search for: hangupcausecode

...ding ISDN cause codes to customer pbx (test scenario for unallocated number) topology: PSTN-E1-AsteriskA-AsteriskB-SOMEPBX INVITE from SOMEPBX to PSTN AsteriskA sends to AsteriskB Status-Line: SIP/2.0 503 Service Unavailable X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 how can i resend HangupCauseCode from AsteriskB to SOMEPBX? i'm tried this on AsteriskB exten => _X.,1,Dial(SIP/AsteriskA,${EXTEN}) exten => _X.,n,Hangup(${SIP_HEADER(X-Asterisk-HangupCauseCode)}) thanks -- --------------------------------------- Marek Cervenka ===================...

...ip:tzl at voip.touk.pl>;tag=as7217acbc Call-ID: 307fda656066a7e264e85cea0742e601 at 192.168.129.74 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 CHeers tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071221/ec5c2dde/attachment.htm

Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( Thanks for help. Chris

Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140723/09e97fd1/attachment.html>

I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get

...-01 at 192.168.0.25>;tag=as112dbb55 To: <sip:dect at 192.168.0.27:5060>;tag=1813732733 Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060 CSeq: 103 BYE User-Agent: Asterisk PBX 15.2.2 ===> X-Asterisk-HangupCause: Bearer capability not available <=== ===> X-Asterisk-HangupCauseCode: 58 <=== Content-Length: 0 Anyone around to give some pointers/clues? -------------- next part -------------- [Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10434 process_sdp: Received AVP profile in audio answer but AVPF is enabled: audio 7200 RTP/AVP 8...

Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the

Hi list, I write on the list looking for help, buy a openvox gw gsm for four channels and I'm a little disappointed with the support openvox, for some reason , The call doesn?t get trough support tells me it was my asterisk server, but does not really work me and my internal calls are working perfectly, I tested with another sangoma FXO gateway and works perfectly. the problem is that

...ISK:5060;branch=z9hG4bK156fd67d Max-Forwards: 70 From:<sip:LOCAL at ASTERISK>;tag=as4502927f To:<sip:REMOTE at SUPPLIER>;tag=gK02498cb1 Call-ID: 205665777_90679951 at SUPPLIER CSeq: 102 BYE User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:SUPPLIER:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ASTERISK:5060;branch=z9hG4bK156fd67d From:<sip:LOCAL at ASTERISK>;tag=as4502927f To:<sip:REMOTE at SUPPLIER>;tag=gK02498cb1 Call-ID: 205665777_90679951 at SUPPLIER CSeq: 102 BYE...

...8005555555 at outbound.vitelity.net>;tag=as5458ca04 To: "John Doe" <sip:1234 at 192.168.11.166>;tag=as466267de Call-ID: 59e9eff8339e32af271c23541298135d at 192.168.11.166:5060 CSeq: 102 BYE User-Agent: packetrino Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 64.2.142.189:5060 (no NAT) Scheduling destruction of SIP dialog '59e9eff8339e32af271c23541298135d at 192.168.11.166:5060' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 64.2.142.189:5060 ---&gt...

...d6 Call-ID: 667168938-1555-4 at BJC.BGI.B.F CSeq: 31 INVITE Server: Asterisk PBX 11.8.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 Content-Length: 0 <------------> <--- SIP read from UDP:192.168.1.5:1555 ---> ACK sip:22222222 at 192.168.1.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK415263616;rport From: <sip:telefono at 192.168.1.22>;tag=1524540678 To: <sip:22222222 at 192.168.1.22>...

...ystem 34" <sip:0@161.49.142.250>;tag=as4e5a553d To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- ?Scheduling destruction of SIP dialog '1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250' in 6400 ms (Method: INVITE) ? > Channel SIP/QuadNortel-09a4c0e0 was answered. ? [Khfemsrv*CLI> -- Executing [smvoice_callprogress@smvoice-dialout:1] GotoIf(&q...

...orwards: 70 > From: <sip:LOCAL at ASTERISK>;tag=as4502927f > To: <sip:REMOTE at SUPPLIER>;tag=gK02498cb1 > Call-ID: 205665777_90679951 at SUPPLIER > CSeq: 102 BYE > User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4 > X-Asterisk-HangupCause: Normal Clearing > X-Asterisk-HangupCauseCode: 16 > Content-Length: 0 > > > --- > > <--- SIP read from UDP:SUPPLIER:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP ASTERISK:5060;branch=z9hG4bK156fd67d > From: <sip:LOCAL at ASTERISK>;tag=as4502927f > To: <sip:REMOTE at SUPPLIER>;tag=gK02498cb1 >...

...p:3013 at 189.6.70.47>;tag=as41b028c6 Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [Nov 13 10:21:18] VERBOSE[25087] logger.c: <--- SIP read from 189.160.126.210:5060 ---> ACK sip:3013 at 189.6.70.47 SIP/2.0 Via: SIP/2.0/UDP 189.160.126.210:5060;branch=z9hG4bKb9f6b05aa5fb314af51ece37c Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47 From: &...

...ll-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M. CSeq: 2 INVITE Server: Asterisk PBX 1.8.11.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Circuit/channel congestion X-Asterisk-HangupCauseCode: 34 Content-Length: 0 <------------> Really destroying SIP dialog '71e46a1e52ecd53c591f47f12589a04c at sip.broadvoice.com' Method: INVITE == Spawn extension (gerencia, 90018006273999, 2) exited non-zero on 'SIP/488-00000000' <--- SIP read f...

...st.com>;tag=as1fa9b502 To: <sip:173242 at 12.194.12.12 >;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400 Call-ID: 04ce1d566f1f17a221caba261e2af4bb at test.localhost.com CSeq: 103 BYE User-Agent: FPBX-2.9.0(1.8.5.0) X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:12.194.12.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.131.12.13:5060 ;received=12.131.12.13;branch=z9hG4bK2ab85b31;rport=5060 From: "77057" <sip:77057 at test.localhost.com>;tag=as1fa9b502 To: <sip:173242 at 12.194.12.1...

...> Call-ID: 7f55e32e-e4c6e11a at 172.16.8.179 > CSeq: 102 INVITE > Server: inmaconsa-Voice-Sip-ipbx > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH, MESSAGE > Supported: replaces, timer > X-Asterisk-HangupCause: Call Rejected > X-Asterisk-HangupCauseCode: 21 > Content-Length: 0 > > > <------------> > == Spawn extension (to_pstn, 984783842, 2) exited non-zero on > 'SIP/101-0000004e' > > <--- SIP read from UDP:190.X.X.1:41316 ---> > ACK sip:984783842 at 50.X.X.X SIP/2.0 > Via: SIP/2.0/UDP 190.X.X....

Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip