-- Luca Pradovera [email protected]
Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C’s phone using the desk phone that sends a REFER request. When leg B is hung up as part of the REFER going through, Adhearsion receives a Hangup and terminates the call. There is not much else going on there, our original idea was to put a B2BUA on the APP server and to have that “swallow” refers so Adhearsion and the APP Asterisk never see it. Thanks! Luca > On Apr 28, 2015, at 19:00, [email protected] wrote: > > Send asterisk-users mailing list submissions to > [email protected] > > 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 > [email protected] > > You can reach the person managing the list at > [email protected] > > 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 proxying a REFER (Matthew Jordan) > 2. hi list need your help (????? ??????) > 3. Re: adding area code (Motty Cruz) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 28 Apr 2015 07:27:29 -0500 > From: Matthew Jordan <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] Asterisk proxying a REFER > Message-ID: > <CAN2PU+6UYLYFDXnt-XZztBz++8gGmKfdYwHRr84F93OzosV=w...@mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > On Mon, Apr 27, 2015 at 10:36 AM, Luca Pradovera > <[email protected]> wrote: >> Hello, >> we are using Asterisk with Adhearsion as our application server, with >> another Asterisk box acting as the office PBX, where all office phones are >> registered. >> >> A REFER to transfer calls within the office results in the Adhearsion >> application call being dropped, because the leg between the PBX and the app >> server is terminated by the PBX following the REFER. >> Is there a way to configure Asterisk 11 to proxy a refer across a bridge >> instead of following it, so the application server can follow it instead? >> > > Hey Luca - > > Unfortunately, there is not a simple or easy configuration setting > that tells Asterisk to proxy the REFER request through. Generally, > Asterisk doesn't like proxying anything. > > There may still be another way to handle this issue, depending on the > setup. Can you provide a bit more information about the channels on > the PBX/Adhearsion server, who sends the REFER request, and what > happens explicitly in the scenario? > > Matt > > -- > Matthew Jordan > Digium, Inc. | Director of Technology > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > > > ------------------------------ > > Message: 2 > Date: Tue, 28 Apr 2015 17:19:46 +0300 > From: ????? ?????? <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: [asterisk-users] hi list need your help > Message-ID: > <CAFgS45v=t-qkfktypjhj5yijwoh+d5pqy2jxf3w8yn9ir+b...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > facing problem with originating webrtc calls > > > 1-when iam doing call from webrtc iget ice working > <--- SIP read from WS:91.196.158.205:1466 ---> > INVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315 > Max-Forwards: 69 > To: <sip:[email protected]> > From: "Anton" <sip:[email protected]>;tag=5i21qaop43 > Call-ID: ocq4hu8eol3kijsgvt6b > CSeq: 1465 INVITE > Authorization: Digest algorithm=MD5, username="1065", realm="77.91.132.9", > nonce="5152b137", uri="sip:[email protected]", > response="446883f3c97a49ea7a9a554a1ba31b6a" > X-Can-Renegotiate: true > Contact: <sip:[email protected];transport=ws;ob> > Content-Type: application/sdp > Session-Expires: 90 > Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS > Supported: timer,ice,outbound > User-Agent: JsSIP 0.6.26 > Content-Length: 2554 > > v=0 > o=- 4785391175048354014 2 IN IP4 127.0.0.1 > s=- > t=0 0 > a=group:BUNDLE audio video > a=msid-semantic: WMS cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br > m=audio 2313 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 > c=IN IP4 192.168.88.26 > a=rtcp:2313 IN IP4 192.168.88.26 > a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host > generation 0 > a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host > generation 0 > a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype > active generation 0 > a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype > active generation 0 > a=ice-ufrag:8nMZ7w8bHdBBoY1a > a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR > a=fingerprint:sha-256 > 6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04 > a=setup:actpass > a=mid:audio > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time > a=sendrecv > a=rtcp-mux > a=rtpmap:111 opus/48000/2 > a=fmtp:111 minptime=10; useinbandfec=1 > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=maxptime:60 > a=ssrc:3696151487 cname:jXfPZ33h32Mx9liw > a=ssrc:3696151487 msid:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br > 8a2acec3-8511-4d36-9b51-05b8752c2ddd > a=ssrc:3696151487 mslabel:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br > a=ssrc:3696151487 label:8a2acec3-8511-4d36-9b51-05b8752c2ddd > m=video 2313 RTP/SAVPF 100 116 117 96 > c=IN IP4 192.168.88.26 > a=rtcp:2313 IN IP4 192.168.88.26 > a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host > generation 0 > a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host > generation 0 > a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype > active generation 0 > a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype > active generation 0 > a=ice-ufrag:8nMZ7w8bHdBBoY1a > a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR > a=fingerprint:sha-256 > 6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04 > a=setup:actpass > a=mid:video > a=extmap:2 urn:ietf:params:rtp-hdrext:toffset > a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time > a=recvonly > a=rtcp-mux > a=rtpmap:100 VP8/90000 > a=rtcp-fb:100 ccm fir > a=rtcp-fb:100 nack > a=rtcp-fb:100 nack pli > a=rtcp-fb:100 goog-remb > a=rtpmap:116 red/90000 > a=rtpmap:117 ulpfec/90000 > a=rtpmap:96 rtx/90000 > a=fmtp:96 apt=100 > > 2-BUT when i do channel originate sip/GOROD/XXXXX extension 1065@office > -- Executing [1065@office:1] Dial("SIP/GOROD-00000004", "SIP/1065") in > new stack > == Using SIP RTP CoS mark 5 > [Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:269 > ast_sockaddr_resolve: getaddrinfo("7cvtd9ihs2e8.invalid", "(null)", ...): > Name or service not known > [Apr 28 14:07:47] WARNING[4006][C-00000032]: chan_sip.c:15869 > __set_address_from_contact: Invalid host name in Contact: (can't resolve in > DNS) : '7cvtd9ihs2e8.invalid' > [Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:98 > ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported > Audio is at 16476 > Adding codec 100003 (ulaw) to SDP > Adding codec 100002 (gsm) to SDP > Adding codec 100004 (alaw) to SDP > Adding codec 100017 (testlaw) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (NAT) to 91.196.158.205:1466: > INVITE sip:[email protected];transport=ws SIP/2.0 > Via: SIP/2.0/WS 77.91.132.9:5060;branch=z9hG4bK3f293d79;rport > Max-Forwards: 70 > From: "asterisk" <sip:[email protected]>;tag=as78119d2b > To: <sip:[email protected];transport=ws> > Contact: <sip:[email protected]:5060;transport=WS> > Call-ID: [email protected]:5060 > CSeq: 102 INVITE > User-Agent: Asterisk PBX 11.15.0 > Date: Tue, 28 Apr 2015 11:07:47 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 437 > > v=0 > o=root 1122885298 1122885298 IN IP4 77.91.132.9 > s=Asterisk PBX 11.15.0 > c=IN IP4 77.91.132.9 > t=0 0 > m=audio 16476 RTP/SAVPF 0 3 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=connection:new > a=setup:actpass > a=fingerprint:SHA-256 > CC:82:C8:04:1F:DC:FE:B7:56:27:26:FF:18:CD:BB:71:99:B8:97:F9:81:2B:08:74:72:67:3B:A9:88:5F:00:34 > a=sendrecv > > thats why i got Failed to set remote offer sdp: Called with SDP without > ice-ufrag and ice-pwd > > Waiting for your advice ---thanks > > > > > -- > Best regards > Antony > ??? (066) 919-75-33 > ??? (063) 656-43-40 > [email protected] <mail%[email protected]> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20150428/4f152d16/attachment-0001.html> > > ------------------------------ > > Message: 3 > Date: Tue, 28 Apr 2015 07:21:12 -0700 > From: Motty Cruz <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] adding area code > Message-ID: <[email protected]> > Content-Type: text/plain; charset="windows-1252"; Format="flowed" > > this code worked for me, > > here is what I did and worked for me: > > exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444) > > exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) > > > Thanks for you help! > > On 04/27/2015 02:56 PM, Matt Riddell wrote: >> >>> On 27Apr, 2015, at 16:39, Motty Cruz <[email protected] >>> <mailto:[email protected]>> wrote: >>> >>> forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. >>> >>> Thanks, >>> >>> >>> On 04/27/2015 02:38 PM, Motty Cruz wrote: >>>> here is what I have: >>>> >>>> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >>>> >>>> exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) >>>> >>>> exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) >>>> >>>> not having success; >>>> >>>> "Got SIP reponse 503" Service Unavailable? >> >> Can you send us the console output when you make the call? >> >> -- >> Cheers, >> >> Matt Riddell >> _______________________________________________ >> >> http://www.venturevoip.com/news.php (Daily Asterisk News) >> http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) >> http://www.venturevoip.com/exchange.php (Full ITSP Solution) >> http://www.venturevoip.com/cc.php (Call Centre Solutions) >> >> >> > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20150428/7cc94c2f/attachment-0001.html> > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of asterisk-users Digest, Vol 129, Issue 32 > ***********************************************
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