Reynaldo Simbulan
2004-Mar-31 01:39 UTC
[Asterisk-Users] RE: RxFax/spandsp: not disconnecting
Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4>>> MCF: 8cHDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag In state 8 Disconnecting Changed from phase 3 to 7 *CLI> show channels Channel (Context Extension Pri ) State Appl. Data Zap/1-1 (faxserver s 3 ) Up RxFAX /var/lib/asterisk/fax/new/20040329-234801-0755965128.tif 1 active channel(s) ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, March 31, 2004 6:27 PM Subject: Asterisk-Users digest, Vol 1 #3273 - 10 msgs> Send Asterisk-Users mailing list submissions to > asterisk-users@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@lists.digium.com > > You can reach the person managing the list at > asterisk-users-admin@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. Re: Asterisk Security Audit? (Steven Critchfield) > 2. DTMF Detection Problem (Ron McMillin) > 3. Re: Caller entered digits ignored during wait.... (Tilghman Lesher) > 4. Re: Sipcall.co.uk & [*] (Dave Cotton) > 5. Re: IAX2 trunk mode over satellite (clive18@webmail.co.za) > 6. Register vith SIP provider from behind NAT (Simon Brown) > 7. Can't talk on Cisco VIP 30 using Chan Skinny (Dean) > 8. Re: Caller entered digits ignored during > wait.... (Stig Andersson) > 9. RE: Exception flag set - snom200 (jc) > > --__--__-- > > Message: 1 > Subject: Re: [Asterisk-Users] Asterisk Security Audit? > From: Steven Critchfield <critch@basesys.com> > To: asterisk-users@lists.digium.com > Date: Tue, 30 Mar 2004 23:03:55 -0600 > Reply-To: asterisk-users@lists.digium.com > > On Tue, 2004-03-30 at 16:53, Jim Rosenberg wrote: > > Has Asterisk ever been audited for common security holes, such as buffer > > overruns? > > > > A quick grep through the source for routines that should never be used, > > like strcpy, strcat, etc., reveals a lot of it. I fear I fear. > > These functions aren't as bad as you make out. They are only dangerous > when used with unchecked buffers that where accepted from outside > sources. There are quite a few instances of strcpy and strcat that are > using string constants and therefore are safe. > > Don't take that as an argument against checking other possible security > concerns. Just as a reminder that the mere existence of certain > functions doesn't mean it is unsafe. > > Also this discussion is probably better dealt with on the -dev list > where the noise level is better suited for the developers you need to > target to actually see this message. > -- > Steven Critchfield <critch@basesys.com> > > > --__--__-- > > Message: 2 > Date: Tue, 30 Mar 2004 21:45:19 -0800 (PST) > From: Ron McMillin <sipnow@sbcglobal.net> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] DTMF Detection Problem > Reply-To: asterisk-users@lists.digium.com > > --0-1376241818-1080711919=:58147 > Content-Type: text/plain; charset=us-ascii > > Hi, > My set up is like this > Asterisk--->SipuraATA----->AnalogPhone > When I'm calling into asterisk from a cell phone, there's no dtmfdetection problem as asterisk can detect correct extensions that I press. But when the phone is further connected to the AnalogPhone thru the ATA, the dtmf signal is really short/weak. I've tried to adjust dtmf lengths, gain, etc. on the ATA and it helps a little bit, but not much. And this seems to be a problem only if I call in from a cell phone. If I were to use a SIP phone to call in, it works much better.> > Is there a way to make Asterisk to regenerate the DTMF tones to improvethe DTMF tones? Such as making it interpret the DTMF tones and regenerate it w/ a certain length regardless of original signal length. The reason I want to DTMF comes to AnalogPhone clearly is because I want to ultimately connect it to a FXSFXO converter and go back out to PSTN line.> > Thank you > Ron > > --0-1376241818-1080711919=:58147 > Content-Type: text/html; charset=us-ascii > > <DIV>Hi,</DIV> > <DIV> My set up is like this</DIV> > <DIV>Asterisk--->SipuraATA----->AnalogPhone</DIV> > <DIV>When I'm calling into asterisk from a cell phone, there's no dtmfdetection problem as asterisk can detect correct extensions that I press. But when the phone is further connected to the AnalogPhone thru the ATA, the dtmf signal is really short/weak. I've tried to adjust dtmf lengths, gain, etc. on the ATA and it helps a little bit, but not much. And this seems to be a problem only if I call in from a cell phone. If I were to use a SIP phone to call in, it works much better.</DIV>> <DIV> </DIV> > <DIV>Is there a way to make Asterisk to regenerate the DTMF tones toimprove the DTMF tones? Such as making it interpret the DTMF tones and regenerate it w/ a certain length regardless of original signal length. The reason I want to DTMF comes to AnalogPhone clearly is because I want to ultimately connect it to a FXSFXO converter and go back out to PSTN line.</DIV>> <DIV> </DIV> > <DIV>Thank you</DIV> > <DIV>Ron</DIV> > --0-1376241818-1080711919=:58147-- > > --__--__-- > > Message: 3 > From: Tilghman Lesher <tilghman@mail.jeffandtilghman.com> > Subject: Re: [Asterisk-Users] Caller entered digits ignored duringwait....> Date: Tue, 30 Mar 2004 23:46:13 -0600 > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > > > On 2004 Mar 30, at 20:56, Gene Kochanowsky wrote: > > > How would you use the t extension to accomplish this? > > exten => s,1,Wait(1) > exten => s,2,Answer > exten => s,3,SetVar(loopCnt=0) > exten => s,4,Background(welcome) > exten => s,5,Background(parties) > > exten => t,1,SetVar(loopCnt=$[${loopCnt} + 1]) > exten => t,2,GotoIf($[${loopCnt} < 3]?s|4) > exten => t,3,Background(vm-goodbye) > exten => t,4,Hangup > > -Tilghman > > > --__--__-- > > Message: 4 > Subject: Re: [Asterisk-Users] Sipcall.co.uk & [*] > From: Dave Cotton <dcotton@linuxautrement.com> > To: Asterisk List <asterisk-users@lists.digium.com> > Date: Wed, 31 Mar 2004 08:19:26 +0200 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-03-31 at 01:33, Matt wrote: > > Hello all. > > > > Has anyone managed to get SIPCALL.co.uk's service working with the [*]box?> > > > I've managed to register with other SIP providers but not SIPcall. > > > I spent a lot of time trying to get * to connect to SIPcall, I even got > directly in contact with the support depart of the supplier of the > hardware, who informed me that it is because * does not handle SIP > correctly, as I had no trouble connecting to SIPPhone, Nikotel, VoIPTalk > etc I decided to drop it. > YMMV > > -- > Dave Cotton <dcotton@linuxautrement.com> > > > --__--__-- > > Message: 5 > From: <clive18@webmail.co.za> > Subject: Re: [Asterisk-Users] IAX2 trunk mode over satellite > To: asterisk-users@lists.digium.com > Date: Wed, 31 Mar 2004 08:27:59 +0200 > Reply-To: asterisk-users@lists.digium.com > > Hi > > I have even used H323 over satelite, and beside the lagg, > no trouble. My only issue is the jitter buffer on IAX2 > seems to be broken. On a very jittery connection, I can > hardly make a decent call on IAX2. > > Good luck! > regards > Clive > > > On Tue, 30 Mar 2004 10:15:52 -0800 > John Todd <jtodd@loligo.com> wrote: > > > > Today has been the day for satellite questions, > > apparently, so I'll proxy one out to the rest of the > > community... I asked this tangentially a month or two > > ago, but I'll put it in a more blunt way: > > > > If you have IAX2 trunking mode experience over satellite, > > please let us know your experiences with that > > protocol/transport combination. > > > > I've got several people asking about IAX2 and trunk mode > > over satellite. I have not experimented with IAX2 over > > satellite (though I have used IAX1 over satellite) and > > I'm wondering if anyone has direct experiences with > > IAX2's jitter buffer control over such long-latency > > connections. > > > > I've had SIP working very well over satellite (despite > > what some people have found to the contrary on this list) > > and other than the lag there have been no issues that > > have come up on a reasonably-managed satellite segment. > > However, the IP overhead really starts to cost > > significant amounts of pennies when you add it up on > > multiple SIP RTP sessions over the same link. Plus, > > packet contention and buffering may (_may_) be an issue > > when pushing multiple simultaneous streams out the same > > transponder. > > > > It would seem to me that IAX2 in trunk mode would be > > optimal for people on very expensive satellite bandwidth, > > as a G.729 9.6kbps channel starts to actually look like > > 9.6kbps instead of 24kbps. However, I have had mixed > > success with IAX2 in certain circumstances. Before I > > start to ask for favors and get satellite time for > > testing, I'd like to see if anyone else has performed > > this experiment. If you'd wish to remain anonymous, > > please mail me directly and I'll appropriately trim > > identity information and re-distribute, or re-write as > > appropriate. > > > > Other hints I have heard/used on VoIP over satellite: > > - use small transmit cell (packet) sizes on your > > satellite gear > > - turn off error correction (why use it for VoIP?) > > - turn off compression (G.729 is already compressed; > > you ARE using > > G.729, right?) > > - ensure minimal latency on the terrestrial portions > > of the call > > - tell your users to suck it up and deal with the > > half-second lag > > > > JT > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > __________________________________________________________________________ > http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price > > --__--__-- > > Message: 6 > Date: Wed, 31 Mar 2004 16:37:06 +1000 > From: "Simon Brown" <Simon.Brown@otterson.com.au> > To: <asterisk-users@lists.digium.com> > Subject: [Asterisk-Users] Register vith SIP provider from behind NAT > Reply-To: asterisk-users@lists.digium.com > > I cannot successfully register with, or even make calls to, a SIP > provider > (such as FWD) with my * server sitting behind a NAT. The firewall is a > Cisco > 827 router running 12.3 IOS. > > Has anyone successfully got their server behind NAT to register or make > a > call to a SIP provider? > > TIA=20 > > Simon > > ----- > This mail was content checked for malicious code and viruses > by GFI MailSecurity. > > > --__--__-- > > Message: 7 > From: Dean <aster@zanadoo.net> > To: asterisk-users@lists.digium.com > Organization: > Date: 31 Mar 2004 00:28:06 -0800 > Subject: [Asterisk-Users] Can't talk on Cisco VIP 30 using Chan Skinny > Reply-To: asterisk-users@lists.digium.com > > I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like > to use with asterisk, I have set them up using chan_skinny. The phones > work well, except the only problem is that it is like the cisco phones > are muted. When I talk on the cisco phones I can hear my self through > the ear peice, but the person who I am calling can not hear me at all. I > have tried various cisco phones from various sources on 2 different > linux computers (one running redhat 7.3 and one running redhat 9) I have > tried using the 0.7.2 code and the latest development code from the CVS > and I still get the same results. All help will be greatly appreciated. > Below is the error log and my skinny.conf > > Thanks, > > Dean > > Error Log: > > Mar 31 00:09:29 WARNING[1024]: Ignoring port for now > Mar 31 00:09:29 WARNING[10251]: Read error on sound device: Resource > temporarily unavailable > Mar 31 00:09:29 WARNING[1024]: Ignoring rxwink > Mar 31 00:16:48 WARNING[1024]: Ignoring port for now > Mar 31 00:16:48 WARNING[10251]: Read error on sound device: Resource > temporarily unavailable > Mar 31 00:16:48 WARNING[1024]: Ignoring rxwink > Mar 31 00:22:30 WARNING[16401]: No audio available on > Skinny/133@flex-6?? > > > Console: > > skinny_answer(Skinny/133@flex-6) on 133@flex-6 > Recieved Open Recieve Channel Ack > us port: 17874 > sin port: 53316 > -- Playing 'voicemail/default/1234/unavail' (language 'en') > -- Playing 'vm-intro' (language 'en') > -- Playing 'beep' (language 'en') > -- x=0, open writing: > /var/spool/asterisk/voicemail/default/1234/INBOX/msg0004 format: wav49, > 0x80dc958 > -- x=1, open writing: > /var/spool/asterisk/voicemail/default/1234/INBOX/msg0004 format: gsm, > 0x811a3f8 > -- x=2, open writing: > /var/spool/asterisk/voicemail/default/1234/INBOX/msg0004 format: wav, > 0x811a570 > Mar 31 00:22:30 WARNING[16401]: app_voicemail.c:1222 play_and_record: No > audio available on Skinny/133@flex-6?? > -- User hung up > == Spawn extension (demo, 1235, 1) exited non-zero on > 'Skinny/133@flex-6' > skinny_hangup(Skinny/133@flex-6) on 133@flex > > Error Log > > Mar 31 00:09:29 WARNING[1024]: Ignoring port for now > Mar 31 00:09:29 WARNING[10251]: Read error on sound device: Resource > temporarily unavailable > Mar 31 00:09:29 WARNING[1024]: Ignoring rxwink > Mar 31 00:16:48 WARNING[1024]: Ignoring port for now > Mar 31 00:16:48 WARNING[10251]: Read error on sound device: Resource > temporarily unavailable > Mar 31 00:16:48 WARNING[1024]: Ignoring rxwink > Mar 31 00:22:30 WARNING[16401]: No audio available on > Skinny/133@flex-6?? > > ; > ; Skinny Configuration for Asterisk > ; > [general] > port = 2000 ; Port to bind to, default tcp/2000 > bindaddr = 0.0.0.0 ; Address to bind to > dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) > keepAlive = 120 > > ;allow = all > ;disallow > > > ; Typical config for 12SP+ > [florian] > device=SEPXXXXXXXXXXXX > version=P002G204 ; Thanks critch > context=demo > line => 120 ; Dial(Skinny/120@florian) > > ; Typical config for 12SP+ > [florianx] > device=SEPXXXXXXXXXXXX > version=P0020301003 ; Thanks critch > context=default > line => 122 ; Dial(Skinny/120@florian) > > ; Typical config for 12SP+ > [flex] > device=SEPXXXXXXXXXXXX > version=P002F202 > context=demo > line => 133 > > > > --__--__-- > > Message: 8 > Date: Wed, 31 Mar 2004 09:08:53 +0200 > To: asterisk-users@lists.digium.com > From: Stig Andersson <stig@ymex.se> > Subject: Re: [Asterisk-Users] Caller entered digits ignored during > wait.... > Reply-To: asterisk-users@lists.digium.com > > Asterisk doesn't accept keys during wait, use Background > and play 1 sec silence instead. > > /Stig > > At 23:46 2004-03-30 -0600, you wrote: > > > >On 2004 Mar 30, at 20:56, Gene Kochanowsky wrote: > > > >> How would you use the t extension to accomplish this? > > > >exten => s,1,Wait(1) > >exten => s,2,Answer > >exten => s,3,SetVar(loopCnt=0) > >exten => s,4,Background(welcome) > >exten => s,5,Background(parties) > > > >exten => t,1,SetVar(loopCnt=$[${loopCnt} + 1]) > >exten => t,2,GotoIf($[${loopCnt} < 3]?s|4) > >exten => t,3,Background(vm-goodbye) > >exten => t,4,Hangup > > > >-Tilghman > > > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -------------------------------------------------------------------------------------> N Y H E T E R! > - Internetaccess (Modem/ISDN64+128 via Ymex - utan abonnemangskostnad!!! > ONLINE-registrering p? www.ymex.se > - Uppringd SMTP, slut p? 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Mailcoach V2.27 - The business E-mail solution.http://www.mailcoach.com/> -------------------------------------------------------------------------------------> Ymex AB| Alv?gen 7 | 871 52 H?rn?sand | Sweden | http://www.ymex.se/ > > --__--__-- > > Message: 9 > From: "jc" <asterisk-user@christoffersonrobb.com> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] Exception flag set - snom200 > Date: Wed, 31 Mar 2004 08:16:07 +0100 > Reply-To: asterisk-users@lists.digium.com > > This is a multi-part message in MIME format. > > ------=_NextPart_000_007D_01C416F8.6B262920 > Content-Type: text/plain; > charset="us-ascii" > Content-Transfer-Encoding: 7bit > > Asterisk CVS-03/11/04 18:18:12 > > snom200-SIP 2.03o > > > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ernest W. > Lessenger > Sent: Wednesday, March 31, 2004 1:01 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Exception flag set - snom200 > > > > What version of asterisk are you using, and what version of the SNOM > firmware? > > > > --Ernest > > > > > _____ > > > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of jc > Sent: Tuesday, March 30, 2004 10:20 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Exception flag set - snom200 > > Sorry I forgot the subject in the last post. > > > > When my snom200 receives an inbound SIP external sip call, it somehow > rejects the call and with a busy tone. The debug shows the following > error: > > > > channel.c:1142 ast_read: Exception flag set on 'SIP/sipphone-7796', but > no exception handler > > > > > > what does this mean and how can I debug it further?? > > > > Thanks > > JC > > > > > ------=_NextPart_000_007D_01C416F8.6B262920 > Content-Type: text/html; > charset="us-ascii" > Content-Transfer-Encoding: quoted-printable > > <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> > <html> > > <head> > <META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; > charset=3Dus-ascii"> > > > <meta name=3DGenerator content=3D"Microsoft Word 10 (filtered)"> > > <style> > <!-- > /* Font Definitions */ > @font-face > {font-family:Tahoma; > panose-1:2 11 6 4 3 5 4 4 2 4;} > /* Style Definitions */ > p.MsoNormal, li.MsoNormal, div.MsoNormal > {margin:0in; > margin-bottom:.0001pt; > font-size:12.0pt; > font-family:"Times New Roman";} > a:link, span.MsoHyperlink > {color:blue; > text-decoration:underline;} > a:visited, span.MsoHyperlinkFollowed > {color:purple; > text-decoration:underline;} > span.emailstyle19 > {font-family:Arial; > color:windowtext;} > span.emailstyle20 > {font-family:Arial; > color:navy;} > span.EmailStyle21 > {font-family:Arial; > color:navy;} > @page Section1 > {size:8.5in 11.0in; > margin:99.35pt 1.25in 83.5pt 1.25in;} > div.Section1 > {page:Section1;} > --> > </style> > > </head> > > <body lang=3DEN-US link=3Dblue vlink=3Dpurple> > > <div class=3DSection1> > > <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span > lang=3DSV > style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Asterisk > CVS-03/11/04 > 18:18:12 </span></font></p> > > <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span > lang=3DSV > style=3D'font-size:10.0pt;font-family:Arial;color:navy'>snom200-SIP > 2.03o</span></font></p> > > <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span > lang=3DSV > style=3D'font-size:10.0pt;font-family:Arial;color:navy'> </span></fo> nt></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 > face=3DTahoma><span > style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original > Message-----<br> > <b><span style=3D'font-weight:bold'>From:</span></b> > asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] <b><span > style=3D'font-weight: > bold'>On Behalf Of </span></b>Ernest W. Lessenger<br> > <b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March > 31, 2004 > 1:01 AM<br> > <b><span style=3D'font-weight:bold'>To:</span></b> > asterisk-users@lists.digium.com<br> > <b><span style=3D'font-weight:bold'>Subject:</span></b> RE: > [Asterisk-Users] > Exception flag set - snom200</span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 > face=3D"Times New Roman"><span > style=3D'font-size:12.0pt'> </span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 > color=3Dblue face=3DArial><span > style=3D'font-size:10.0pt;font-family:Arial;color:blue'>What version of > asterisk > are you using, and what version of the SNOM firmware?</span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 > face=3D"Times New Roman"><span > style=3D'font-size:12.0pt'> </span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 > color=3Dblue face=3DArial><span > style=3D'font-size:10.0pt;font-family:Arial;color:blue'>--Ernest</span></> font></p> > > <blockquote style=3D'border:none;border-left:solid blue > 1.5pt;padding:0in 0in 0in 4.0pt; > margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'> > > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 > face=3D"Times New Roman"><span > style=3D'font-size:12.0pt'> </span></font></p> > > <div class=3DMsoNormal align=3Dcenter > style=3D'margin-left:.5in;text-align:center'><font > size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'> > > <hr size=3D2 width=3D"100%" align=3Dcenter> > > </span></font></div> > > <p class=3DMsoNormal > style=3D'margin-right:0in;margin-bottom:12.0pt;margin-left: > .5in'><b><font size=3D2 face=3DTahoma><span > style=3D'font-size:10.0pt;font-family: > Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2 > face=3DTahoma><span > style=3D'font-size:10.0pt;font-family:Tahoma'> > asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] <b><span > style=3D'font-weight: > bold'>On Behalf Of </span></b>jc<br> > <b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, March 30, > 2004 > 10:20 AM<br> > <b><span style=3D'font-weight:bold'>To:</span></b> > asterisk-users@lists.digium.com<br> > <b><span style=3D'font-weight:bold'>Subject:</span></b> [Asterisk-Users] > Exception flag set - snom200</span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 > color=3Dnavy face=3DArial><span > style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Sorry I forgot > the > subject in the last post.</span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 > face=3D"Times New Roman"><span > style=3D'font-size:12.0pt'> </span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 > color=3Dnavy face=3DArial><span > style=3D'font-size:10.0pt;font-family:Arial;color:navy'>When my snom200 > receives > an inbound SIP external sip call, it somehow rejects the call and with a > busy > tone. The debug shows the following error:</span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 > face=3D"Times New Roman"><span > style=3D'font-size:12.0pt'> </span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 > color=3Dnavy face=3DArial><span > style=3D'font-size:10.0pt;font-family:Arial;color:navy'>channel.c:1142 > ast_read: > Exception flag set on 'SIP/sipphone-7796', but no exception > handler</span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 > face=3D"Times New Roman"><span > style=3D'font-size:12.0pt'> </span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 > face=3D"Times New Roman"><span > style=3D'font-size:12.0pt'> </span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 > color=3Dnavy face=3DArial><span > style=3D'font-size:10.0pt;font-family:Arial;color:navy'>what does this > mean and > how can I debug it further??</span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 > face=3D"Times New Roman"><span > style=3D'font-size:12.0pt'> </span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 > color=3Dnavy face=3DArial><span > style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Thanks > </span></font></p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 > color=3Dnavy face=3DArial><span > style=3D'font-size:10.0pt;font-family:Arial;color:navy'>JC</span></font><> /p> > > <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 > face=3D"Times New Roman"><span > style=3D'font-size:12.0pt'> </span></font></p> > > </blockquote> > > </div> > > </body> > > </html> > > ------=_NextPart_000_007D_01C416F8.6B262920-- > > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest >
Steve Underwood
2004-Mar-31 06:49 UTC
[Asterisk-Users] RE: RxFax/spandsp: not disconnecting
Reynaldo Simbulan wrote:>Hi Steve, > >I am having this problem in which RxFax is still holding the file after >receiving a complete fax. Somehow the zap channel is still active but on the >fax client it was sent successfully. >If you call the line it is still busy. > > >There are disconnect issues if the line lacks answer supervision. I am working on that now. Regards, Steve
Steve Underwood
2004-Mar-31 09:01 UTC
[Asterisk-Users] RE: RxFax/spandsp: not disconnecting
Reynaldo Simbulan wrote:>Hi Steve, > >I am having this problem in which RxFax is still holding the file after >receiving a complete fax. Somehow the zap channel is still active but on the >fax client it was sent successfully. >If you call the line it is still busy. > > >spandsp-0.0.1k.tar.gz and updated app_rxfax.c and app_txfax.c files are available for download. They address this disconnect issue, and have a few other minor tweaks. Regards, Steve