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bruce bruce

2010-Jun-10 15:10 UTC

[asterisk-users] Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?

Hi Guys,

I have Spikko setup as provider of DID and outbound routes and I can make
calls out but no inbound calls via DID can be made. I did a sip debug which
is reported below. I never receive the call though, I have a catch all in my
inbound routes and it doesn't hit my context at all or not sip invite comes
in:


FreePBX:

Trunk Name:
*Spikko*

Peer Detail
*username=MyUsername*
*type=friend*
*secret=MyPassword*
*host=sip.spikko.com*
*nat=no*
*port=5090*
*fromuser=MyUsername*
*disallow=all*
*allow=g729&gsm&ulaw&alaw*

Register String:
*MyUsername:MyPassword at sip.spikko.com:5090/MyUsername*


Inbound Router:
*Send Any DID and ANY CID to Music on Hold*


Sip debug:

*Really destroying SIP dialog '
417b3c8f3a97a82d4629343a53b2feb6 at 177.177.177.177' Method: REGISTER*
*tel*CLI>*
*<--- SIP read from UDP:82.80.252.29:5090 --->*
*INVITE sip:MyUsername at 177.177.177.177 <sip%3AMyUsername at
177.177.177.177>SIP/2.0
*
*Via: SIP/2.0/UDP 82.80.252.234:5090;branch=z9hG4bK07b38a0c;rport*
*From: "Unknown" <sip:Unknown at
82.80.252.234:5090>;tag=as24089849*
*To: <sip:MyUsername at 177.177.177.177 <sip%3AMyUsername at
177.177.177.177>>*
*Contact: <sip:Unknown at 82.80.252.234:5090>*
*Call-ID: 55a4cf1f4e5575e97f8b3b23495f0a43 at 82.80.252.234*
*CSeq: 102 INVITE*
*User-Agent: AG1*
*Max-Forwards: 70*
*Date: Thu, 10 Jun 2010 14:58:09 GMT*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY*
*Supported: replaces*
*Content-Type: application/sdp*
*Content-Length: 331*
*
*
*v=0*
*o=root 6129 6129 IN IP4 82.80.252.234*
*s=session*
*c=IN IP4 82.80.252.234*
*t=0 0*
*m=audio 10172 RTP/AVP 18 3 97 101*
*a=rtpmap:18 G729/8000*
*a=fmtp:18 annexb=no*
*a=rtpmap:3 GSM/8000*
*a=rtpmap:97 iLBC/8000*
*a=fmtp:97 mode=30*
*a=rtpmap:101 telephone-event/8000*
*a=fmtp:101 0-16*
*a=silenceSupp:off - - - -*
*a=ptime:20*
*a=sendrecv*
*
*
*<------------->*
*--- (14 headers 16 lines) ---*
*Using INVITE request as basis request -
55a4cf1f4e5575e97f8b3b23495f0a43 at 82.80.252.234*
*Found peer 'Spikko' for 'Unknown' from 82.80.252.29:5090*


I also sometimes get this even though trunk shows registered and can make
calls out:
*<--- Transmitting (no NAT) to 82.80.252.29:5090 --->*
*SIP/2.0 489 Bad event*
*Via: SIP/2.0/UDP 82.80.252.234:5090
;branch=z9hG4bK463b703d;received=82.80.252.29;rport=5090*
*From: "asterisk" <sip:asterisk at
82.80.252.234:5090>;tag=as4af8cf81*
*To: <sip:saarshalom at 173.203.29.102 <sip%3Asaarshalom at
173.203.29.102>>;tag=as64c0ba34**Call-ID: 497197a679122f5d448d324f571f3c3e at 82.80.252.234*
*CSeq: 102 NOTIFY*
*Server: Asterisk PBX 1.6.2.7*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO*
*Supported: replaces, timer*
*Content-Length: 0*

Thanks,
Bruce
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Zeeshan Zakaria

2010-Jun-10 15:35 UTC

head link

[asterisk-users] Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?

FreePBX questions should be asked at FreePBX forums.

As for the asterisk part, where are you defining the context to receive
incoming calls? Probably in the trunk settings (Peer Details) you need to
add "context=from-trunk" if FreePBX still uses it as the default
context for
incoming calls.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-10 11:24 AM, "bruce bruce" <bruceb444 at gmail.com>
wrote:

Hi Guys,

I have Spikko setup as provider of DID and outbound routes and I can make
calls out but no inbound calls via DID can be made. I did a sip debug which
is reported below. I never receive the call though, I have a catch all in my
inbound routes and it doesn't hit my context at all or not sip invite comes
in:


FreePBX:

Trunk Name:
*Spikko*

Peer Detail
*username=MyUsername*
*type=friend*
*secret=MyPassword*
*host=sip.spikko.com*
*nat=no*
*port=5090*
*fromuser=MyUsername*
*disallow=all*
*allow=g729&gsm&ulaw&alaw*

Register String:
*MyUsername:MyPassword at sip.spikko.com:5090/MyUsername*


Inbound Router:
*Send Any DID and ANY CID to Music on Hold*


Sip debug:

*Really destroying SIP dialog '
417b3c8f3a97a82d4629343a53b2feb6 at 177.177.177.177' Method: REGISTER*
*tel*CLI>*
*<--- SIP read from UDP:82.80.252.29:5090 --->*
*INVITE sip:MyUsername at 177.177.177.177 <sip%3AMyUsername at
177.177.177.177>SIP/2.0
*
*Via: SIP/2.0/UDP 82.80.252.234:5090;branch=z9hG4bK07b38a0c;rport*
*From: "Unknown" <sip:Unknown at
82.80.252.234:5090>;tag=as24089849*
*To: <sip:MyUsername at 177.177.177.177 <sip%3AMyUsername at
177.177.177.177>>*
*Contact: <sip:Unknown at 82.80.252.234:5090>*
*Call-ID: 55a4cf1f4e5575e97f8b3b23495f0a43 at 82.80.252.234*
*CSeq: 102 INVITE*
*User-Agent: AG1*
*Max-Forwards: 70*
*Date: Thu, 10 Jun 2010 14:58:09 GMT*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY*
*Supported: replaces*
*Content-Type: application/sdp*
*Content-Length: 331*
*
*
*v=0*
*o=root 6129 6129 IN IP4 82.80.252.234*
*s=session*
*c=IN IP4 82.80.252.234*
*t=0 0*
*m=audio 10172 RTP/AVP 18 3 97 101*
*a=rtpmap:18 G729/8000*
*a=fmtp:18 annexb=no*
*a=rtpmap:3 GSM/8000*
*a=rtpmap:97 iLBC/8000*
*a=fmtp:97 mode=30*
*a=rtpmap:101 telephone-event/8000*
*a=fmtp:101 0-16*
*a=silenceSupp:off - - - -*
*a=ptime:20*
*a=sendrecv*
*
*
*<------------->*
*--- (14 headers 16 lines) ---*
*Using INVITE request as basis request -
55a4cf1f4e5575e97f8b3b23495f0a43 at 82.80.252.234*
*Found peer 'Spikko' for 'Unknown' from 82.80.252.29:5090*


I also sometimes get this even though trunk shows registered and can make
calls out:
*<--- Transmitting (no NAT) to 82.80.252.29:5090 --->*
*SIP/2.0 489 Bad event*
*Via: SIP/2.0/UDP 82.80.252.234:5090
;branch=z9hG4bK463b703d;received=82.80.252.29;rport=5090*
*From: "asterisk" <sip:asterisk at
82.80.252.234:5090>;tag=as4af8cf81*
*To: <sip:saarshalom at 173.203.29.102 <sip%3Asaarshalom at
173.203.29.102>>;tag=as64c0ba34**Call-ID: 497197a679122f5d448d324f571f3c3e at 82.80.252.234*
*CSeq: 102 NOTIFY*
*Server: Asterisk PBX 1.6.2.7*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO*
*Supported: replaces, timer*
*Content-Length: 0*

Thanks,
Bruce

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