<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><br class=""><div apple-content-edited="true" class="">
<div style="color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; " class=""><div class="">--</div><div class="">Luca Pradovera</div><div class=""><a href="mailto:luca.pradovera@gmail.com" class="">luca.pradovera@gmail.com</a></div><div class=""><br class=""></div></div><br class="Apple-interchange-newline">Hello,
</div><div apple-content-edited="true" class="">sorry, I managed to lose the reply amidst the traffic.</div><div apple-content-edited="true" class=""><br class=""></div><div apple-content-edited="true" class="">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.</div><div apple-content-edited="true" class=""><br class=""></div><div apple-content-edited="true" class="">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.</div><div apple-content-edited="true" class="">When leg B is hung up as part of the REFER going through, Adhearsion receives a Hangup and terminates the call.</div><div apple-content-edited="true" class=""><br class=""></div><div apple-content-edited="true" class="">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.</div><div apple-content-edited="true" class=""><br class=""></div><div apple-content-edited="true" class="">Thanks!</div><div apple-content-edited="true" class=""><br class=""></div><div apple-content-edited="true" class="">Luca</div><div apple-content-edited="true" class=""><br class=""></div>
<br class=""><div><blockquote type="cite" class=""><div class="">On Apr 28, 2015, at 19:00, <a href="mailto:asterisk-users-request@lists.digium.com" class="">asterisk-users-request@lists.digium.com</a> wrote:</div><br class="Apple-interchange-newline"><div class="">Send asterisk-users mailing list submissions to<br class=""><span class="Apple-tab-span" style="white-space:pre"> </span><a href="mailto:asterisk-users@lists.digium.com" class="">asterisk-users@lists.digium.com</a><br class=""><br class="">To subscribe or unsubscribe via the World Wide Web, visit<br class=""><span class="Apple-tab-span" style="white-space:pre"> </span>http://lists.digium.com/mailman/listinfo/asterisk-users<br class="">or, via email, send a message with subject or body 'help' to<br class=""><span class="Apple-tab-span" style="white-space:pre"> </span>asterisk-users-request@lists.digium.com<br class=""><br class="">You can reach the person managing the list at<br class=""><span class="Apple-tab-span" style="white-space:pre"> </span>asterisk-users-owner@lists.digium.com<br class=""><br class="">When replying, please edit your Subject line so it is more specific<br class="">than "Re: Contents of asterisk-users digest..."<br class=""><br class=""><br class="">Today's Topics:<br class=""><br class=""> 1. Re: Asterisk proxying a REFER (Matthew Jordan)<br class=""> 2. hi list need your help (????? ??????)<br class=""> 3. Re: adding area code (Motty Cruz)<br class=""><br class=""><br class="">----------------------------------------------------------------------<br class=""><br class="">Message: 1<br class="">Date: Tue, 28 Apr 2015 07:27:29 -0500<br class="">From: Matthew Jordan <mjordan@digium.com><br class="">To: Asterisk Users Mailing List - Non-Commercial Discussion<br class=""><span class="Apple-tab-span" style="white-space:pre"> </span><asterisk-users@lists.digium.com><br class="">Subject: Re: [asterisk-users] Asterisk proxying a REFER<br class="">Message-ID:<br class=""><span class="Apple-tab-span" style="white-space:pre"> </span><CAN2PU+6UYLYFDXnt-XZztBz++8gGmKfdYwHRr84F93OzosV=WQ@mail.gmail.com><br class="">Content-Type: text/plain; charset=UTF-8<br class=""><br class="">On Mon, Apr 27, 2015 at 10:36 AM, Luca Pradovera<br class=""><luca.pradovera@gmail.com> wrote:<br class=""><blockquote type="cite" class="">Hello,<br class="">we are using Asterisk with Adhearsion as our application server, with<br class="">another Asterisk box acting as the office PBX, where all office phones are<br class="">registered.<br class=""><br class="">A REFER to transfer calls within the office results in the Adhearsion<br class="">application call being dropped, because the leg between the PBX and the app<br class="">server is terminated by the PBX following the REFER.<br class=""> Is there a way to configure Asterisk 11 to proxy a refer across a bridge<br class="">instead of following it, so the application server can follow it instead?<br class=""><br class=""></blockquote><br class="">Hey Luca -<br class=""><br class="">Unfortunately, there is not a simple or easy configuration setting<br class="">that tells Asterisk to proxy the REFER request through. Generally,<br class="">Asterisk doesn't like proxying anything.<br class=""><br class="">There may still be another way to handle this issue, depending on the<br class="">setup. Can you provide a bit more information about the channels on<br class="">the PBX/Adhearsion server, who sends the REFER request, and what<br class="">happens explicitly in the scenario?<br class=""><br class="">Matt<br class=""><br class="">-- <br class="">Matthew Jordan<br class="">Digium, Inc. | Director of Technology<br class="">445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br class="">Check us out at: http://digium.com & http://asterisk.org<br class=""><br class=""><br class=""><br class="">------------------------------<br class=""><br class="">Message: 2<br class="">Date: Tue, 28 Apr 2015 17:19:46 +0300<br class="">From: ????? ?????? <satskiy.a@gmail.com><br class="">To: Asterisk Users Mailing List - Non-Commercial Discussion<br class=""><span class="Apple-tab-span" style="white-space:pre"> </span><asterisk-users@lists.digium.com><br class="">Subject: [asterisk-users] hi list need your help<br class="">Message-ID:<br class=""><span class="Apple-tab-span" style="white-space:pre"> </span><CAFgS45v=t-qkfkTypJhj5YijWOh+D5pQY2JXF3w8YN9iR+B5mg@mail.gmail.com><br class="">Content-Type: text/plain; charset="utf-8"<br class=""><br class="">facing problem with originating webrtc calls<br class=""><br class=""><br class="">1-when iam doing call from webrtc iget ice working<br class=""><--- SIP read from WS:91.196.158.205:1466 ---><br class="">INVITE sip:0669197533@77.91.132.9 SIP/2.0<br class="">Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315<br class="">Max-Forwards: 69<br class="">To: <sip:0669197533@77.91.132.9><br class="">From: "Anton" <sip:1065@77.91.132.9>;tag=5i21qaop43<br class="">Call-ID: ocq4hu8eol3kijsgvt6b<br class="">CSeq: 1465 INVITE<br class="">Authorization: Digest algorithm=MD5, username="1065", realm="77.91.132.9",<br class="">nonce="5152b137", uri="sip:0669197533@77.91.132.9",<br class="">response="446883f3c97a49ea7a9a554a1ba31b6a"<br class="">X-Can-Renegotiate: true<br class="">Contact: <sip:0momhddj@7cvtd9ihs2e8.invalid;transport=ws;ob><br class="">Content-Type: application/sdp<br class="">Session-Expires: 90<br class="">Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS<br class="">Supported: timer,ice,outbound<br class="">User-Agent: JsSIP 0.6.26<br class="">Content-Length: 2554<br class=""><br class="">v=0<br class="">o=- 4785391175048354014 2 IN IP4 127.0.0.1<br class="">s=-<br class="">t=0 0<br class="">a=group:BUNDLE audio video<br class="">a=msid-semantic: WMS cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br<br class="">m=audio 2313 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126<br class="">c=IN IP4 192.168.88.26<br class="">a=rtcp:2313 IN IP4 192.168.88.26<br class="">a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host<br class="">generation 0<br class="">a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host<br class="">generation 0<br class="">a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype<br class="">active generation 0<br class="">a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype<br class="">active generation 0<br class="">a=ice-ufrag:8nMZ7w8bHdBBoY1a<br class="">a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR<br class="">a=fingerprint:sha-256<br class="">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<br class="">a=setup:actpass<br class="">a=mid:audio<br class="">a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level<br class="">a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time<br class="">a=sendrecv<br class="">a=rtcp-mux<br class="">a=rtpmap:111 opus/48000/2<br class="">a=fmtp:111 minptime=10; useinbandfec=1<br class="">a=rtpmap:103 ISAC/16000<br class="">a=rtpmap:104 ISAC/32000<br class="">a=rtpmap:9 G722/8000<br class="">a=rtpmap:0 PCMU/8000<br class="">a=rtpmap:8 PCMA/8000<br class="">a=rtpmap:106 CN/32000<br class="">a=rtpmap:105 CN/16000<br class="">a=rtpmap:13 CN/8000<br class="">a=rtpmap:126 telephone-event/8000<br class="">a=maxptime:60<br class="">a=ssrc:3696151487 cname:jXfPZ33h32Mx9liw<br class="">a=ssrc:3696151487 msid:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br<br class="">8a2acec3-8511-4d36-9b51-05b8752c2ddd<br class="">a=ssrc:3696151487 mslabel:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br<br class="">a=ssrc:3696151487 label:8a2acec3-8511-4d36-9b51-05b8752c2ddd<br class="">m=video 2313 RTP/SAVPF 100 116 117 96<br class="">c=IN IP4 192.168.88.26<br class="">a=rtcp:2313 IN IP4 192.168.88.26<br class="">a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host<br class="">generation 0<br class="">a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host<br class="">generation 0<br class="">a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype<br class="">active generation 0<br class="">a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype<br class="">active generation 0<br class="">a=ice-ufrag:8nMZ7w8bHdBBoY1a<br class="">a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR<br class="">a=fingerprint:sha-256<br class="">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<br class="">a=setup:actpass<br class="">a=mid:video<br class="">a=extmap:2 urn:ietf:params:rtp-hdrext:toffset<br class="">a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time<br class="">a=recvonly<br class="">a=rtcp-mux<br class="">a=rtpmap:100 VP8/90000<br class="">a=rtcp-fb:100 ccm fir<br class="">a=rtcp-fb:100 nack<br class="">a=rtcp-fb:100 nack pli<br class="">a=rtcp-fb:100 goog-remb<br class="">a=rtpmap:116 red/90000<br class="">a=rtpmap:117 ulpfec/90000<br class="">a=rtpmap:96 rtx/90000<br class="">a=fmtp:96 apt=100<br class=""><br class="">2-BUT when i do channel originate sip/GOROD/XXXXX extension 1065@office<br class=""> -- Executing [1065@office:1] Dial("SIP/GOROD-00000004", "SIP/1065") in<br class="">new stack<br class=""> == Using SIP RTP CoS mark 5<br class="">[Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:269<br class="">ast_sockaddr_resolve: getaddrinfo("7cvtd9ihs2e8.invalid", "(null)", ...):<br class="">Name or service not known<br class="">[Apr 28 14:07:47] WARNING[4006][C-00000032]: chan_sip.c:15869<br class="">__set_address_from_contact: Invalid host name in Contact: (can't resolve in<br class="">DNS) : '7cvtd9ihs2e8.invalid'<br class="">[Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:98<br class="">ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported<br class="">Audio is at 16476<br class="">Adding codec 100003 (ulaw) to SDP<br class="">Adding codec 100002 (gsm) to SDP<br class="">Adding codec 100004 (alaw) to SDP<br class="">Adding codec 100017 (testlaw) to SDP<br class="">Adding non-codec 0x1 (telephone-event) to SDP<br class="">Reliably Transmitting (NAT) to 91.196.158.205:1466:<br class="">INVITE sip:0momhddj@7cvtd9ihs2e8.invalid;transport=ws SIP/2.0<br class="">Via: SIP/2.0/WS 77.91.132.9:5060;branch=z9hG4bK3f293d79;rport<br class="">Max-Forwards: 70<br class="">From: "asterisk" <sip:asterisk@77.91.132.9>;tag=as78119d2b<br class="">To: <sip:0momhddj@7cvtd9ihs2e8.invalid;transport=ws><br class="">Contact: <sip:asterisk@77.91.132.9:5060;transport=WS><br class="">Call-ID: 17a96e0848cdd7d226d3665a36c65c77@77.91.132.9:5060<br class="">CSeq: 102 INVITE<br class="">User-Agent: Asterisk PBX 11.15.0<br class="">Date: Tue, 28 Apr 2015 11:07:47 GMT<br class="">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,<br class="">PUBLISH, MESSAGE<br class="">Supported: replaces, timer<br class="">Content-Type: application/sdp<br class="">Content-Length: 437<br class=""><br class="">v=0<br class="">o=root 1122885298 1122885298 IN IP4 77.91.132.9<br class="">s=Asterisk PBX 11.15.0<br class="">c=IN IP4 77.91.132.9<br class="">t=0 0<br class="">m=audio 16476 RTP/SAVPF 0 3 8 101<br class="">a=rtpmap:0 PCMU/8000<br class="">a=rtpmap:3 GSM/8000<br class="">a=rtpmap:8 PCMA/8000<br class="">a=rtpmap:101 telephone-event/8000<br class="">a=fmtp:101 0-16<br class="">a=ptime:20<br class="">a=connection:new<br class="">a=setup:actpass<br class="">a=fingerprint:SHA-256<br class="">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<br class="">a=sendrecv<br class=""><br class="">thats why i got Failed to set remote offer sdp: Called with SDP without<br class="">ice-ufrag and ice-pwd<br class=""><br class="">Waiting for your advice ---thanks<br class=""><br class=""><br class=""><br class=""><br class="">-- <br class="">Best regards<br class="">Antony<br class="">??? (066) 919-75-33<br class="">??? (063) 656-43-40<br class="">satskiy.a@gmail.com <mail%3Asatskiy.a@gmail.com><br class="">-------------- next part --------------<br class="">An HTML attachment was scrubbed...<br class="">URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150428/4f152d16/attachment-0001.html><br class=""><br class="">------------------------------<br class=""><br class="">Message: 3<br class="">Date: Tue, 28 Apr 2015 07:21:12 -0700<br class="">From: Motty Cruz <motty.cruz@gmail.com><br class="">To: Asterisk Users Mailing List - Non-Commercial Discussion<br class=""><span class="Apple-tab-span" style="white-space:pre"> </span><asterisk-users@lists.digium.com><br class="">Subject: Re: [asterisk-users] adding area code<br class="">Message-ID: <553F9758.5080907@gmail.com><br class="">Content-Type: text/plain; charset="windows-1252"; Format="flowed"<br class=""><br class="">this code worked for me,<br class=""><br class="">here is what I did and worked for me:<br class=""><br class="">exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444)<br class=""><br class="">exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80)<br class=""><br class=""><br class="">Thanks for you help!<br class=""><br class="">On 04/27/2015 02:56 PM, Matt Riddell wrote:<br class=""><blockquote type="cite" class=""><br class=""><blockquote type="cite" class="">On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz@gmail.com <br class=""><mailto:motty.cruz@gmail.com>> wrote:<br class=""><br class="">forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS.<br class=""><br class="">Thanks,<br class=""><br class=""><br class="">On 04/27/2015 02:38 PM, Motty Cruz wrote:<br class=""><blockquote type="cite" class="">here is what I have:<br class=""><br class="">exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381)<br class=""><br class="">exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1})<br class=""><br class="">exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80)<br class=""><br class="">not having success;<br class=""><br class="">"Got SIP reponse 503" Service Unavailable?<br class=""></blockquote></blockquote><br class="">Can you send us the console output when you make the call?<br class=""><br class="">--<br class="">Cheers,<br class=""><br class="">Matt Riddell<br class="">_______________________________________________<br class=""><br class="">http://www.venturevoip.com/news.php (Daily Asterisk News)<br class="">http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk)<br class="">http://www.venturevoip.com/exchange.php (Full ITSP Solution)<br class="">http://www.venturevoip.com/cc.php (Call Centre Solutions)<br class=""><br class=""><br class=""><br class=""></blockquote><br class="">-------------- next part --------------<br class="">An HTML attachment was scrubbed...<br class="">URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150428/7cc94c2f/attachment-0001.html><br class=""><br class="">------------------------------<br class=""><br class="">_______________________________________________<br class="">--Bandwidth and Colocation Provided by http://www.api-digital.com--<br class=""><br class="">asterisk-users mailing list<br class="">To UNSUBSCRIBE or update options visit:<br class=""> http://lists.digium.com/mailman/listinfo/asterisk-users<br class=""><br class="">End of asterisk-users Digest, Vol 129, Issue 32<br class="">***********************************************<br class=""></div></blockquote></div><br class=""></body></html>