Here's the cisco AS5300 settings from our provider codec preference 1 g729r8 codec preference 2 g729br8 codec preference 3 g723r53 codec preference 4 g723r63 codec preference 5 g723ar53 codec preference 6 g723ar63
On Mon, Jan 9, 2012 at 5:18 PM, Roi Stork <roi.st...@gmail.com> wrote: > 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:6598715...@yyy.yy.yy.yy>;tag=as6e218907 > To: <sip:34546598715...@xxx.xxx.xxx.xxx> > Date: Fri, 06 Jan 2012 04:51:39 GMT > Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy > 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:unkn...@yyy.yy.yy.yy>;tag=as5c8e3f97 > To: <sip:zzz.zz.zz.zz> > Contact: <sip:unkn...@yyy.yy.yy.yy> > Call-ID: 7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy > 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:unkn...@yyy.yy.yy.yy>;tag=as5c8e3f97 > To: <sip:zzz.zz.zz.zz> > Contact: <sip:unkn...@yyy.yy.yy.yy> > Call-ID: 7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy > 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 '7bdb028f789e3afa58b22db472d9d...@yyy.yy.yy.yy' > 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:6598715...@yyy.yy.yy.yy>;tag=as6e218907 > To: <sip:34546598715...@xxx.xxx.xxx.xxx>;tag=B6534850-EC6 > Date: Fri, 06 Jan 2012 04:51:39 GMT > Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy > 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#6598715...@xxx.xxx.xxx.xxx>;party=called;screen=no;privacy=off > Contact: <sip:34546598715...@xxx.xxx.xxx.xxx: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:6598715...@yyy.yy.yy.yy>;tag=as6e218907 > To: <sip:34546598715...@xxx.xxx.xxx.xxx>;tag=B6534850-EC6 > Date: Fri, 06 Jan 2012 04:51:39 GMT > Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy > 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:34546598715...@xxx.xxx.xxx.xxx: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:34546598715...@xxx.xxx.xxx.xxx:5060> > set_destination: Parsing <sip:34546598715...@xxx.xxx.xxx.xxx: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:34546598715...@xxx.xxx.xxx.xxx:5060 SIP/2.0 > Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK17854b94;rport > Max-Forwards: 70 > From: "6598715968" <sip:6598715...@yyy.yy.yy.yy>;tag=as6e218907 > To: <sip:34546598715...@xxx.xxx.xxx.xxx>;tag=B6534850-EC6 > Contact: <sip:6598715...@yyy.yy.yy.yy> > Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy > 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:34546591394...@xxx.xxx.xxx.xxx SIP/2.0 > Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK4ea95f20;rport > Max-Forwards: 70 > From: "6598715968" <sip:6598715...@yyy.yy.yy.yy>;tag=as492477b7 > To: <sip:34546591394...@xxx.xxx.xxx.xxx> > Contact: <sip:6598715...@yyy.yy.yy.yy> > Call-ID: 4d866149766030b331fee79f62bc2...@yyy.yy.yy.yy > 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:34546598715...@xxx.xxx.xxx.xxx>;tag=B6534850-EC6 > Date: Fri, 06 Jan 2012 04:51:39 GMT > Call-ID: 7e54da423b0e6e457475ab17694e5...@yyy.yy.yy.yy > 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#6598715...@xxx.xxx.xxx.xxx>;party=called;screen=no;privacy=off > Contact: <sip:34546598715...@xxx.xxx.xxx.xxx: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 > <abalas...@evaristesys.com>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 <roi.st...@gmail.com> 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