[asterisk-users] Cisco AS5300 and Digium g729A codec
Roi Stork
roi.stork at gmail.com
Mon Jan 9 20:30:18 CST 2012
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.stork at 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:6598715968 at yyy.yy.yy.yy>;tag=as6e218907
> To: <sip:34546598715968 at xxx.xxx.xxx.xxx>
> Date: Fri, 06 Jan 2012 04:51:39 GMT
> Call-ID: 7e54da423b0e6e457475ab17694e5165 at 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:Unknown at yyy.yy.yy.yy>;tag=as5c8e3f97
> To: <sip:zzz.zz.zz.zz>
> Contact: <sip:Unknown at yyy.yy.yy.yy>
> Call-ID: 7bdb028f789e3afa58b22db472d9dfb5 at 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:Unknown at yyy.yy.yy.yy>;tag=as5c8e3f97
> To: <sip:zzz.zz.zz.zz>
> Contact: <sip:Unknown at yyy.yy.yy.yy>
> Call-ID: 7bdb028f789e3afa58b22db472d9dfb5 at 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 '7bdb028f789e3afa58b22db472d9dfb5 at 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:6598715968 at yyy.yy.yy.yy>;tag=as6e218907
> To: <sip:34546598715968 at xxx.xxx.xxx.xxx>;tag=B6534850-EC6
> Date: Fri, 06 Jan 2012 04:51:39 GMT
> Call-ID: 7e54da423b0e6e457475ab17694e5165 at 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#6598715968 at xxx.xxx.xxx.xxx>;party=called;screen=no;privacy=off
> Contact: <sip:34546598715968 at 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:6598715968 at yyy.yy.yy.yy>;tag=as6e218907
> To: <sip:34546598715968 at xxx.xxx.xxx.xxx>;tag=B6534850-EC6
> Date: Fri, 06 Jan 2012 04:51:39 GMT
> Call-ID: 7e54da423b0e6e457475ab17694e5165 at 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:34546598715968 at 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:34546598715968 at xxx.xxx.xxx.xxx:5060>
> set_destination: Parsing <sip:34546598715968 at 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:34546598715968 at 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:6598715968 at yyy.yy.yy.yy>;tag=as6e218907
> To: <sip:34546598715968 at xxx.xxx.xxx.xxx>;tag=B6534850-EC6
> Contact: <sip:6598715968 at yyy.yy.yy.yy>
> Call-ID: 7e54da423b0e6e457475ab17694e5165 at 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 at 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:34546591394459 at 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:6598715968 at yyy.yy.yy.yy>;tag=as492477b7
> To: <sip:34546591394459 at xxx.xxx.xxx.xxx>
> Contact: <sip:6598715968 at yyy.yy.yy.yy>
> Call-ID: 4d866149766030b331fee79f62bc2030 at 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:34546598715968 at xxx.xxx.xxx.xxx>;tag=B6534850-EC6
> Date: Fri, 06 Jan 2012 04:51:39 GMT
> Call-ID: 7e54da423b0e6e457475ab17694e5165 at 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#6598715968 at xxx.xxx.xxx.xxx>;party=called;screen=no;privacy=off
> Contact: <sip:34546598715968 at 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 <abalashov at 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.stork at 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
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120110/fe8f61ee/attachment-0001.htm>
More information about the asterisk-users
mailing list