626-00000002", "and") in new stack > -- <SIP/2063161626-00000002> Playing 'and.ulaw' (language 'en') > -- Executing [s at conference-calls:4] > BackGround("SIP/2063161626-00000001", "thank-you-for-calling") in new > stack > -- <SIP/2063161626-00000001> Playing 'thank-you-for-calling.ulaw' > (language 'en') > -- Executing [s at conference-calls:4] > BackGround("SIP/2063161626-00000002", "thank-you-for-calling") in new > stack > -- <SIP/2063161626-00000002> Playing 'thank-you-for-calling.ulaw' > (language 'en') > -- Executing [s at conference-calls:5] > BackGround("SIP/2063161626-00000001", "conference-reservations") in > new stack > -- <SIP/2063161626-00000001> Playing > 'conference-reservations.ulaw' (language 'en') > -- Executing [s at conference-calls:5] > BackGround("SIP/2063161626-00000002", "conference-reservations") in > new stack > -- <SIP/2063161626-00000002> Playing > 'conference-reservations.ulaw' (language 'en') > -- Executing [s at conference-calls:6] > Wait("SIP/2063161626-00000001", "2") in new stack > -- Executing [s at conference-calls:6] > Wait("SIP/2063161626-00000002", "2") in new stack > -- Executing [s at conference-calls:7] > BackGround("SIP/2063161626-00000001", "enter-conf-pin-number") in new > stack > -- <SIP/2063161626-00000001> Playing 'enter-conf-pin-number.ulaw' > (language 'en') > -- Executing [s at conference-calls:7] > BackGround("SIP/2063161626-00000002", "enter-conf-pin-number") in new > stack > -- <SIP/2063161626-00000002> Playing 'enter-conf-pin-number.ulaw' > (language 'en') > -- Executing [s at conference-calls:8] > WaitExten("SIP/2063161626-00000001", "10") in new stack > -- Executing [s at conference-calls:8] > WaitExten("SIP/2063161626-00000002", "10") in new stack > -- Timeout on SIP/2063161626-00000001, going to 't' > -- Executing [t at conference-calls:1] > Playback("SIP/2063161626-00000001", "vm-goodbye") in new stack > -- <SIP/2063161626-00000001> Playing 'vm-goodbye.ulaw' (language 'en') > -- Timeout on SIP/2063161626-00000002, going to 't' > -- Executing [t at conference-calls:1] > Playback("SIP/2063161626-00000002", "vm-goodbye") in new stack > -- <SIP/2063161626-00000002> Playing 'vm-goodbye.ulaw' (language 'en') > -- Executing [t at conference-calls:2] > Hangup("SIP/2063161626-00000001", "") in new stack > == Spawn extension (conference-calls, t, 2) exited non-zero on > 'SIP/2063161626-00000001' > -- Executing [t at conference-calls:2] > Hangup("SIP/2063161626-00000002", "") in new stack > == Spawn extension (conference-calls, t, 2) exited non-zero on > 'SIP/2063161626-00000002' > > Has anyone else encountered this problem before? I saw one posting on > the listserv, but it said to add in the pcm lib and I did that and no > change. > > Help. > > Thanks a bunch, > > Joe > > > > ------------------------------ > > Message: 11 > Date: Thu, 26 Aug 2010 22:25:37 -0400 > From: M S <101mcs at gmail.com> > Subject: Re: [asterisk-users] double DTMF digits > To: asterisk-users at lists.digium.com > Message-ID: > <AANLkTimn9i+Sxf_qKLTWFurohU7R6u-tx58iahgjDfns at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > How were you able to determine that the far end was sending the digits in > RFC2833 plus SIP INFO? > > On Thu, Aug 26, 2010 at 3:23 PM, Andres <andres at telesip.net> wrote: > >> >> I have seen this before. Upon careful analisys we saw that the far end >> was sending the digits in RFC2833 plus SIP INFO (or Inband, I can't >> remember). Thus Asterisk detected double digits. The solution was to >> ask the remote end to only send RFC2833. >> >> Andres >> http://www.telesip.net >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/fd1e4b43/attachment-0001.htm > > ------------------------------ > > Message: 12 > Date: Fri, 27 Aug 2010 09:36:48 +0100 > From: "Lee Archer" <Lee.Archer at thebigword.com> > Subject: Re: [asterisk-users] Use of AGISIGHUP > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <B916037C74E10442982E494BA3F108F60D2E301D at MAIL1.thebigword.com> > Content-Type: text/plain; charset="US-ASCII" > > Thanks for the replies. I am already ignoring the signal but it doesn't > seem to be making much difference on this system with this script. It's > an old legacy script I should hopefully be dropping and writing properly > within the dial plan. > > I will keep trying! > > Thanks > > Lee > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve > Edwards > Sent: 26 August 2010 21:02 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Use of AGISIGHUP > >>> On Thu, 26 Aug 2010, Lee Archer wrote: >> >>>> I am setting AGISIGHUP=no before running a Perl script via AGI but >>>> it doesn?t seem to be doing anything as the script is still exiting >>>> on a hangup and not completing properly. > >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve >> Edwards >> >>> I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's > >>> a bad idea to protect lazy programmers :) > > On Thu, 26 Aug 2010, Danny Nicholas wrote: > >> Here's a one-liner that should "fix" the problem >> >> local $SIG{HUP} = 'IGNORE'; >> >> Does that make me lazy? > > Not at all. If that is the correct "response" for your program, it's > perfect: > > 1) The program will execute correctly on your system, my system, any > system regardless of the configuration. > > 2) It tells the next guy explicitly what you intended to happen upon > receiving the signal. > > -- > Thanks in advance, > ------------------------------------------------------------------------ > - > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 > PST > Newline Fax: > +1-760-731-3000 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 13 > Date: Fri, 27 Aug 2010 11:27:57 +0200 > From: Xavier <magicrhesus at ouranos.be> > Subject: [asterisk-users] dynamic MeetMe, min. digits > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4C77851D.4090802 at ouranos.be> > Content-Type: text/plain; charset="iso-8859-1" > > Hi All, > > Is there a way to use the dynamic feature of the meetme application (D) > and to set an option to configure the minimum length of the numbers for > the conference and the associated pin. > In my case, I'd like them to be at least four digits. > > Thanks in advance ! > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/6c7202ae/attachment-0001.htm > > ------------------------------ > > Message: 14 > Date: Fri, 27 Aug 2010 05:58:57 -0400 > From: Doug Lytle <support at drdos.info> > Subject: Re: [asterisk-users] dynamic MeetMe, min. digits > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4C778C61.4080104 at drdos.info> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Xavier wrote: >> Hi All, >> >> Is there a way to use the dynamic feature of the meetme application >> (D) and to set an option to configure the minimum length of the >> numbers for the conference and the associated pin. > > You can use the read application to get the password and then check the > length, before going onto the conference setup. > > > > Doug > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > > > ------------------------------ > > Message: 15 > Date: Fri, 27 Aug 2010 12:28:38 +0200 > From: "Xavier D." <magicrhesus at ouranos.be> > Subject: Re: [asterisk-users] dynamic MeetMe, min. digits > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4C779356.1070405 at ouranos.be> > Content-Type: text/plain; charset="iso-8859-1" > > Yes but what about the conference number ? > > On 08/27/2010 11:58 AM, Doug Lytle wrote: >> Xavier wrote: >>> Hi All, >>> >>> Is there a way to use the dynamic feature of the meetme application >>> (D) and to set an option to configure the minimum length of the >>> numbers for the conference and the associated pin. >> You can use the read application to get the password and then check the >> length, before going onto the conference setup. >> >> >> >> Doug >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/37a4794d/attachment-0001.htm > > ------------------------------ > > Message: 16 > Date: Fri, 27 Aug 2010 17:09:33 +0530 > From: Tino <tino at sparksupport.com> > Subject: [asterisk-users] music on hold in blind transfer > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTikFKs7JCW-vKObVO6cBW0Fc+-KoC9ndHSO1pC1t at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello, > > Is it possible to avoid playing music on hold during a blind transfer ? > > Thanks > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/843562d4/attachment-0001.htm > > ------------------------------ > > Message: 17 > Date: Fri, 27 Aug 2010 17:35:26 +0530 > From: Tino <tino at sparksupport.com> > Subject: [asterisk-users] queue agent and blind transfer > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTinmCcg5f5EbjGO4moeNu7Cd4o6tz5f-fuMb+0S5 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello, > > When an agent does a blind transfer the call hangups for him but shows as > "In use" in queue in my CRM (used for auto dialing). As a result the agent > have to wait until the transfered call completes. Is there any way to change > this behaviour ? > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/89a65d22/attachment-0001.htm > > ------------------------------ > > Message: 18 > Date: Fri, 27 Aug 2010 08:51:04 -0400 > From: Dan Journo <dan at keshercommunications.com> > Subject: [asterisk-users] Call Forwarding > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <31C6BA8C3525D840B022617ACBB7BC036FE20831FF at VMBX123.ihostexchange.net> > Content-Type: text/plain; charset="us-ascii" > > Hi, > > I'm currently programming an interface for my Asterisk service. > > I've noticed an issue if someone sets up call forwarding on their sip phone. > Asterisk receives a 302 "Moved Temporarily" message, and forwards the call > successfully. > > However, the CDR is not correct. > > If I set up call forwarding to a mobile on extension 201, and then place a > call from extension 202, the call gets diverted. > I answer the call and talk for 30 seconds, then I hang up. > > The CDR shows two calls:- > > 2010-08-27 13:38:24 - 202 -> 201 - Answered - Billsec is 30 > 2010-08-27 13:38:24 - 202 -> 5551234 - Answered - Billsec is 0 > > 5551234 is the mobile number. > The second CDR entry should read 30 seconds, and the first should read 0 (or > 30) > > Since it isn't behaving like I want, is there any way to disable the feature > that allows a SIP phone to perform call forwarding? > > Thanks > Dan > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/72518e7d/attachment-0001.htm > > ------------------------------ > > Message: 19 > Date: Fri, 27 Aug 2010 08:52:58 -0400 > From: Paul Belanger <paul.belanger at polybeacon.com> > Subject: Re: [asterisk-users] music on hold in blind transfer > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTimnsLn2Y8T1venZrnbXQ1A8JGTvDUN7XiO0oNQn at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Fri, Aug 27, 2010 at 7:39 AM, Tino <tino at sparksupport.com> wrote: >> Is it possible to avoid playing music on hold during a blind transfer ? >> > Please do not cross-post the same message to multiple lists. > > Yes, configure an empty MoH class or not loading MoH are some options, also: > > *CLI> core show application Dial > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > > > ------------------------------ > > Message: 20 > Date: Fri, 27 Aug 2010 15:13:22 +0200 > From: Stefan Schmidt <sst at sil.at> > Subject: Re: [asterisk-users] Call Forwarding > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4C77B9F2.5030900 at sil.at> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Dan Journo schrieb: >> >> >> >> Since it isn't behaving like I want, is there any way to disable the >> feature that allows a SIP phone to perform call forwarding? >> >> >> >> Thanks >> >> Dan >> >> >> > Hello, > > in asterisk 1.6.x there is a Dial option i which suppress a 302 redirect > which is very nice when dialing more than one phone at once, but you can > use it also if you just dial one channel. > > see output of core show application dial: > > i - Asterisk will ignore any forwarding requests it may receive on > this > dial attempt. > > > best regards > > steve > > -- > F?r weitere Fragen stehen wir gerne unter voip at sil.at oder > 059944 - 2440 zur Verf?gung. > > Mit freundlichen Gr?ssen > -- > Stefan Schmidt > Sysadmin/VOIP // voip at sil.at // Tel 059944-2440// > ------------------------------------------------- > SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // > A-1160 Wien // Fax 059944-9000 // www.sil.at // > ------------------------------------------------- > > > > > ------------------------------ > > Message: 21 > Date: Fri, 27 Aug 2010 15:13:54 +0200 > From: Alex Hermann <alex at speakup.nl> > Subject: [asterisk-users] Duplicate channel variables after transfer > To: asterisk-users at lists.digium.com > Message-ID: <201008271513.54789.alex at speakup.nl> > Content-Type: text/plain; charset="us-ascii" > > Hi all, > > > with an (attended) transfer i see the following happening: > > 1) A calls B1 > 2) B2 calls C > 3) B2 transfers call to A > 4) A talks to C > > > At step 3, the channel A is connected to channel C and B1 and B2 are hung > up. > In the h extension for channel B2, the channel is renamed to B2<ZOMBIE> and > i > see that the channel variables of A have been merged into B2<ZOMBIE>. If > there > were duplicate names for variables, the channel now has those variables > doubled. The DumpChan() application called from the h extension confirms > this. > > In my case the channels are all SIP channels and in the h extension I want > to > access the SIPCALLID variable of the A channel. Every access to it gives me > the wrong value namely that of channel B1. How do i access the _second_ > variable named SIPCALLID in the channel? > > Extract from DumpChan() as an example: > > Dumping Info For Channel: SIP/sipout-00000055<ZOMBIE>: > ===============================================================================> Info: > Name= SIP/sipout-00000055<ZOMBIE> > Type= SIP > UniqueID= 1282913436.108 > .... > Variables: > ... > SIPCALLID=eae94252-ebf238ff at 172.28.4.112 > ... > SIPCALLID=lyvkqtybsgrtsnh at 172.28.4.113 > ... > ===============================================================================> > > I want to get lyvkqtybsgrtsnh at 172.28.4.113 instead of eae94252- > ebf238ff at 172.28.4.112 as a result. > > -- > Greetings, > > Alex Hermann > > > > > ------------------------------ > > Message: 22 > Date: Fri, 27 Aug 2010 15:46:44 +0200 > From: Andra? <atletek at gmail.com> > Subject: Re: [asterisk-users] CDR on Transfer... > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTim2m+fkmnS5UpGrQwQTAKpbB-N_wa29XyRK9-Qm at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Did you find the solution? > > On Thu, Aug 26, 2010 at 7:25 PM, Carlos Chavez > <cursor at telecomabmex.com>wrote: > >> I have searched for some time but I have not found an asnwer on how >> to >> fix the CDR when a call is transferred. The problem is that if someone >> dials a cell phone and then transfers the call to another extensi?n the >> CDR for the cell call stops and there is no way to track that the call >> was transferred so we can bill correctly. Many people have asked this >> question but there is no answer, only a mention that it should be fixed >> in 1.6 which it is not (at least on 1.6.2.11). >> >> Any tips oh how to correct this problem? A lot of customers give >> me >> grief about this because they cannot properly bill people for their cell >> calls. >> >> -- >> Telecomunicaciones Abiertas de M?xico S.A. de C.V. >> Carlos Ch?vez Prats >> Director de Tecnolog?a >> +52-55-91169161 ext 2001 >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/cafd0bbf/attachment.htm > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > AstriCon 2010 - October 26-28 Washington, DC > Register Now: http://www.astricon.net/ > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of asterisk-users Digest, Vol 73, Issue 58 > ********************************************** >-- Sent from my mobile device Your kind advice is highly appreciated. Warmest regards, Jonathan Leong Chief Executive Officer eSky Multimedia Sdn. Bhd. Address: 51-01-02 Jalan Austin Heights 3, Taman Mount Austin 81100 Johor Bahru, Johor, Malaysia R&D Lab: Esky Multimedia Resources Lab, AR0008 (Faculty of Engineering, MMU), Persiaran Multimedia 63100 Cyberjaya, Selangor, Malaysia USA Office : 1584, Meridian Ave, San Jose 95125 CA web : www.e-numX.com e-numX : 8881000-2288 e-mail: jonathan at e-numx.com Tel : +6.07.352.7777 Fax : +6.03.9235.1122 Cell : +6.012.772.2700 Malaysia DID : +6.03.2772.7398 USA DID : +1.408.587.7999
Jonathan Leong
2010-Aug-27 14:05 UTC
[asterisk-users] asterisk-users Digest, Vol 73, Issue 58
On 8/27/10, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote:> Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request at lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner at lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. CDR on Transfer... (Carlos Chavez) > 2. Re: Asterisk 1.6.1.17 ACK/BYE question (Trevor Benson) > 3. Re: Use of AGISIGHUP (Danny Nicholas) > 4. double DTMF digits (M S) > 5. Re: double DTMF digits (Andres) > 6. Re: Use of AGISIGHUP (Steve Edwards) > 7. Re: Use of AGISIGHUP (Danny Nicholas) > 8. Re: Use of AGISIGHUP (Steve Edwards) > 9. Re: double DTMF digits (Matt Desbiens) > 10. Asterisk 1.6 Displaying in Debug that it is playing a ulaw > file using BackGround() but no audio is heard from the phone > (Joe Wood) > 11. Re: double DTMF digits (M S) > 12. Re: Use of AGISIGHUP (Lee Archer) > 13. dynamic MeetMe, min. digits (Xavier) > 14. Re: dynamic MeetMe, min. digits (Doug Lytle) > 15. Re: dynamic MeetMe, min. digits (Xavier D.) > 16. music on hold in blind transfer (Tino) > 17. queue agent and blind transfer (Tino) > 18. Call Forwarding (Dan Journo) > 19. Re: music on hold in blind transfer (Paul Belanger) > 20. Re: Call Forwarding (Stefan Schmidt) > 21. Duplicate channel variables after transfer (Alex Hermann) > 22. Re: CDR on Transfer... (Andra?) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 26 Aug 2010 12:25:07 -0500 > From: Carlos Chavez <cursor at telecomabmex.com> > Subject: [asterisk-users] CDR on Transfer... > To: Asterisk <asterisk-users at lists.digium.com> > Message-ID: <1282843507.2830.13.camel at cursor.telecomabmex.com> > Content-Type: text/plain; charset="utf-8" > > I have searched for some time but I have not found an asnwer on how to > fix the CDR when a call is transferred. The problem is that if someone > dials a cell phone and then transfers the call to another extensi?n the > CDR for the cell call stops and there is no way to track that the call > was transferred so we can bill correctly. Many people have asked this > question but there is no answer, only a mention that it should be fixed > in 1.6 which it is not (at least on 1.6.2.11). > > Any tips oh how to correct this problem? A lot of customers give me > grief about this because they cannot properly bill people for their cell > calls. > > -- > Telecomunicaciones Abiertas de M?xico S.A. de C.V. > Carlos Ch?vez Prats > Director de Tecnolog?a > +52-55-91169161 ext 2001 > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 198 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/f970d9ee/attachment-0001.pgp > > ------------------------------ > > Message: 2 > Date: Thu, 26 Aug 2010 10:30:16 -0700 > From: Trevor Benson <tbenson at a-1networks.com> > Subject: Re: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <65F20266-3E46-4DCC-A17D-D181F8E4A6AE at a-1networks.com> > Content-Type: text/plain; charset="windows-1252" > > We have a box running 1.6.2.11 on CentOS 5 using the RPM's from the Digium > CentOS repository. We just left a 60 second voicemail on the system and had > the full audio as well in the inbox. Not sure how your SIP configuration > ties your SBC in, but native "users" created via users.conf and sip.conf > appears to be working for me. Wouldnt be able to test more without knowing > what settings you had between Asterisk and the SBC. > > > -- > Trevor Benson > dCAP, LPIC-1, CLA, Network+, MCP, CNA > A1 Networks - Network Engineer > DID (707)703-1041 > FAX (707)703-1983 > > > > > > > On Aug 26, 2010, at 8:47 AM, Steven C. Blair wrote: > >> >> As a test we built Asterisk v1.6.2.11 on a new server. This version of >> Asterisk exhibits the same behavior. From ngrep?s perspective we see an >> ACK followed immediately by a BYE message. The user hears the recording >> being played, begins to leave a message and is disconnected about 10 >> seconds into the call. >> >> >> >> From: asterisk-users-bounces at lists.digium.com >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven C. >> Blair >> Sent: Wednesday, August 25, 2010 2:08 PM >> To: asterisk-users at lists.digium.com >> Subject: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question >> >> >> We?re running Asterisk 1.6.1.17 for our campus voicemail server and >> Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are >> diverted to voicemail using a 302 redirect when the called party doesn?t >> answer. In this case the caller is able to hear the greetings and begin to >> leave a message only to have Asterisk terminate the call mid-recording. >> >> We?re uncertain why this is happening and this is where we are hoping you >> can help. In our environment the caller is any set on the PSTN. They call >> one of our IP phones which no one answers. Our proxy, SER, responds to the >> SBC with a 302 redirect and the call is diverted to Asterisk. The caller >> hears the unavailable greeting for 6-4050, begins to leave a message and >> is cut-off after about 10 seconds. In an ngrep trace we see Asterisk >> receive an ACK from the SBC and it immediately responds with a BYE message >> for that call. >> >> Has anyone else experienced this type of issue? >> >> >> --- >> >> ISC Networking & Telecommunications >> 3401 Walnut Street, Suite 221A >> Philadelphia, PA 19104 >> 215-573-8396 >> 215-898-9348 (fax) >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/59a22a5a/attachment-0001.htm > > ------------------------------ > > Message: 3 > Date: Thu, 26 Aug 2010 12:58:46 -0500 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] Use of AGISIGHUP > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <201008261730.o7QHULt4029526 at mail.debsinc.com> > Content-Type: text/plain; charset="us-ascii" > > Can you post the CLI output showing the hangup/script failure? > > > > _____ > > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Lee Archer > Sent: Thursday, August 26, 2010 11:39 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Use of AGISIGHUP > > > > Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but it > doesn't seem to be doing anything as the script is still exiting on a hangup > and not completing properly. I am using 1.4.35 and have tried various > combinations. Can anyone shed any light on this? > > Regards > > Lee > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/7a77e4e1/attachment-0001.htm > > ------------------------------ > > Message: 4 > Date: Thu, 26 Aug 2010 14:55:50 -0400 > From: M S <101mcs at gmail.com> > Subject: [asterisk-users] double DTMF digits > To: asterisk-users at lists.digium.com > Message-ID: > <AANLkTinoT4+HrPBHdKAx6WOU-vtyUPq3KcH9vRAE=E53 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > I've been getting complaints lately that callers to my IVR are pressing a > digit once but the system is responding as if they pressed it twice (once > for each of two consecutive menus). > I'm using an AGI script and logging all DTMF entries - and to the script, at > least, it looks like the digit is being pressed twice. The TN being called > is a VOIP number (provided by Flowroute) and being forwarded via SIP to my > asterisk 1.6.2.4 server. The dtmfmode is set to rfc28333 in sip.conf. > > The first time this happened, I figured the caller pressed the number twice > without realizing it. It's happening to too many people for that to be > plausible anymore. I also experienced it once myself, months ago, when I > entered my tn as 1234567890 and had it read back to me as 1122334455. > > Can anyone give me some pointers where to start troubleshooting? Can > overloading a system cause such an error? > > Thanks, > Mira > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/c9db8975/attachment-0001.htm > > ------------------------------ > > Message: 5 > Date: Thu, 26 Aug 2010 15:23:44 -0400 > From: Andres <andres at telesip.net> > Subject: Re: [asterisk-users] double DTMF digits > To: asterisk-users at lists.digium.com > Message-ID: <4C76BF40.20204 at telesip.net> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 8/26/2010 2:55 PM, M S wrote: >> Hi, >> >> I've been getting complaints lately that callers to my IVR are >> pressing a digit once but the system is responding as if they pressed >> it twice (once for each of two consecutive menus). >> I'm using an AGI script and logging all DTMF entries - and to the >> script, at least, it looks like the digit is being pressed twice. The >> TN being called is a VOIP number (provided by Flowroute) and being >> forwarded via SIP to my asterisk 1.6.2.4 server. The dtmfmode is set >> to rfc28333 in sip.conf. >> >> The first time this happened, I figured the caller pressed the number >> twice without realizing it. It's happening to too many people for >> that to be plausible anymore. I also experienced it once myself, >> months ago, when I entered my tn as 1234567890 and had it read back to >> me as 1122334455. >> >> Can anyone give me some pointers where to start troubleshooting? Can >> overloading a system cause such an error? >> >> Thanks, > I have seen this before. Upon careful analisys we saw that the far end > was sending the digits in RFC2833 plus SIP INFO (or Inband, I can't > remember). Thus Asterisk detected double digits. The solution was to > ask the remote end to only send RFC2833. > > Andres > http://www.telesip.net > > > > ------------------------------ > > Message: 6 > Date: Thu, 26 Aug 2010 12:41:25 -0700 (PDT) > From: Steve Edwards <asterisk.org at sedwards.com> > Subject: Re: [asterisk-users] Use of AGISIGHUP > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <alpine.DEB.2.00.1008261233160.25774 at localhost.localdomain> > Content-Type: text/plain; charset="iso-8859-7" > > On Thu, 26 Aug 2010, Lee Archer wrote: > >> Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but >> it doesn?t seem to be doing anything as the script is still exiting on a >> hangup and not completing properly.? I am using 1.4.35 and have tried >> various combinations.? Can anyone shed any light on this? > > I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's a > bad idea to protect lazy programmers :) > > Program defensively! > > Trap the HUP and do what is appropriate for your script -- even if that is > to ignore it. > > If the successful execution of your system depends on a setting, how long > will it take the next guy to figure out when the setting disappears > unexpectedly? > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > ------------------------------ > > Message: 7 > Date: Thu, 26 Aug 2010 14:52:53 -0500 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] Use of AGISIGHUP > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <201008261924.o7QJOREn030652 at mail.debsinc.com> > Content-Type: text/plain; charset="iso-8859-1" > >>From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Edwards >>Subject: Re: [asterisk-users] Use of AGISIGHUP > >>On Thu, 26 Aug 2010, Lee Archer wrote: > >>> Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but >>> it doesn?t seem to be doing anything as the script is still exiting on a >>> hangup and not completing properly.? I am using 1.4.35 and have tried >>> various combinations.? Can anyone shed any light on this? > >>I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's a > bad idea to protect lazy programmers :) > >>Program defensively! > >>Trap the HUP and do what is appropriate for your script -- even if that is > to ignore it. > >>If the successful execution of your system depends on a setting, how long > will it take the next guy to figure out when the setting disappears > unexpectedly? > > As usual, Steve is right. Here's a one-liner that should "fix" the problem > > local $SIG{HUP} = 'IGNORE'; > > Does that make me lazy? > > TIA. > > > > > ------------------------------ > > Message: 8 > Date: Thu, 26 Aug 2010 13:02:20 -0700 (PDT) > From: Steve Edwards <asterisk.org at sedwards.com> > Subject: Re: [asterisk-users] Use of AGISIGHUP > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <alpine.DEB.2.00.1008261257210.15301 at localhost.localdomain> > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > >>> On Thu, 26 Aug 2010, Lee Archer wrote: >> >>>> I am setting AGISIGHUP=no before running a Perl script via AGI but it >>>> doesn?t seem to be doing anything as the script is still exiting on a >>>> hangup and not completing properly. > >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve >> Edwards >> >>> I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's a >>> bad idea to protect lazy programmers :) > > On Thu, 26 Aug 2010, Danny Nicholas wrote: > >> Here's a one-liner that should "fix" the problem >> >> local $SIG{HUP} = 'IGNORE'; >> >> Does that make me lazy? > > Not at all. If that is the correct "response" for your program, it's > perfect: > > 1) The program will execute correctly on your system, my system, any > system regardless of the configuration. > > 2) It tells the next guy explicitly what you intended to happen upon > receiving the signal. > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > > > ------------------------------ > > Message: 9 > Date: Thu, 26 Aug 2010 16:51:04 -0400 > From: Matt Desbiens <desbiensm at gmail.com> > Subject: Re: [asterisk-users] double DTMF digits > To: andres at telesip.net, Asterisk Users Mailing List - Non-Commercial > Discussion <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTinfR_xKtGAnOUn4TMfPm6Afhq=GWLAyFf5ZA-_d at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > We've actually had issues with Flowroute in the past where DTMF was a > constant issue. My best suggestion for course of action is find another > provider. NexVortex is pretty solid all around. They also had the quickest > recourse for when GNAPS went bottoms up last month and sent pretty much all > VoIP traffic in New England into a tailspin. > > --Matt > > On Thu, Aug 26, 2010 at 3:23 PM, Andres <andres at telesip.net> wrote: > >> On 8/26/2010 2:55 PM, M S wrote: >> > Hi, >> > >> > I've been getting complaints lately that callers to my IVR are >> > pressing a digit once but the system is responding as if they pressed >> > it twice (once for each of two consecutive menus). >> > I'm using an AGI script and logging all DTMF entries - and to the >> > script, at least, it looks like the digit is being pressed twice. The >> > TN being called is a VOIP number (provided by Flowroute) and being >> > forwarded via SIP to my asterisk 1.6.2.4 server. The dtmfmode is set >> > to rfc28333 in sip.conf. >> > >> > The first time this happened, I figured the caller pressed the number >> > twice without realizing it. It's happening to too many people for >> > that to be plausible anymore. I also experienced it once myself, >> > months ago, when I entered my tn as 1234567890 and had it read back to >> > me as 1122334455. >> > >> > Can anyone give me some pointers where to start troubleshooting? Can >> > overloading a system cause such an error? >> > >> > Thanks, >> I have seen this before. Upon careful analisys we saw that the far end >> was sending the digits in RFC2833 plus SIP INFO (or Inband, I can't >> remember). Thus Asterisk detected double digits. The solution was to >> ask the remote end to only send RFC2833. >> >> Andres >> http://www.telesip.net >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/2d8b5f7d/attachment-0001.htm > > ------------------------------ > > Message: 10 > Date: Thu, 26 Aug 2010 18:58:31 -0700 > From: Joe Wood <schmoe at gmail.com> > Subject: [asterisk-users] Asterisk 1.6 Displaying in Debug that it is > playing a ulaw file using BackGround() but no audio is heard from the > phone > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTin5Axv=J7MOmCGoxeLQPbdNnouKXKyyJT=zdZcN at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > First off, let me first say that this is not a one-way audio problem. > Sometimes I can get 'her' to speak to me, other times I can't for a > long time. > > I'm just using a very simple system to dump people into MeetMe(). > Nothing fancy. > > I do have the following in my modules.conf: > > preload => format_mp3.so > preload => codec_ulaw.so > preload => format_pcm.so > > My extensions.conf looks like: > > [general] > autofallthrough=yes > static=no > writeprotect=no > extenpatternmatchnew=yes > clearglobalvars=no > > > [conference-calls] > exten => s,1,Answer() > exten => s,n,Background(welcome) > exten => s,n,Background(and) > exten => s,n,Background(thank-you-for-calling) > exten => s,n,Background(conference-reservations) > exten => s,n,Wait(2) > exten => s,n,Background(enter-conf-pin-number) > exten => s,n,WaitExten(10) > exten => i,1,Playback(pbx-invalid) > exten => i,n,Goto(conference-calls,9000,1) > exten => t,1,Playback(vm-goodbye) > exten => t,n,Hangup() > > exten => ${EXTEN},1,Meetme(${EXTEN}) > > > == Using SIP RTP CoS mark 5 > -- Executing [s at conference-calls:1] > Answer("SIP/2063161626-00000001", "") in new stack > == Using SIP RTP CoS mark 5 > -- Executing [s at conference-calls:1] > Answer("SIP/2063161626-00000002", "") in new stack > -- Executing [s at conference-calls:2] > BackGround("SIP/2063161626-00000001", "welcome") in new stack > -- <SIP/2063161626-00000001> Playing 'welcome.ulaw' (language 'en') > -- Executing [s at conference-calls:2] > BackGround("SIP/2063161626-00000002", "welcome") in new stack > -- <SIP/2063161626-00000002> Playing 'welcome.ulaw' (language 'en') > -- Executing [s at conference-calls:3] > BackGround("SIP/2063161626-00000001", "and") in new stack > -- <SIP/2063161626-00000001> Playing 'and.ulaw' (language 'en') > -- Executing [s at conference-calls:3] > BackGround("SIP/2063161626-00000002", "and") in new stack > -- <SIP/2063161626-00000002> Playing 'and.ulaw' (language 'en') > -- Executing [s at conference-calls:4] > BackGround("SIP/2063161626-00000001", "thank-you-for-calling") in new > stack > -- <SIP/2063161626-00000001> Playing 'thank-you-for-calling.ulaw' > (language 'en') > -- Executing [s at conference-calls:4] > BackGround("SIP/2063161626-00000002", "thank-you-for-calling") in new > stack > -- <SIP/2063161626-00000002> Playing 'thank-you-for-calling.ulaw' > (language 'en') > -- Executing [s at conference-calls:5] > BackGround("SIP/2063161626-00000001", "conference-reservations") in > new stack > -- <SIP/2063161626-00000001> Playing > 'conference-reservations.ulaw' (language 'en') > -- Executing [s at conference-calls:5] > BackGround("SIP/2063161626-00000002", "conference-reservations") in > new stack > -- <SIP/2063161626-00000002> Playing > 'conference-reservations.ulaw' (language 'en') > -- Executing [s at conference-calls:6] > Wait("SIP/2063161626-00000001", "2") in new stack > -- Executing [s at conference-calls:6] > Wait("SIP/2063161626-00000002", "2") in new stack > -- Executing [s at conference-calls:7] > BackGround("SIP/2063161626-00000001", "enter-conf-pin-number") in new > stack > -- <SIP/2063161626-00000001> Playing 'enter-conf-pin-number.ulaw' > (language 'en') > -- Executing [s at conference-calls:7] > BackGround("SIP/2063161626-00000002", "enter-conf-pin-number") in new > stack > -- <SIP/2063161626-00000002> Playing 'enter-conf-pin-number.ulaw' > (language 'en') > -- Executing [s at conference-calls:8] > WaitExten("SIP/2063161626-00000001", "10") in new stack > -- Executing [s at conference-calls:8] > WaitExten("SIP/2063161626-00000002", "10") in new stack > -- Timeout on SIP/2063161626-00000001, going to 't' > -- Executing [t at conference-calls:1] > Playback("SIP/2063161626-00000001", "vm-goodbye") in new stack > -- <SIP/2063161626-00000001> Playing 'vm-goodbye.ulaw' (language 'en') > -- Timeout on SIP/2063161626-00000002, going to 't' > -- Executing [t at conference-calls:1] > Playback("SIP/2063161626-00000002", "vm-goodbye") in new stack > -- <SIP/2063161626-00000002> Playing 'vm-goodbye.ulaw' (language 'en') > -- Executing [t at conference-calls:2] > Hangup("SIP/2063161626-00000001", "") in new stack > == Spawn extension (conference-calls, t, 2) exited non-zero on > 'SIP/2063161626-00000001' > -- Executing [t at conference-calls:2] > Hangup("SIP/2063161626-00000002", "") in new stack > == Spawn extension (conference-calls, t, 2) exited non-zero on > 'SIP/2063161626-00000002' > > Has anyone else encountered this problem before? I saw one posting on > the listserv, but it said to add in the pcm lib and I did that and no > change. > > Help. > > Thanks a bunch, > > Joe > > > > ------------------------------ > > Message: 11 > Date: Thu, 26 Aug 2010 22:25:37 -0400 > From: M S <101mcs at gmail.com> > Subject: Re: [asterisk-users] double DTMF digits > To: asterisk-users at lists.digium.com > Message-ID: > <AANLkTimn9i+Sxf_qKLTWFurohU7R6u-tx58iahgjDfns at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > How were you able to determine that the far end was sending the digits in > RFC2833 plus SIP INFO? > > On Thu, Aug 26, 2010 at 3:23 PM, Andres <andres at telesip.net> wrote: > >> >> I have seen this before. Upon careful analisys we saw that the far end >> was sending the digits in RFC2833 plus SIP INFO (or Inband, I can't >> remember). Thus Asterisk detected double digits. The solution was to >> ask the remote end to only send RFC2833. >> >> Andres >> http://www.telesip.net >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/fd1e4b43/attachment-0001.htm > > ------------------------------ > > Message: 12 > Date: Fri, 27 Aug 2010 09:36:48 +0100 > From: "Lee Archer" <Lee.Archer at thebigword.com> > Subject: Re: [asterisk-users] Use of AGISIGHUP > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <B916037C74E10442982E494BA3F108F60D2E301D at MAIL1.thebigword.com> > Content-Type: text/plain; charset="US-ASCII" > > Thanks for the replies. I am already ignoring the signal but it doesn't > seem to be making much difference on this system with this script. It's > an old legacy script I should hopefully be dropping and writing properly > within the dial plan. > > I will keep trying! > > Thanks > > Lee > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve > Edwards > Sent: 26 August 2010 21:02 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Use of AGISIGHUP > >>> On Thu, 26 Aug 2010, Lee Archer wrote: >> >>>> I am setting AGISIGHUP=no before running a Perl script via AGI but >>>> it doesn?t seem to be doing anything as the script is still exiting >>>> on a hangup and not completing properly. > >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve >> Edwards >> >>> I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's > >>> a bad idea to protect lazy programmers :) > > On Thu, 26 Aug 2010, Danny Nicholas wrote: > >> Here's a one-liner that should "fix" the problem >> >> local $SIG{HUP} = 'IGNORE'; >> >> Does that make me lazy? > > Not at all. If that is the correct "response" for your program, it's > perfect: > > 1) The program will execute correctly on your system, my system, any > system regardless of the configuration. > > 2) It tells the next guy explicitly what you intended to happen upon > receiving the signal. > > -- > Thanks in advance, > ------------------------------------------------------------------------ > - > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 > PST > Newline Fax: > +1-760-731-3000 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 13 > Date: Fri, 27 Aug 2010 11:27:57 +0200 > From: Xavier <magicrhesus at ouranos.be> > Subject: [asterisk-users] dynamic MeetMe, min. digits > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4C77851D.4090802 at ouranos.be> > Content-Type: text/plain; charset="iso-8859-1" > > Hi All, > > Is there a way to use the dynamic feature of the meetme application (D) > and to set an option to configure the minimum length of the numbers for > the conference and the associated pin. > In my case, I'd like them to be at least four digits. > > Thanks in advance ! > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/6c7202ae/attachment-0001.htm > > ------------------------------ > > Message: 14 > Date: Fri, 27 Aug 2010 05:58:57 -0400 > From: Doug Lytle <support at drdos.info> > Subject: Re: [asterisk-users] dynamic MeetMe, min. digits > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4C778C61.4080104 at drdos.info> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Xavier wrote: >> Hi All, >> >> Is there a way to use the dynamic feature of the meetme application >> (D) and to set an option to configure the minimum length of the >> numbers for the conference and the associated pin. > > You can use the read application to get the password and then check the > length, before going onto the conference setup. > > > > Doug > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > > > ------------------------------ > > Message: 15 > Date: Fri, 27 Aug 2010 12:28:38 +0200 > From: "Xavier D." <magicrhesus at ouranos.be> > Subject: Re: [asterisk-users] dynamic MeetMe, min. digits > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4C779356.1070405 at ouranos.be> > Content-Type: text/plain; charset="iso-8859-1" > > Yes but what about the conference number ? > > On 08/27/2010 11:58 AM, Doug Lytle wrote: >> Xavier wrote: >>> Hi All, >>> >>> Is there a way to use the dynamic feature of the meetme application >>> (D) and to set an option to configure the minimum length of the >>> numbers for the conference and the associated pin. >> You can use the read application to get the password and then check the >> length, before going onto the conference setup. >> >> >> >> Doug >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/37a4794d/attachment-0001.htm > > ------------------------------ > > Message: 16 > Date: Fri, 27 Aug 2010 17:09:33 +0530 > From: Tino <tino at sparksupport.com> > Subject: [asterisk-users] music on hold in blind transfer > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTikFKs7JCW-vKObVO6cBW0Fc+-KoC9ndHSO1pC1t at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello, > > Is it possible to avoid playing music on hold during a blind transfer ? > > Thanks > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/843562d4/attachment-0001.htm > > ------------------------------ > > Message: 17 > Date: Fri, 27 Aug 2010 17:35:26 +0530 > From: Tino <tino at sparksupport.com> > Subject: [asterisk-users] queue agent and blind transfer > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTinmCcg5f5EbjGO4moeNu7Cd4o6tz5f-fuMb+0S5 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello, > > When an agent does a blind transfer the call hangups for him but shows as > "In use" in queue in my CRM (used for auto dialing). As a result the agent > have to wait until the transfered call completes. Is there any way to change > this behaviour ? > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/89a65d22/attachment-0001.htm > > ------------------------------ > > Message: 18 > Date: Fri, 27 Aug 2010 08:51:04 -0400 > From: Dan Journo <dan at keshercommunications.com> > Subject: [asterisk-users] Call Forwarding > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <31C6BA8C3525D840B022617ACBB7BC036FE20831FF at VMBX123.ihostexchange.net> > Content-Type: text/plain; charset="us-ascii" > > Hi, > > I'm currently programming an interface for my Asterisk service. > > I've noticed an issue if someone sets up call forwarding on their sip phone. > Asterisk receives a 302 "Moved Temporarily" message, and forwards the call > successfully. > > However, the CDR is not correct. > > If I set up call forwarding to a mobile on extension 201, and then place a > call from extension 202, the call gets diverted. > I answer the call and talk for 30 seconds, then I hang up. > > The CDR shows two calls:- > > 2010-08-27 13:38:24 - 202 -> 201 - Answered - Billsec is 30 > 2010-08-27 13:38:24 - 202 -> 5551234 - Answered - Billsec is 0 > > 5551234 is the mobile number. > The second CDR entry should read 30 seconds, and the first should read 0 (or > 30) > > Since it isn't behaving like I want, is there any way to disable the feature > that allows a SIP phone to perform call forwarding? > > Thanks > Dan > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/72518e7d/attachment-0001.htm > > ------------------------------ > > Message: 19 > Date: Fri, 27 Aug 2010 08:52:58 -0400 > From: Paul Belanger <paul.belanger at polybeacon.com> > Subject: Re: [asterisk-users] music on hold in blind transfer > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTimnsLn2Y8T1venZrnbXQ1A8JGTvDUN7XiO0oNQn at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Fri, Aug 27, 2010 at 7:39 AM, Tino <tino at sparksupport.com> wrote: >> Is it possible to avoid playing music on hold during a blind transfer ? >> > Please do not cross-post the same message to multiple lists. > > Yes, configure an empty MoH class or not loading MoH are some options, also: > > *CLI> core show application Dial > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > > > ------------------------------ > > Message: 20 > Date: Fri, 27 Aug 2010 15:13:22 +0200 > From: Stefan Schmidt <sst at sil.at> > Subject: Re: [asterisk-users] Call Forwarding > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4C77B9F2.5030900 at sil.at> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Dan Journo schrieb: >> >> >> >> Since it isn't behaving like I want, is there any way to disable the >> feature that allows a SIP phone to perform call forwarding? >> >> >> >> Thanks >> >> Dan >> >> >> > Hello, > > in asterisk 1.6.x there is a Dial option i which suppress a 302 redirect > which is very nice when dialing more than one phone at once, but you can > use it also if you just dial one channel. > > see output of core show application dial: > > i - Asterisk will ignore any forwarding requests it may receive on > this > dial attempt. > > > best regards > > steve > > -- > F?r weitere Fragen stehen wir gerne unter voip at sil.at oder > 059944 - 2440 zur Verf?gung. > > Mit freundlichen Gr?ssen > -- > Stefan Schmidt > Sysadmin/VOIP // voip at sil.at // Tel 059944-2440// > ------------------------------------------------- > SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // > A-1160 Wien // Fax 059944-9000 // www.sil.at // > ------------------------------------------------- > > > > > ------------------------------ > > Message: 21 > Date: Fri, 27 Aug 2010 15:13:54 +0200 > From: Alex Hermann <alex at speakup.nl> > Subject: [asterisk-users] Duplicate channel variables after transfer > To: asterisk-users at lists.digium.com > Message-ID: <201008271513.54789.alex at speakup.nl> > Content-Type: text/plain; charset="us-ascii" > > Hi all, > > > with an (attended) transfer i see the following happening: > > 1) A calls B1 > 2) B2 calls C > 3) B2 transfers call to A > 4) A talks to C > > > At step 3, the channel A is connected to channel C and B1 and B2 are hung > up. > In the h extension for channel B2, the channel is renamed to B2<ZOMBIE> and > i > see that the channel variables of A have been merged into B2<ZOMBIE>. If > there > were duplicate names for variables, the channel now has those variables > doubled. The DumpChan() application called from the h extension confirms > this. > > In my case the channels are all SIP channels and in the h extension I want > to > access the SIPCALLID variable of the A channel. Every access to it gives me > the wrong value namely that of channel B1. How do i access the _second_ > variable named SIPCALLID in the channel? > > Extract from DumpChan() as an example: > > Dumping Info For Channel: SIP/sipout-00000055<ZOMBIE>: > ===============================================================================> Info: > Name= SIP/sipout-00000055<ZOMBIE> > Type= SIP > UniqueID= 1282913436.108 > .... > Variables: > ... > SIPCALLID=eae94252-ebf238ff at 172.28.4.112 > ... > SIPCALLID=lyvkqtybsgrtsnh at 172.28.4.113 > ... > ===============================================================================> > > I want to get lyvkqtybsgrtsnh at 172.28.4.113 instead of eae94252- > ebf238ff at 172.28.4.112 as a result. > > -- > Greetings, > > Alex Hermann > > > > > ------------------------------ > > Message: 22 > Date: Fri, 27 Aug 2010 15:46:44 +0200 > From: Andra? <atletek at gmail.com> > Subject: Re: [asterisk-users] CDR on Transfer... > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <AANLkTim2m+fkmnS5UpGrQwQTAKpbB-N_wa29XyRK9-Qm at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Did you find the solution? > > On Thu, Aug 26, 2010 at 7:25 PM, Carlos Chavez > <cursor at telecomabmex.com>wrote: > >> I have searched for some time but I have not found an asnwer on how >> to >> fix the CDR when a call is transferred. The problem is that if someone >> dials a cell phone and then transfers the call to another extensi?n the >> CDR for the cell call stops and there is no way to track that the call >> was transferred so we can bill correctly. Many people have asked this >> question but there is no answer, only a mention that it should be fixed >> in 1.6 which it is not (at least on 1.6.2.11). >> >> Any tips oh how to correct this problem? A lot of customers give >> me >> grief about this because they cannot properly bill people for their cell >> calls. >> >> -- >> Telecomunicaciones Abiertas de M?xico S.A. de C.V. >> Carlos Ch?vez Prats >> Director de Tecnolog?a >> +52-55-91169161 ext 2001 >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/cafd0bbf/attachment.htm > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > AstriCon 2010 - October 26-28 Washington, DC > Register Now: http://www.astricon.net/ > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of asterisk-users Digest, Vol 73, Issue 58 > ********************************************** >-- Sent from my mobile device Your kind advice is highly appreciated. Warmest regards, Jonathan Leong Chief Executive Officer eSky Multimedia Sdn. Bhd. Address: 51-01-02 Jalan Austin Heights 3, Taman Mount Austin 81100 Johor Bahru, Johor, Malaysia R&D Lab: Esky Multimedia Resources Lab, AR0008 (Faculty of Engineering, MMU), Persiaran Multimedia 63100 Cyberjaya, Selangor, Malaysia USA Office : 1584, Meridian Ave, San Jose 95125 CA web : www.e-numX.com e-numX : 8881000-2288 e-mail: jonathan at e-numx.com Tel : +6.07.352.7777 Fax : +6.03.9235.1122 Cell : +6.012.772.2700 Malaysia DID : +6.03.2772.7398 USA DID : +1.408.587.7999
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