[asterisk-users] Cisco AS5300 and Digium g729A codec

Tue Jan 10 01:51:11 CST 2012

The problem has been fixed.

We are able to hear audio in our calls after adding these lines in the
AS5300 config:

sip-ua
g729-annexb override

There's an issue regarding codec matching in IOS versions 12.3(18) or higher:
https://supportforums.cisco.com/docs/DOC-3186

On Tue, Jan 10, 2012 at 10:30 AM, Roi Stork <roi.stork at gmail.com> wrote:
>
> 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
>>
>>
>