Francisco Perez-Landaeta
2004-Nov-04 06:50 UTC
[Asterisk-Users] BROADVOICE fails to register
I have the following registration information for broadvoice, but it fails to register. Can anyone tell me what is the problem ? I am not behind a firewall and I have added this below the [general] Thanks, My number and password is hidded. I received from broadvoice the following : primary_dns_ip: 147.135.0.6 secondary_dns_ip: 147.138.8.6 proxy_ip: proxy.broadvoice.com proxy_port: 5060 registrar_ip: sip.broadvoice.com registrar_port: 5060 phone_number: 954XXXXXXX auth_id: 954XXXXXXX auth_password: password tftp_ip: config.broadvoice.com ntp_ip: ntp.nasa.gov DTMF: InBand registration_time_out: 10_seconds voice_mail_key: *86 BELOW IS WHAT I TOOK FROM THE WIKI, EVEN IF I CHANGE THE IP ADDRESSES IT FAILS TO REGISTER. sip.conf Under the general context: ;externip=<IP or Dynamic DNS Name> ; Needed if behind NAT context=incoming dtmfmode=inband register => 954XXXXXXX:password@147.135.0.129 [Broadvoice] type=peer username=954XXXXXXX fromuser=954XXXXXXX secret=password host=147.135.0.129 context=incoming fromdomain=sip.broadvoice.com nat=yes canreinvite=no dtmfmode=inband extensions.conf For incoming calls [incoming] ;This extension line will ring SIP ;extension 2001 for 60 seconds then hang up. Modify as necessary to fit your dialplan exten => s,1,Dial(SIP/2001,60,tr) exten => s,2,hangup extensions.conf For outgoing calls: [toll-trunks] ;Pattern match for local call plan, use appropriate pattern if on nationwide plan. exten => _9NXXXXXX,1,Dial(SIP/${EXTEN:1}@broadvoice) exten => _9NXXXXXX,2,Congestion ; ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, November 03, 2004 7:01 PM Subject: Asterisk-Users Digest, Vol 4, Issue 52> 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-owner@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: ISDN Dialplan (Paulo Adriano) > 2. Re: Sip clients not longer registering (David Filion) > 3. RE: Installing X100P Asterisk - Unable to createchannel of > type 'Zap' (Vikas Deolaliker) > 4. MusicOnhold on Bridged calls (Paulo Adriano) > 5. Re: Automatically restart asterisk if not running > (Brancaleoni Matteo) > 6. What do I need to ask my T1 supplier? (Scott Nelson) > 7. getting cid from spa3k pstn to * (Randy Bush) > 8. How change default law for T100P (Manuel Marin) > 9. Re: What do I need to ask my T1 supplier? (niles@atheos.net) > 10. Re: What do I need to ask my T1 supplier? (TC) > 11. Cisco 79XX - Using built-in 3way conference (Matthew Boehm) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 03 Nov 2004 22:06:10 +0000 > From: "Paulo Adriano" <pauloadriano@wavelis.pt> > Subject: Re: [Asterisk-Users] ISDN Dialplan > To: <asterisk-users@lists.digium.com>, <psvasterisk@psv.nu> > Cc: mje@posix.co.za > Message-ID: <s1895669.024@wavelis> > Content-Type: text/plain; charset="us-ascii" > > It*s solved now. You are right it was a matter of formating the > outgoing number. > > Thanks > > Paulo > > >>>psvasterisk@psv.nu 11/03 1:34 pm >>> > > On Wed, 3 Nov 2004, Paulo Adriano wrote: > > > >After trying that syntax I still have the same problem, one thing is > > >very strange is the number that Asterisk reports as the incoming. My > > >ESN*s numbers are 219898334 and 219898335 but on the console I see > > >219898334,1 and 219898335,1 > > > Are you sure it is not trying to tell you extension 219898334 and > priority > > 1 in the dialplan? > > > [snip} > > > >THIS IS AN OUTGOING CALL TRY TO NUMBER 213570150 MY DIALPLAN > REQUIRES 9 to go outside > > > > > >*CLI> -- Executing Dial(SIP/21-9da8, Modem/g1/9213570150||tr) in > new stack > > >Nov 3 18:37:35 WARNING[1110502320]: chan_modem.c:191 modem_call: > Destination g1/9213570150 requres a real destination > (device:destination) > > > -- Couldn't call g1/9213570150 > > > -- Hungup 'Modem[i4l]/ttyI1' > > > == Everyone is busy/congested at this time > > >Nov 3 18:37:45 WARNING[1110502320]: pbx.c:1933 ast_pbx_run: Timeout, > but no rule 't' in context 'local-access' > > > -- Saved useragent SJLabs-SJphone/1.30.248 for peer 21 > > > The source for chan_modem suggests that the dial string should be > > formatted like > > Dial(Modem/g1:9213570150||tr) > > > Peter > > > > > > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL:http://lists.digium.com/pipermail/asterisk-users/attachments/20041103/d471c6ed/attachment-0001.html> > ------------------------------ > > Message: 2 > Date: Wed, 03 Nov 2004 17:11:39 -0500 > From: David Filion <dfilion@dotality.com> > Subject: Re: [Asterisk-Users] Sip clients not longer registering > To: asterisk-users@lists.digium.com > Message-ID: <4189579B.4030804@dotality.com> > Content-Type: text/plain; charset=us-ascii; format=flowed > > > > Message: 8 > Date: Wed, 03 Nov 2004 22:32:46 +0100 > From: "Olle E. Johansson" <oej@edvina.net> > Subject: Re: [Asterisk-Users] Sip clients not longer registering > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <41894E7E.1060006@edvina.net> > Content-Type: text/plain; charset=us-ascii; format=flowed > > David Filion wrote: > > > > >> Hi, > >> > >> We have been using Asterisk since version 0.9x with little or no > >> problems. However, for an unknow reasons, our sip clients can nolonger > >> register. We updated to Asterisk 1.0.2 hoping that would solve the > >> problem, but no luck. > >> > > > > > There is a chance that our change NAT logic is problematic in yournetwork.> (rport support) > > check the sample sip.conf for various options to nat= and try them. > It's in configs/sip.conf.sample > > /O > > > > Thanks, I'll give them a try. > > David > > > > > ------------------------------ > > Message: 3 > Date: Wed, 3 Nov 2004 14:15:49 -0800 > From: "Vikas Deolaliker" <vikasd@yahoo.com> > Subject: RE: [Asterisk-Users] Installing X100P Asterisk - Unable to > createchannel of type 'Zap' > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <20041103221544.2A0F82FE4ED@lists.digium.com> > Content-Type: text/plain; charset="us-ascii" > > > Look at your logs in /var/logs/asterisk/. I am pretty certain it is afault> in your /etc/asterisk/Zapata.conf file. > > Vikas > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of SethRemington> Sent: Wednesday, November 03, 2004 1:01 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Installing X100P Asterisk - Unable to > createchannel of type 'Zap' > > On Wed, 2004-11-03 at 13:02, Frank Kostin wrote: > > Hello list, > > I am trying to install a Digium X100P into a Redhat Asterisk. > > Kernel seems to be OK, card OK. > > Zaptel Configuration seems to be OK. > > # ztcfg -vv > > Channel map: > > Channel 01: FXS Kewlstart (Default) (Slaves: 01) > > 1 channels configured. > > > > Asterisk works fine with IP SIP but not with X100P > > I get the error on Asterisk CLI> > > .... channel.c:1919 ast_request: "No channel type registered for > > 'Zap'" > > and than ....app_dial.c:763 dial_exec "Unable to create channel of > > type 'Zap'". > > > > Does anyone know what might be the problem ? > > Thanks for any help > > Did you: > > modprobe zaptel > modprobe wcfxo > > Also make sure that you compiled and installed zaptel *before* you > installed asterisk. If you did it afterwords simply re-compile asterisk > and you should be good. > > -Seth > > -- > Seth Remington > SaberLogic, LLC > 661-B Weber Drive > Wadsworth, Ohio 44281 > Phone: (330)335-6442 > Fax: (330)336-8559 > > _______________________________________________ > 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 > > > > > ------------------------------ > > Message: 4 > Date: Wed, 03 Nov 2004 22:27:23 +0000 > From: "Paulo Adriano" <pauloadriano@wavelis.pt> > Subject: [Asterisk-Users] MusicOnhold on Bridged calls > To: <asterisk-users@lists.digium.com> > Message-ID: <s1895b59.027@wavelis> > Content-Type: text/plain; charset="us-ascii" > > Now that my bridged calls are working fine with ISDN I have a question ? > > > When my customers call in and my ext is not available the call is routed > out to my mobile. > > Everything works but I would like to know if there is a way of having > the calling sign (tone) always on . With my current config the origin > caller phone hear a calling beeping tone only until the call is passed > on a bridged call. Then and until the end user (in this case , myself in > the cellular phone) answers the call , the origin has a big silence. > This may cause him to think that something append during the call. > > How can I configure * in order to avoid the calling tone from > desapearing on a bridged call. Even music would be acceptable as the > call is being tranfered to a pstn number. > > Thanks in advance > Paulo > > Francisco Paulo Adriano > WaveLIS LDA > Mobile +351 91 870 87 98 > Office + 351 21 989 83 34 > Fax +351 21 989 83 35 > E-mail : pauloadriano@wavelis.pt > > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL:http://lists.digium.com/pipermail/asterisk-users/attachments/20041103/34b4f9c5/attachment-0001.html> > ------------------------------ > > Message: 5 > Date: Wed, 03 Nov 2004 23:28:04 +0100 > From: Brancaleoni Matteo <mbrancaleoni@espia.it> > Subject: Re: [Asterisk-Users] Automatically restart asterisk if not > running > To: Matthew Marlowe <matthew.marlowe@gmail.com>, Asterisk Users > Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Cc: asterisk-users@lists.digium.com > Message-ID: <1099520884.11249.2.camel@athlon64> > Content-Type: text/plain > > Hi > Il mer, 2004-11-03 alle 21:41, Matthew Marlowe ha scritto: > > I once found a script, I think it was on the mailing list that would > > run in crontab and restart asterisk if not running... Does anyone > > happen to have a copy of that? > > I suggest you to use something like a superdaemon, > ie a process (normally spawned by init) that > checks other processes. > I suggest you monit (http://www.tildeslash.com/monit/), > it can monitor processes in various ways and other > vital system informations. I'm pretty happy > with it (using it in all my * installations) > > Matteo > -- > Brancaleoni Matteo <mbrancaleoni@espia.it> > Espia Srl > > > > ------------------------------ > > Message: 6 > Date: Wed, 3 Nov 2004 17:32:27 -0500 > From: Scott Nelson <asterisk@nelson.saint-louis.mo.us> > Subject: [Asterisk-Users] What do I need to ask my T1 supplier? > To: asterisk-users@lists.digium.com > Message-ID: <200411031632.27499.asterisk@nelson.saint-louis.mo.us> > Content-Type: text/plain; charset="us-ascii" > > My employer is switching to a new T1 supplier (it was AT&T, we are nowgoing> with XO), and sometime in the future we want to replace our PBX with an > Asterisk system. > > What do I need to know to make sure the T1 line is "provisioned" (is thatthe> right term?) correctly for a Digium T100P/TE410P/TE405P? > > They will split the T1 line into 10 channels of voice and 14 channels ofdata.> >From what I understand, they will terminate the T1 into a channel bank,and> then from that give is 10 POTS phone jacks and one data port (to go to an > Adtran router for our Internet access). > > Any comments and/or suggestions? > > Scott > > > ------------------------------ > > Message: 7 > Date: Wed, 3 Nov 2004 14:40:29 -0800 > From: Randy Bush <randy@psg.com> > Subject: [Asterisk-Users] getting cid from spa3k pstn to * > To: splatters <asterisk-users@lists.digium.com> > Message-ID: <16777.24157.228132.850877@ran.psg.com> > Content-Type: text/plain; charset=us-ascii > > i am still going crazy with this one. i can not get callerid from a call > received on the spa3k pstn to asterisk. THIS USED TO WORK! > > in order to get the cid from the spa3k to *, i need to turn on > PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES > > the sip.conf entry looks like > [spa3k] > type=friend > host=dynamic > port=5061 > auth=md5 > secret=hidden > qualify=1000 > dtmfmode=rfc2833 > canreinvite=yes > context=spa3k-ext > > this produces a sip exchange as follows: > > No. Time Source DestinationProtocol Info> 1 0.000000 spa3k asterisk.foo.edu SIP/SDP Request: INVITEsip:105@asterisk.foo.edu, with session description> > Frame 1 (1095 bytes on wire, 1095 bytes captured) > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu(asterisk-ip-ad)> User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) > Session Initiation Protocol > Request-Line: INVITE sip:105@asterisk.foo.edu SIP/2.0 > Message Header > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-f9456447 > From: CallerName<sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1> SIP Display info: CallerName > SIP from address: sip:2065551212@asterisk.foo.edu > SIP tag: 54e649b356424567o1 > To: <sip:105@asterisk.foo.edu> > SIP to address: sip:105@asterisk.foo.edu > Remote-Party-ID: CallerName<sip:2065551212@asterisk.foo.edu>;screen=yes;party=calling> Call-ID: 51efe8a3-2d73b337@spa3k > CSeq: 101 INVITE > Max-Forwards: 70 > Contact: biwa 0431 <sip:biwaa1-in@spa3k:5061> > Expires: 240 > User-Agent: Sipura/SPA3000-2.0.11(GWa) > Content-Length: 430 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura > Content-Type: application/sdp > Message body > Session Description Protocol > > No. Time Source DestinationProtocol Info> 2 0.000514 asterisk.foo.edu spa3k SIP Status: 407 ProxyAuthentication Required> > Frame 2 (520 bytes on wire, 520 bytes captured) > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), DstAddr: spa3k (spa3k)> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5061 (5061) > Session Initiation Protocol > Status-Line: SIP/2.0 407 Proxy Authentication Required > Message Header > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-f9456447 > From: CallerName<sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1> SIP Display info: CallerName > SIP from address: sip:2065551212@asterisk.foo.edu > SIP tag: 54e649b356424567o1 > To: <sip:105@asterisk.foo.edu>;tag=as741941ff > SIP to address: sip:105@asterisk.foo.edu > SIP tag: as741941ff > Call-ID: 51efe8a3-2d73b337@spa3k > CSeq: 101 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:105@asterisk-ip-ad> > Proxy-Authenticate: Digest realm="asterisk", nonce="263c07e5" > Content-Length: 0 > > No. Time Source DestinationProtocol Info> 3 0.090441 spa3k asterisk.foo.edu SIP Request: ACKsip:105@asterisk.foo.edu> > Frame 3 (453 bytes on wire, 453 bytes captured) > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu(asterisk-ip-ad)> User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) > Session Initiation Protocol > Request-Line: ACK sip:105@asterisk.foo.edu SIP/2.0 > Message Header > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-f9456447 > From: CallerName<sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1> SIP Display info: CallerName > SIP from address: sip:2065551212@asterisk.foo.edu > SIP tag: 54e649b356424567o1 > To: <sip:105@asterisk.foo.edu>;tag=as741941ff > SIP to address: sip:105@asterisk.foo.edu > SIP tag: as741941ff > Call-ID: 51efe8a3-2d73b337@spa3k > CSeq: 101 ACK > Max-Forwards: 70 > Contact: biwa 0431 <sip:biwaa1-in@spa3k:5061> > User-Agent: Sipura/SPA3000-2.0.11(GWa) > Content-Length: 0 > > No. Time Source DestinationProtocol Info> 4 0.135913 spa3k asterisk.foo.edu SIP/SDP Request: INVITEsip:105@asterisk.foo.edu, with session description> > Frame 4 (1265 bytes on wire, 1265 bytes captured) > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu(asterisk-ip-ad)> User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) > Session Initiation Protocol > Request-Line: INVITE sip:105@asterisk.foo.edu SIP/2.0 > Message Header > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-e2744867 > From: CallerName<sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1> SIP Display info: CallerName > SIP from address: sip:2065551212@asterisk.foo.edu > SIP tag: 54e649b356424567o1 > To: <sip:105@asterisk.foo.edu> > SIP to address: sip:105@asterisk.foo.edu > Remote-Party-ID: CallerName<sip:2065551212@asterisk.foo.edu>;screen=yes;party=calling> Call-ID: 51efe8a3-2d73b337@spa3k > CSeq: 102 INVITE > Max-Forwards: 70 > Proxy-Authorization: Digestusername="biwaa1-in",realm="asterisk",nonce="263c07e5",uri="sip:105@asterisk foo.edu",algorithm=MD5,response="f8e02292686b3b5cb2117186b1474ba9"> Contact: biwa 0431 <sip:biwaa1-in@spa3k:5061> > Expires: 240 > User-Agent: Sipura/SPA3000-2.0.11(GWa) > Content-Length: 430 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura > Content-Type: application/sdp > Message body > Session Description Protocol > > No. Time Source DestinationProtocol Info> 5 0.136261 asterisk.foo.edu spa3k SIP Status: 403Forbidden> > Frame 5 (437 bytes on wire, 437 bytes captured) > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), DstAddr: spa3k (spa3k)> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5061 (5061) > Session Initiation Protocol > Status-Line: SIP/2.0 403 Forbidden > Message Header > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-e2744867 > From: CallerName<sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1> SIP Display info: CallerName > SIP from address: sip:2065551212@asterisk.foo.edu > SIP tag: 54e649b356424567o1 > To: <sip:105@asterisk.foo.edu>;tag=as741941ff > SIP to address: sip:105@asterisk.foo.edu > SIP tag: as741941ff > Call-ID: 51efe8a3-2d73b337@spa3k > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:105@asterisk-ip-ad> > Content-Length: 0 > > No. Time Source DestinationProtocol Info> 6 0.383761 spa3k asterisk.foo.edu SIP Request: ACKsip:105@asterisk.foo.edu> > Frame 6 (623 bytes on wire, 623 bytes captured) > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu(asterisk-ip-ad)> User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) > Session Initiation Protocol > Request-Line: ACK sip:105@asterisk.foo.edu SIP/2.0 > Message Header > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-e2744867 > From: CallerName<sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1> SIP Display info: CallerName > SIP from address: sip:2065551212@asterisk.foo.edu > SIP tag: 54e649b356424567o1 > To: <sip:105@asterisk.foo.edu>;tag=as741941ff > SIP to address: sip:105@asterisk.foo.edu > SIP tag: as741941ff > Call-ID: 51efe8a3-2d73b337@spa3k > CSeq: 102 ACK > Max-Forwards: 70 > Proxy-Authorization: Digestusername="biwaa1-in",realm="asterisk",nonce="263c07e5",uri="sip:105@asterisk foo.edu",algorithm=MD5,response="c33e3a4bab8eef38ca12b9ddf192b796"> Contact: biwa 0431 <sip:biwaa1-in@spa3k:5061> > User-Agent: Sipura/SPA3000-2.0.11(GWa) > Content-Length: 0 > > No. Time Source DestinationProtocol Info> 7 7.079655 asterisk.foo.edu spa3k SIP Request: OPTIONSsip:spa3k> > Frame 7 (463 bytes on wire, 463 bytes captured) > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), DstAddr: spa3k (spa3k)> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) > Session Initiation Protocol > Request-Line: OPTIONS sip:spa3k SIP/2.0 > Message Header > Via: SIP/2.0/UDP asterisk-ip-ad:5060;branch=z9hG4bK1b040c15 > From: "Unknown" <sip:Unknown@asterisk-ip-ad>;tag=as3f547347 > SIP Display info: "Unknown" > SIP from address: sip:Unknown@asterisk-ip-ad > SIP tag: as3f547347 > To: <sip:spa3k> > SIP to address: sip:spa3k > Contact: <sip:Unknown@asterisk-ip-ad> > Call-ID: 159ec16b69ac62e334905b487158eeed@asterisk-ip-ad > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Date: Mon, 01 Nov 2004 17:34:49 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Length: 0 > > No. Time Source DestinationProtocol Info> 8 7.079766 asterisk.foo.edu spa3k SIP Request: OPTIONSsip:spa3k:5061> > Frame 8 (473 bytes on wire, 473 bytes captured) > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), DstAddr: spa3k (spa3k)> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5061 (5061) > Session Initiation Protocol > Request-Line: OPTIONS sip:spa3k:5061 SIP/2.0 > Message Header > Via: SIP/2.0/UDP asterisk-ip-ad:5060;branch=z9hG4bK29909a71 > From: "Unknown" <sip:Unknown@asterisk-ip-ad>;tag=as67500153 > SIP Display info: "Unknown" > SIP from address: sip:Unknown@asterisk-ip-ad > SIP tag: as67500153 > To: <sip:spa3k:5061> > SIP to address: sip:spa3k:5061 > Contact: <sip:Unknown@asterisk-ip-ad> > Call-ID: 2b80a2980a32bf7809b8648328ced971@asterisk-ip-ad > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Date: Mon, 01 Nov 2004 17:34:49 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Length: 0 > > No. Time Source Destination Protocol Info> 9 7.173099 spa3k asterisk.foo.edu SIP Status: 404 NotFound> > Frame 9 (361 bytes on wire, 361 bytes captured) > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu(asterisk-ip-ad)> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) > Session Initiation Protocol > Status-Line: SIP/2.0 404 Not Found > Message Header > To: <sip:spa3k>;tag=828c8dcf8cd9e760i0 > SIP to address: sip:spa3k > SIP tag: 828c8dcf8cd9e760i0 > From: "Unknown" <sip:Unknown@asterisk-ip-ad>;tag=as3f547347 > SIP Display info: "Unknown" > SIP from address: sip:Unknown@asterisk-ip-ad > SIP tag: as3f547347 > Call-ID: 159ec16b69ac62e334905b487158eeed@asterisk-ip-ad > CSeq: 102 OPTIONS > Via: SIP/2.0/UDP asterisk-ip-ad:5060;branch=z9hG4bK1b040c15 > Server: Sipura/SPA3000-2.0.11(GWa) > Content-Length: 0 > > note that the From: has the cid, as does the Remote-Party-ID:. and the > Contact: has the spa3k's id and display name. and > the From: and/or Remote-Party-ID: cause asterisk respond with 407 Proxy > Authentication Required, and things do not improve from there > > if i set the spa3k config to have > PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = NO > > Frame 1 (1072 bytes on wire, 1072 bytes captured) > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr:666.42.7.11 (666.42.7.11)> User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) > Session Initiation Protocol > Request-Line: INVITE sip:105@my.asterisk.su SIP/2.0 > Method: INVITE > Resent Packet: False > Message Header > Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-f5998d8a > From: spa3k pstn <sip:spa3k@my.asterisk.su>;tag=8fc58211a0dc60f2o1 > To: <sip:105@my.asterisk.su> > Remote-Party-ID: spa3k pstn<sip:spa3k@my.asterisk.su>;screen=yes;party=calling> Call-ID: daed83bd-b2b66b36@42.666.11.7 > CSeq: 101 INVITE > Max-Forwards: 70 > Contact: spa3k pstn <sip:biwaa1@42.666.11.7:5061> > Expires: 240 > User-Agent: Sipura/SPA3000-2.0.11(GWa) > Content-Length: 430 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura > Content-Type: application/sdp > Message body > Session Description Protocol > > the connection completes, but asterisk does not have the pstn caller id. > > randy > > > > ------------------------------ > > Message: 8 > Date: Wed, 3 Nov 2004 15:40:33 -0700 (MST) > From: Manuel Marin <mmg@transtelco.com.mx> > Subject: [Asterisk-Users] How change default law for T100P > To: asterisk-users@lists.digium.com > Message-ID: > <653967.1099521633455.SLOX.WebMail.wwwrun@iGrup.transtelco.com.mx> > Content-Type: text/plain; charset=us-ascii > > I would like to know if there is a way to change default ulaw for a T1 > card. I see in the zap show channel X that is working as ulaw. How do I > change it in zapata.conf or zaptel.conf to alaw. Iam interconnecting a > Meridian PBX but I need to configure it as alaw. > > > > > ------------------------------ > > Message: 9 > Date: Wed, 3 Nov 2004 17:40:55 -0500 > From: niles@atheos.net > Subject: Re: [Asterisk-Users] What do I need to ask my T1 supplier? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <6C40DB70-2DE9-11D9-B4EE-000A957899C8@atheos.net> > Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed > > > On Nov 3, 2004, at 5:32 PM, Scott Nelson wrote: > > > My employer is switching to a new T1 supplier (it was AT&T, we are now > > going > > with XO), and sometime in the future we want to replace our PBX with an > > Asterisk system. > > > > What do I need to know to make sure the T1 line is "provisioned" (is > > that the > > right term?) correctly for a Digium T100P/TE410P/TE405P? > > > > They will split the T1 line into 10 channels of voice and 14 channels > > of data. > > From what I understand, they will terminate the T1 into a channel > > bank, and > > then from that give is 10 POTS phone jacks and one data port (to go to > > an > > Adtran router for our Internet access). > > > > Any comments and/or suggestions? > > > > Scott > > _______________________________________________ > > Scott, > > you can skip the channel bank & router, and use asterisk with a T100P to > serve your data & voice. You can find all the info you need on the Wiki > http://www.voip-info.org/tiki-index.php? > page=Asterisk%20Data%20Configuration > > I use this setup for 11 voice channels and 256K of data from Nuvox. > Niles > > > > > ------------------------------ > > Message: 10 > Date: Wed, 03 Nov 2004 14:47:09 -0800 > From: TC <trclark@shaw.ca> > Subject: Re: [Asterisk-Users] What do I need to ask my T1 supplier? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <032601c4c1f7$0dd0ff20$c901a8c0@w2ktopcat> > Content-Type: text/plain; charset=iso-8859-1 > > > > >They will split the T1 line into 10 channels of voice and 14 channels of > data. > >From what I understand, they will terminate the T1 into a channel bank,and> >then from that give is 10 POTS phone jacks and one data port (to go to an > >Adtran router for our Internet access). > > >Any comments and/or suggestions? > > what would be realy nice from them is to present those 10 voice channels > as not POTS but as the first 10 channels of a pri t1 net interface ie a > fractional t1 voice > and skip the a/d nonsense I know an adit 600 with a router & t1 cards cando> that for you > > > > > ------------------------------ > > Message: 11 > Date: Wed, 3 Nov 2004 17:01:11 -0600 > From: "Matthew Boehm" <mboehm@cytelcom.com> > Subject: [Asterisk-Users] Cisco 79XX - Using built-in 3way conference > To: <asterisk-users@lists.digium.com> > Message-ID: <016b01c4c1f9$1508c410$8100000a@cytelcom.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hey guys, > This has worked before but for some reason isn't anymore and I have noclue> what to check. > Here are the steps I follow: > > 1. Place call to PSTN number. They answer and we talk. > 2. I press 'Conference' button on Cisco phone. > 3. Line 1 is now on hold and I get a new dial tone. > 4. Place call 2 to another PSTN. They answer and we talk. > 5. I press 'Join' on the Cisco phone. Caller 1 gets dropped and I get the > following > message in * console: > > Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from > 10.0.0.122 > > Now, 10.0.0.122 is the IP of my Cisco phone. * has 2 NICs, 1 is 10.0.3.10 > and the other is external public IP. I can make/recieve calls all daylong.> But recently this conference stopped working. > > Any ideas on what to check? The error doesn't make sense since the 2 calls > are present. Right before I press join, I can put caller 2 on hold and > resume caller 1 and vice versa. It isn't until I press 'Join' that call 1is> dropped. > > This works fine if caller 1 and 2 are both other phones in the office or > caller 1 is a phone in office and 2 is PSTN. Just doesn't work when both > PSTN. Worked before.. > > THanks, > Matthew > > > > ------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest, Vol 4, Issue 52 > ********************************************* >
Try two things (one at a time, to see which one helps you) in sip.conf: 1. After the IP address on the register statement, append "/954XXXXXX" 2. In the [Broadvoice] section, change NAT to no. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 4 Nov 2004, Francisco Perez-Landaeta wrote:> I have the following registration information for broadvoice, but it fails > to register. Can anyone tell me what is the problem ? I am not behind a > firewall and I have added this below the [general] > > Thanks, > > My number and password is hidded. I received from broadvoice the following : > > primary_dns_ip: 147.135.0.6 > secondary_dns_ip: 147.138.8.6 > proxy_ip: proxy.broadvoice.com > proxy_port: 5060 > registrar_ip: sip.broadvoice.com > registrar_port: 5060 > phone_number: 954XXXXXXX > auth_id: 954XXXXXXX > auth_password: password > tftp_ip: config.broadvoice.com > ntp_ip: ntp.nasa.gov > DTMF: InBand > registration_time_out: 10_seconds > voice_mail_key: *86 > > > BELOW IS WHAT I TOOK FROM THE WIKI, EVEN IF I CHANGE THE IP ADDRESSES IT > FAILS TO REGISTER. > > sip.conf > Under the general context: > ;externip=<IP or Dynamic DNS Name> ; Needed if behind NAT > context=incoming > dtmfmode=inband > register => 954XXXXXXX:password@147.135.0.129 > > [Broadvoice] > type=peer > username=954XXXXXXX > fromuser=954XXXXXXX > secret=password > host=147.135.0.129 > context=incoming > fromdomain=sip.broadvoice.com > nat=yes > canreinvite=no > dtmfmode=inband > > extensions.conf For incoming calls > [incoming] > ;This extension line will ring SIP > ;extension 2001 for 60 seconds then hang up. Modify as necessary to fit your > dialplan > exten => s,1,Dial(SIP/2001,60,tr) > exten => s,2,hangup > > extensions.conf For outgoing calls: > [toll-trunks] > ;Pattern match for local call plan, use appropriate pattern if on nationwide > plan. > exten => _9NXXXXXX,1,Dial(SIP/${EXTEN:1}@broadvoice) > exten => _9NXXXXXX,2,Congestion > > > ; > ----- Original Message ----- > From: <asterisk-users-request@lists.digium.com> > To: <asterisk-users@lists.digium.com> > Sent: Wednesday, November 03, 2004 7:01 PM > Subject: Asterisk-Users Digest, Vol 4, Issue 52 > > > > 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-owner@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: ISDN Dialplan (Paulo Adriano) > > 2. Re: Sip clients not longer registering (David Filion) > > 3. RE: Installing X100P Asterisk - Unable to createchannel of > > type 'Zap' (Vikas Deolaliker) > > 4. MusicOnhold on Bridged calls (Paulo Adriano) > > 5. Re: Automatically restart asterisk if not running > > (Brancaleoni Matteo) > > 6. What do I need to ask my T1 supplier? (Scott Nelson) > > 7. getting cid from spa3k pstn to * (Randy Bush) > > 8. How change default law for T100P (Manuel Marin) > > 9. Re: What do I need to ask my T1 supplier? (niles@atheos.net) > > 10. Re: What do I need to ask my T1 supplier? (TC) > > 11. Cisco 79XX - Using built-in 3way conference (Matthew Boehm) > > > > > > ---------------------------------------------------------------------- > > > > Message: 1 > > Date: Wed, 03 Nov 2004 22:06:10 +0000 > > From: "Paulo Adriano" <pauloadriano@wavelis.pt> > > Subject: Re: [Asterisk-Users] ISDN Dialplan > > To: <asterisk-users@lists.digium.com>, <psvasterisk@psv.nu> > > Cc: mje@posix.co.za > > Message-ID: <s1895669.024@wavelis> > > Content-Type: text/plain; charset="us-ascii" > > > > It*s solved now. You are right it was a matter of formating the > > outgoing number. > > > > Thanks > > > > Paulo > > > > >>>psvasterisk@psv.nu 11/03 1:34 pm >>> > > > > On Wed, 3 Nov 2004, Paulo Adriano wrote: > > > > > > >After trying that syntax I still have the same problem, one thing is > > > > >very strange is the number that Asterisk reports as the incoming. My > > > > >ESN*s numbers are 219898334 and 219898335 but on the console I see > > > > >219898334,1 and 219898335,1 > > > > > > Are you sure it is not trying to tell you extension 219898334 and > > priority > > > > 1 in the dialplan? > > > > > > [snip} > > > > > > >THIS IS AN OUTGOING CALL TRY TO NUMBER 213570150 MY DIALPLAN > > REQUIRES 9 to go outside > > > > > > > > > >*CLI> -- Executing Dial(SIP/21-9da8, Modem/g1/9213570150||tr) in > > new stack > > > > >Nov 3 18:37:35 WARNING[1110502320]: chan_modem.c:191 modem_call: > > Destination g1/9213570150 requres a real destination > > (device:destination) > > > > > -- Couldn't call g1/9213570150 > > > > > -- Hungup 'Modem[i4l]/ttyI1' > > > > > == Everyone is busy/congested at this time > > > > >Nov 3 18:37:45 WARNING[1110502320]: pbx.c:1933 ast_pbx_run: Timeout, > > but no rule 't' in context 'local-access' > > > > > -- Saved useragent SJLabs-SJphone/1.30.248 for peer 21 > > > > > > The source for chan_modem suggests that the dial string should be > > > > formatted like > > > > Dial(Modem/g1:9213570150||tr) > > > > > > Peter > > > > > > > > > > > > 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 > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20041103/d471c6ed/attachment-0001.html > > > > ------------------------------ > > > > Message: 2 > > Date: Wed, 03 Nov 2004 17:11:39 -0500 > > From: David Filion <dfilion@dotality.com> > > Subject: Re: [Asterisk-Users] Sip clients not longer registering > > To: asterisk-users@lists.digium.com > > Message-ID: <4189579B.4030804@dotality.com> > > Content-Type: text/plain; charset=us-ascii; format=flowed > > > > > > > > Message: 8 > > Date: Wed, 03 Nov 2004 22:32:46 +0100 > > From: "Olle E. Johansson" <oej@edvina.net> > > Subject: Re: [Asterisk-Users] Sip clients not longer registering > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > <asterisk-users@lists.digium.com> > > Message-ID: <41894E7E.1060006@edvina.net> > > Content-Type: text/plain; charset=us-ascii; format=flowed > > > > David Filion wrote: > > > > > > > > >> Hi, > > >> > > >> We have been using Asterisk since version 0.9x with little or no > > >> problems. However, for an unknow reasons, our sip clients can nolonger > > >> register. We updated to Asterisk 1.0.2 hoping that would solve the > > >> problem, but no luck. > > >> > > > > > > > > There is a chance that our change NAT logic is problematic in your > network. > > (rport support) > > > > check the sample sip.conf for various options to nat= and try them. > > It's in configs/sip.conf.sample > > > > /O > > > > > > > > Thanks, I'll give them a try. > > > > David > > > > > > > > > > ------------------------------ > > > > Message: 3 > > Date: Wed, 3 Nov 2004 14:15:49 -0800 > > From: "Vikas Deolaliker" <vikasd@yahoo.com> > > Subject: RE: [Asterisk-Users] Installing X100P Asterisk - Unable to > > createchannel of type 'Zap' > > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > <asterisk-users@lists.digium.com> > > Message-ID: <20041103221544.2A0F82FE4ED@lists.digium.com> > > Content-Type: text/plain; charset="us-ascii" > > > > > > Look at your logs in /var/logs/asterisk/. I am pretty certain it is a > fault > > in your /etc/asterisk/Zapata.conf file. > > > > Vikas > > > > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Seth > Remington > > Sent: Wednesday, November 03, 2004 1:01 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Installing X100P Asterisk - Unable to > > createchannel of type 'Zap' > > > > On Wed, 2004-11-03 at 13:02, Frank Kostin wrote: > > > Hello list, > > > I am trying to install a Digium X100P into a Redhat Asterisk. > > > Kernel seems to be OK, card OK. > > > Zaptel Configuration seems to be OK. > > > # ztcfg -vv > > > Channel map: > > > Channel 01: FXS Kewlstart (Default) (Slaves: 01) > > > 1 channels configured. > > > > > > Asterisk works fine with IP SIP but not with X100P > > > I get the error on Asterisk CLI> > > > .... channel.c:1919 ast_request: "No channel type registered for > > > 'Zap'" > > > and than ....app_dial.c:763 dial_exec "Unable to create channel of > > > type 'Zap'". > > > > > > Does anyone know what might be the problem ? > > > Thanks for any help > > > > Did you: > > > > modprobe zaptel > > modprobe wcfxo > > > > Also make sure that you compiled and installed zaptel *before* you > > installed asterisk. If you did it afterwords simply re-compile asterisk > > and you should be good. > > > > -Seth > > > > -- > > Seth Remington > > SaberLogic, LLC > > 661-B Weber Drive > > Wadsworth, Ohio 44281 > > Phone: (330)335-6442 > > Fax: (330)336-8559 > > > > _______________________________________________ > > 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 > > > > > > > > > > ------------------------------ > > > > Message: 4 > > Date: Wed, 03 Nov 2004 22:27:23 +0000 > > From: "Paulo Adriano" <pauloadriano@wavelis.pt> > > Subject: [Asterisk-Users] MusicOnhold on Bridged calls > > To: <asterisk-users@lists.digium.com> > > Message-ID: <s1895b59.027@wavelis> > > Content-Type: text/plain; charset="us-ascii" > > > > Now that my bridged calls are working fine with ISDN I have a question ? > > > > > > When my customers call in and my ext is not available the call is routed > > out to my mobile. > > > > Everything works but I would like to know if there is a way of having > > the calling sign (tone) always on . With my current config the origin > > caller phone hear a calling beeping tone only until the call is passed > > on a bridged call. Then and until the end user (in this case , myself in > > the cellular phone) answers the call , the origin has a big silence. > > This may cause him to think that something append during the call. > > > > How can I configure * in order to avoid the calling tone from > > desapearing on a bridged call. Even music would be acceptable as the > > call is being tranfered to a pstn number. > > > > Thanks in advance > > Paulo > > > > Francisco Paulo Adriano > > WaveLIS LDA > > Mobile +351 91 870 87 98 > > Office + 351 21 989 83 34 > > Fax +351 21 989 83 35 > > E-mail : pauloadriano@wavelis.pt > > > > > > > > > > > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20041103/34b4f9c5/attachment-0001.html > > > > ------------------------------ > > > > Message: 5 > > Date: Wed, 03 Nov 2004 23:28:04 +0100 > > From: Brancaleoni Matteo <mbrancaleoni@espia.it> > > Subject: Re: [Asterisk-Users] Automatically restart asterisk if not > > running > > To: Matthew Marlowe <matthew.marlowe@gmail.com>, Asterisk Users > > Mailing List - Non-Commercial Discussion > > <asterisk-users@lists.digium.com> > > Cc: asterisk-users@lists.digium.com > > Message-ID: <1099520884.11249.2.camel@athlon64> > > Content-Type: text/plain > > > > Hi > > Il mer, 2004-11-03 alle 21:41, Matthew Marlowe ha scritto: > > > I once found a script, I think it was on the mailing list that would > > > run in crontab and restart asterisk if not running... Does anyone > > > happen to have a copy of that? > > > > I suggest you to use something like a superdaemon, > > ie a process (normally spawned by init) that > > checks other processes. > > I suggest you monit (http://www.tildeslash.com/monit/), > > it can monitor processes in various ways and other > > vital system informations. I'm pretty happy > > with it (using it in all my * installations) > > > > Matteo > > -- > > Brancaleoni Matteo <mbrancaleoni@espia.it> > > Espia Srl > > > > > > > > ------------------------------ > > > > Message: 6 > > Date: Wed, 3 Nov 2004 17:32:27 -0500 > > From: Scott Nelson <asterisk@nelson.saint-louis.mo.us> > > Subject: [Asterisk-Users] What do I need to ask my T1 supplier? > > To: asterisk-users@lists.digium.com > > Message-ID: <200411031632.27499.asterisk@nelson.saint-louis.mo.us> > > Content-Type: text/plain; charset="us-ascii" > > > > My employer is switching to a new T1 supplier (it was AT&T, we are now > going > > with XO), and sometime in the future we want to replace our PBX with an > > Asterisk system. > > > > What do I need to know to make sure the T1 line is "provisioned" (is that > the > > right term?) correctly for a Digium T100P/TE410P/TE405P? > > > > They will split the T1 line into 10 channels of voice and 14 channels of > data. > > >From what I understand, they will terminate the T1 into a channel bank, > and > > then from that give is 10 POTS phone jacks and one data port (to go to an > > Adtran router for our Internet access). > > > > Any comments and/or suggestions? > > > > Scott > > > > > > ------------------------------ > > > > Message: 7 > > Date: Wed, 3 Nov 2004 14:40:29 -0800 > > From: Randy Bush <randy@psg.com> > > Subject: [Asterisk-Users] getting cid from spa3k pstn to * > > To: splatters <asterisk-users@lists.digium.com> > > Message-ID: <16777.24157.228132.850877@ran.psg.com> > > Content-Type: text/plain; charset=us-ascii > > > > i am still going crazy with this one. i can not get callerid from a call > > received on the spa3k pstn to asterisk. THIS USED TO WORK! > > > > in order to get the cid from the spa3k to *, i need to turn on > > PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES > > > > the sip.conf entry looks like > > [spa3k] > > type=friend > > host=dynamic > > port=5061 > > auth=md5 > > secret=hidden > > qualify=1000 > > dtmfmode=rfc2833 > > canreinvite=yes > > context=spa3k-ext > > > > this produces a sip exchange as follows: > > > > No. Time Source Destination > Protocol Info > > 1 0.000000 spa3k asterisk.foo.edu SIP/SDP Request: INVITE > sip:105@asterisk.foo.edu, with session description > > > > Frame 1 (1095 bytes on wire, 1095 bytes captured) > > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu > (asterisk-ip-ad) > > User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) > > Session Initiation Protocol > > Request-Line: INVITE sip:105@asterisk.foo.edu SIP/2.0 > > Message Header > > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-f9456447 > > From: CallerName > <sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1 > > SIP Display info: CallerName > > SIP from address: sip:2065551212@asterisk.foo.edu > > SIP tag: 54e649b356424567o1 > > To: <sip:105@asterisk.foo.edu> > > SIP to address: sip:105@asterisk.foo.edu > > Remote-Party-ID: CallerName > <sip:2065551212@asterisk.foo.edu>;screen=yes;party=calling > > Call-ID: 51efe8a3-2d73b337@spa3k > > CSeq: 101 INVITE > > Max-Forwards: 70 > > Contact: biwa 0431 <sip:biwaa1-in@spa3k:5061> > > Expires: 240 > > User-Agent: Sipura/SPA3000-2.0.11(GWa) > > Content-Length: 430 > > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > > Supported: x-sipura > > Content-Type: application/sdp > > Message body > > Session Description Protocol > > > > No. Time Source Destination > Protocol Info > > 2 0.000514 asterisk.foo.edu spa3k SIP Status: 407 Proxy > Authentication Required > > > > Frame 2 (520 bytes on wire, 520 bytes captured) > > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a > > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), Dst > Addr: spa3k (spa3k) > > User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5061 (5061) > > Session Initiation Protocol > > Status-Line: SIP/2.0 407 Proxy Authentication Required > > Message Header > > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-f9456447 > > From: CallerName > <sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1 > > SIP Display info: CallerName > > SIP from address: sip:2065551212@asterisk.foo.edu > > SIP tag: 54e649b356424567o1 > > To: <sip:105@asterisk.foo.edu>;tag=as741941ff > > SIP to address: sip:105@asterisk.foo.edu > > SIP tag: as741941ff > > Call-ID: 51efe8a3-2d73b337@spa3k > > CSeq: 101 INVITE > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Contact: <sip:105@asterisk-ip-ad> > > Proxy-Authenticate: Digest realm="asterisk", nonce="263c07e5" > > Content-Length: 0 > > > > No. Time Source Destination > Protocol Info > > 3 0.090441 spa3k asterisk.foo.edu SIP Request: ACK > sip:105@asterisk.foo.edu > > > > Frame 3 (453 bytes on wire, 453 bytes captured) > > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu > (asterisk-ip-ad) > > User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) > > Session Initiation Protocol > > Request-Line: ACK sip:105@asterisk.foo.edu SIP/2.0 > > Message Header > > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-f9456447 > > From: CallerName > <sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1 > > SIP Display info: CallerName > > SIP from address: sip:2065551212@asterisk.foo.edu > > SIP tag: 54e649b356424567o1 > > To: <sip:105@asterisk.foo.edu>;tag=as741941ff > > SIP to address: sip:105@asterisk.foo.edu > > SIP tag: as741941ff > > Call-ID: 51efe8a3-2d73b337@spa3k > > CSeq: 101 ACK > > Max-Forwards: 70 > > Contact: biwa 0431 <sip:biwaa1-in@spa3k:5061> > > User-Agent: Sipura/SPA3000-2.0.11(GWa) > > Content-Length: 0 > > > > No. Time Source Destination > Protocol Info > > 4 0.135913 spa3k asterisk.foo.edu SIP/SDP Request: INVITE > sip:105@asterisk.foo.edu, with session description > > > > Frame 4 (1265 bytes on wire, 1265 bytes captured) > > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu > (asterisk-ip-ad) > > User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) > > Session Initiation Protocol > > Request-Line: INVITE sip:105@asterisk.foo.edu SIP/2.0 > > Message Header > > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-e2744867 > > From: CallerName > <sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1 > > SIP Display info: CallerName > > SIP from address: sip:2065551212@asterisk.foo.edu > > SIP tag: 54e649b356424567o1 > > To: <sip:105@asterisk.foo.edu> > > SIP to address: sip:105@asterisk.foo.edu > > Remote-Party-ID: CallerName > <sip:2065551212@asterisk.foo.edu>;screen=yes;party=calling > > Call-ID: 51efe8a3-2d73b337@spa3k > > CSeq: 102 INVITE > > Max-Forwards: 70 > > Proxy-Authorization: Digest > username="biwaa1-in",realm="asterisk",nonce="263c07e5",uri="sip:105@asterisk > foo.edu",algorithm=MD5,response="f8e02292686b3b5cb2117186b1474ba9" > > Contact: biwa 0431 <sip:biwaa1-in@spa3k:5061> > > Expires: 240 > > User-Agent: Sipura/SPA3000-2.0.11(GWa) > > Content-Length: 430 > > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > > Supported: x-sipura > > Content-Type: application/sdp > > Message body > > Session Description Protocol > > > > No. Time Source Destination > Protocol Info > > 5 0.136261 asterisk.foo.edu spa3k SIP Status: 403 > Forbidden > > > > Frame 5 (437 bytes on wire, 437 bytes captured) > > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a > > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), Dst > Addr: spa3k (spa3k) > > User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5061 (5061) > > Session Initiation Protocol > > Status-Line: SIP/2.0 403 Forbidden > > Message Header > > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-e2744867 > > From: CallerName > <sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1 > > SIP Display info: CallerName > > SIP from address: sip:2065551212@asterisk.foo.edu > > SIP tag: 54e649b356424567o1 > > To: <sip:105@asterisk.foo.edu>;tag=as741941ff > > SIP to address: sip:105@asterisk.foo.edu > > SIP tag: as741941ff > > Call-ID: 51efe8a3-2d73b337@spa3k > > CSeq: 102 INVITE > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Contact: <sip:105@asterisk-ip-ad> > > Content-Length: 0 > > > > No. Time Source Destination > Protocol Info > > 6 0.383761 spa3k asterisk.foo.edu SIP Request: ACK > sip:105@asterisk.foo.edu > > > > Frame 6 (623 bytes on wire, 623 bytes captured) > > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu > (asterisk-ip-ad) > > User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) > > Session Initiation Protocol > > Request-Line: ACK sip:105@asterisk.foo.edu SIP/2.0 > > Message Header > > Via: SIP/2.0/UDP spa3k:5061;branch=z9hG4bK-e2744867 > > From: CallerName > <sip:2065551212@asterisk.foo.edu>;tag=54e649b356424567o1 > > SIP Display info: CallerName > > SIP from address: sip:2065551212@asterisk.foo.edu > > SIP tag: 54e649b356424567o1 > > To: <sip:105@asterisk.foo.edu>;tag=as741941ff > > SIP to address: sip:105@asterisk.foo.edu > > SIP tag: as741941ff > > Call-ID: 51efe8a3-2d73b337@spa3k > > CSeq: 102 ACK > > Max-Forwards: 70 > > Proxy-Authorization: Digest > username="biwaa1-in",realm="asterisk",nonce="263c07e5",uri="sip:105@asterisk > foo.edu",algorithm=MD5,response="c33e3a4bab8eef38ca12b9ddf192b796" > > Contact: biwa 0431 <sip:biwaa1-in@spa3k:5061> > > User-Agent: Sipura/SPA3000-2.0.11(GWa) > > Content-Length: 0 > > > > No. Time Source Destination > Protocol Info > > 7 7.079655 asterisk.foo.edu spa3k SIP Request: OPTIONS > sip:spa3k > > > > Frame 7 (463 bytes on wire, 463 bytes captured) > > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a > > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), Dst > Addr: spa3k (spa3k) > > User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) > > Session Initiation Protocol > > Request-Line: OPTIONS sip:spa3k SIP/2.0 > > Message Header > > Via: SIP/2.0/UDP asterisk-ip-ad:5060;branch=z9hG4bK1b040c15 > > From: "Unknown" <sip:Unknown@asterisk-ip-ad>;tag=as3f547347 > > SIP Display info: "Unknown" > > SIP from address: sip:Unknown@asterisk-ip-ad > > SIP tag: as3f547347 > > To: <sip:spa3k> > > SIP to address: sip:spa3k > > Contact: <sip:Unknown@asterisk-ip-ad> > > Call-ID: 159ec16b69ac62e334905b487158eeed@asterisk-ip-ad > > CSeq: 102 OPTIONS > > User-Agent: Asterisk PBX > > Date: Mon, 01 Nov 2004 17:34:49 GMT > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Content-Length: 0 > > > > No. Time Source Destination > Protocol Info > > 8 7.079766 asterisk.foo.edu spa3k SIP Request: OPTIONS > sip:spa3k:5061 > > > > Frame 8 (473 bytes on wire, 473 bytes captured) > > Ethernet II, Src: 00:30:48:80:b3:72, Dst: 00:20:fc:1e:ce:3a > > Internet Protocol, Src Addr: asterisk.foo.edu (asterisk-ip-ad), Dst > Addr: spa3k (spa3k) > > User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5061 (5061) > > Session Initiation Protocol > > Request-Line: OPTIONS sip:spa3k:5061 SIP/2.0 > > Message Header > > Via: SIP/2.0/UDP asterisk-ip-ad:5060;branch=z9hG4bK29909a71 > > From: "Unknown" <sip:Unknown@asterisk-ip-ad>;tag=as67500153 > > SIP Display info: "Unknown" > > SIP from address: sip:Unknown@asterisk-ip-ad > > SIP tag: as67500153 > > To: <sip:spa3k:5061> > > SIP to address: sip:spa3k:5061 > > Contact: <sip:Unknown@asterisk-ip-ad> > > Call-ID: 2b80a2980a32bf7809b8648328ced971@asterisk-ip-ad > > CSeq: 102 OPTIONS > > User-Agent: Asterisk PBX > > Date: Mon, 01 Nov 2004 17:34:49 GMT > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Content-Length: 0 > > > > No. Time Source Destination Protoc > ol Info > > 9 7.173099 spa3k asterisk.foo.edu SIP Status: 404 Not > Found > > > > Frame 9 (361 bytes on wire, 361 bytes captured) > > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > > Internet Protocol, Src Addr: spa3k (spa3k), Dst Addr: asterisk.foo.edu > (asterisk-ip-ad) > > User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) > > Session Initiation Protocol > > Status-Line: SIP/2.0 404 Not Found > > Message Header > > To: <sip:spa3k>;tag=828c8dcf8cd9e760i0 > > SIP to address: sip:spa3k > > SIP tag: 828c8dcf8cd9e760i0 > > From: "Unknown" <sip:Unknown@asterisk-ip-ad>;tag=as3f547347 > > SIP Display info: "Unknown" > > SIP from address: sip:Unknown@asterisk-ip-ad > > SIP tag: as3f547347 > > Call-ID: 159ec16b69ac62e334905b487158eeed@asterisk-ip-ad > > CSeq: 102 OPTIONS > > Via: SIP/2.0/UDP asterisk-ip-ad:5060;branch=z9hG4bK1b040c15 > > Server: Sipura/SPA3000-2.0.11(GWa) > > Content-Length: 0 > > > > note that the From: has the cid, as does the Remote-Party-ID:. and the > > Contact: has the spa3k's id and display name. and > > the From: and/or Remote-Party-ID: cause asterisk respond with 407 Proxy > > Authentication Required, and things do not improve from there > > > > if i set the spa3k config to have > > PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = NO > > > > Frame 1 (1072 bytes on wire, 1072 bytes captured) > > Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 > > Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: > 666.42.7.11 (666.42.7.11) > > User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) > > Session Initiation Protocol > > Request-Line: INVITE sip:105@my.asterisk.su SIP/2.0 > > Method: INVITE > > Resent Packet: False > > Message Header > > Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-f5998d8a > > From: spa3k pstn <sip:spa3k@my.asterisk.su>;tag=8fc58211a0dc60f2o1 > > To: <sip:105@my.asterisk.su> > > Remote-Party-ID: spa3k pstn > <sip:spa3k@my.asterisk.su>;screen=yes;party=calling > > Call-ID: daed83bd-b2b66b36@42.666.11.7 > > CSeq: 101 INVITE > > Max-Forwards: 70 > > Contact: spa3k pstn <sip:biwaa1@42.666.11.7:5061> > > Expires: 240 > > User-Agent: Sipura/SPA3000-2.0.11(GWa) > > Content-Length: 430 > > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > > Supported: x-sipura > > Content-Type: application/sdp > > Message body > > Session Description Protocol > > > > the connection completes, but asterisk does not have the pstn caller id. > > > > randy > > > > > > > > ------------------------------ > > > > Message: 8 > > Date: Wed, 3 Nov 2004 15:40:33 -0700 (MST) > > From: Manuel Marin <mmg@transtelco.com.mx> > > Subject: [Asterisk-Users] How change default law for T100P > > To: asterisk-users@lists.digium.com > > Message-ID: > > <653967.1099521633455.SLOX.WebMail.wwwrun@iGrup.transtelco.com.mx> > > Content-Type: text/plain; charset=us-ascii > > > > I would like to know if there is a way to change default ulaw for a T1 > > card. I see in the zap show channel X that is working as ulaw. How do I > > change it in zapata.conf or zaptel.conf to alaw. Iam interconnecting a > > Meridian PBX but I need to configure it as alaw. > > > > > > > > > > ------------------------------ > > > > Message: 9 > > Date: Wed, 3 Nov 2004 17:40:55 -0500 > > From: niles@atheos.net > > Subject: Re: [Asterisk-Users] What do I need to ask my T1 supplier? > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > <asterisk-users@lists.digium.com> > > Message-ID: <6C40DB70-2DE9-11D9-B4EE-000A957899C8@atheos.net> > > Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed > > > > > > On Nov 3, 2004, at 5:32 PM, Scott Nelson wrote: > > > > > My employer is switching to a new T1 supplier (it was AT&T, we are now > > > going > > > with XO), and sometime in the future we want to replace our PBX with an > > > Asterisk system. > > > > > > What do I need to know to make sure the T1 line is "provisioned" (is > > > that the > > > right term?) correctly for a Digium T100P/TE410P/TE405P? > > > > > > They will split the T1 line into 10 channels of voice and 14 channels > > > of data. > > > From what I understand, they will terminate the T1 into a channel > > > bank, and > > > then from that give is 10 POTS phone jacks and one data port (to go to > > > an > > > Adtran router for our Internet access). > > > > > > Any comments and/or suggestions? > > > > > > Scott > > > _______________________________________________ > > > > Scott, > > > > you can skip the channel bank & router, and use asterisk with a T100P to > > serve your data & voice. You can find all the info you need on the Wiki > > http://www.voip-info.org/tiki-index.php? > > page=Asterisk%20Data%20Configuration > > > > I use this setup for 11 voice channels and 256K of data from Nuvox. > > Niles > > > > > > > > > > ------------------------------ > > > > Message: 10 > > Date: Wed, 03 Nov 2004 14:47:09 -0800 > > From: TC <trclark@shaw.ca> > > Subject: Re: [Asterisk-Users] What do I need to ask my T1 supplier? > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > <asterisk-users@lists.digium.com> > > Message-ID: <032601c4c1f7$0dd0ff20$c901a8c0@w2ktopcat> > > Content-Type: text/plain; charset=iso-8859-1 > > > > > > > > >They will split the T1 line into 10 channels of voice and 14 channels of > > data. > > >From what I understand, they will terminate the T1 into a channel bank, > and > > >then from that give is 10 POTS phone jacks and one data port (to go to an > > >Adtran router for our Internet access). > > > > >Any comments and/or suggestions? > > > > what would be realy nice from them is to present those 10 voice channels > > as not POTS but as the first 10 channels of a pri t1 net interface ie a > > fractional t1 voice > > and skip the a/d nonsense I know an adit 600 with a router & t1 cards can > do > > that for you > > > > > > > > > > ------------------------------ > > > > Message: 11 > > Date: Wed, 3 Nov 2004 17:01:11 -0600 > > From: "Matthew Boehm" <mboehm@cytelcom.com> > > Subject: [Asterisk-Users] Cisco 79XX - Using built-in 3way conference > > To: <asterisk-users@lists.digium.com> > > Message-ID: <016b01c4c1f9$1508c410$8100000a@cytelcom.com> > > Content-Type: text/plain; charset="iso-8859-1" > > > > Hey guys, > > This has worked before but for some reason isn't anymore and I have no > clue > > what to check. > > Here are the steps I follow: > > > > 1. Place call to PSTN number. They answer and we talk. > > 2. I press 'Conference' button on Cisco phone. > > 3. Line 1 is now on hold and I get a new dial tone. > > 4. Place call 2 to another PSTN. They answer and we talk. > > 5. I press 'Join' on the Cisco phone. Caller 1 gets dropped and I get the > > following > > message in * console: > > > > Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from > > 10.0.0.122 > > > > Now, 10.0.0.122 is the IP of my Cisco phone. * has 2 NICs, 1 is 10.0.3.10 > > and the other is external public IP. I can make/recieve calls all day > long. > > But recently this conference stopped working. > > > > Any ideas on what to check? The error doesn't make sense since the 2 calls > > are present. Right before I press join, I can put caller 2 on hold and > > resume caller 1 and vice versa. It isn't until I press 'Join' that call 1 > is > > dropped. > > > > This works fine if caller 1 and 2 are both other phones in the office or > > caller 1 is a phone in office and 2 is PSTN. Just doesn't work when both > > PSTN. Worked before.. > > > > THanks, > > Matthew > > > > > > > > ------------------------------ > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > End of Asterisk-Users Digest, Vol 4, Issue 52 > > ********************************************* > > > _______________________________________________ > 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 > > This message has been categorized as "Legitimate" by Bayesian Analyzer. > If you do not agree, please click on the link below to train the Analyzer. > http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-04%5C0020ec53bb61418696f6cac85fe8e8c3&C=2 > > -- > ----------------------------------------------------------------------- > This message has been inspected by DynaComm i:mail > ----------------------------------------------------------------------- > >