Hi Alex, here's the config and the sip debug output. Guide: xxx.xxx.xxx.xxx - the provider's cisco as5300 trunk ip add yyy.yy.yy.yy - our asterisk 1.6.2.14 server
sip config: type=peer disallow=all allow=g729 host=xxx.xxx.xxx.xxx fromdomain=xxx.xxx.xxx.xxx dtmfmode=rfc2833 nat=no canreinvite=yes context=from-trunk-sip-iaccess sip debug: v=0 o=root 249777024 249777024 IN IP4 yyy.yy.yy.yy s=Asterisk PBX 1.6.2.14 c=IN IP4 yyy.yy.yy.yy t=0 0 m=audio 13702 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: "6598715968" <sip:[email protected]>;tag=as6e218907 To: <sip:[email protected]> Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: [email protected] Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Retransmitting #3 (no NAT) to zzz.zz.zz.zz:5060: OPTIONS sip:zzz.zz.zz.zz SIP/2.0 Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport Max-Forwards: 70 From: "Unknown" <sip:[email protected]>;tag=as5c8e3f97 To: <sip:zzz.zz.zz.zz> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.14 Date: Fri, 06 Jan 2012 06:23:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:69.90.209.57:5060 ---> <-------------> Retransmitting #4 (no NAT) to zzz.zz.zz.zz:5060: OPTIONS sip:zzz.zz.zz.zz SIP/2.0 Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport Max-Forwards: 70 From: "Unknown" <sip:[email protected]>;tag=as5c8e3f97 To: <sip:zzz.zz.zz.zz> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.14 Date: Fri, 06 Jan 2012 06:23:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '[email protected]' Method: OPTIONS <--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: "6598715968" <sip:[email protected]>;tag=as6e218907 To: <sip:[email protected]>;tag=B6534850-EC6 Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: [email protected] Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: "6598715968" <sip:1234#[email protected]>;party=called;screen=no;privacy=off Contact: <sip:[email protected]:5060> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 223 v=0 o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx s=SIP Call c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18132 RTP/AVP 18 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 <-------------> --- (15 headers 10 lines) --- Found RTP audio format 18 Found audio description format G729 for ID 18 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port xxx.xxx.xxx.xxx:18132 <--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport From: "6598715968" <sip:[email protected]>;tag=as6e218907 To: <sip:[email protected]>;tag=B6534850-EC6 Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: [email protected] Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:[email protected]:5060> Supported: replaces Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 223 v=0 o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx s=SIP Call c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18132 RTP/AVP 18 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 <-------------> --- (15 headers 10 lines) --- list_route: hop: <sip:[email protected]:5060> set_destination: Parsing <sip:[email protected]:5060> for address/port to send to set_destination: set destination to xxx.xxx.xxx.xxx, port 5060 Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK17854b94;rport Max-Forwards: 70 From: "6598715968" <sip:[email protected]>;tag=as6e218907 To: <sip:[email protected]>;tag=B6534850-EC6 Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.14 Content-Length: 0 --- > Channel SIP/xxx.xxx.xxx.xxx-00003693 was answered. -- Executing [6591394459@a2billing-callback:1] DeadAGI("SIP/xxx.xxx.xxx.xxx-00003693", "a2billing.php,1,callback") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- AGI Script Executing Application: (DIAL) Options: (SIP/xxx.xxx.xxx.xxx/34546591394459,60,HRrL(370239000:61000:30000)) -- Limit Data for this call: > timelimit = 370239000 > play_warning = 61000 > play_to_caller = yes > play_to_callee = no > warning_freq = 30000 > start_sound = > warning_sound = timeleft > end_sound = == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at yyy.yy.yy.yy port 14212 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK4ea95f20;rport Max-Forwards: 70 From: "6598715968" <sip:[email protected]>;tag=as492477b7 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.14 Date: Fri, 06 Jan 2012 06:23:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 331 v=0 o=root 1686167830 1686167830 IN IP4 yyy.yy.yy.yy s=Asterisk PBX 1.6.2.14 c=IN IP4 yyy.yy.yy.yy t=0 0 m=audio 14212 RTP/AVP 18 0 8 3 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv To: <sip:[email protected]>;tag=B6534850-EC6 Date: Fri, 06 Jan 2012 04:51:39 GMT Call-ID: [email protected] Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: "6598715968" <sip:1234#[email protected]>;party=called;screen=no;privacy=off Contact: <sip:[email protected]:5060> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 223 v=0 o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx s=SIP Call c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18132 RTP/AVP 18 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 On Mon, Jan 9, 2012 at 4:33 PM, Alex Balashov <[email protected]>wrote: > You are hereby encouraged to post your AS5300 IOS config, sip.conf peer > declaration, and packet capture. Those three things would aid greatly in > diagnosis, especially the capture. > > -- > This message was painstakingly thumbed out on my mobile, so apologies for > brevity, errors, and general sloppiness. > > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On Jan 9, 2012, at 3:20 AM, Roi Stork <[email protected]> wrote: > > > Hi, > > > > We have a problem connecting to a Cisco AS5300 trunk. > > > > We set the sip peer to allow only g729. The call attempt is able to > connect, but when answered, no audio is heard or transmitted. > > > > Our asterisk version is 1.6.2.14 . Codec is licensed, bought from Digium. > > > > We do not have this problem on our other providers using asterisk and > other non-cisco systems. > > Anyone else having this same problem? > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
