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<P><FONT SIZE=2>in the SIP/2.0 180 Ringing, the SDP shows:<BR>
<BR>
a=sendonly<BR>
<BR>
this is "hold" by rfc 3264. then when the other end picks up, a new SDP is probably sent with<BR>
<BR>
a=sendrecv<BR>
<BR>
I believe your server is acting correctly.<BR>
<BR>
-----Original Message-----<BR>
From: asterisk-users-bounces@lists.digium.com on behalf of Tarek Sawah<BR>
Sent: Fri 4/30/2010 12:11 PM<BR>
To: Asterisk Users<BR>
Subject: Re: [asterisk-users] Strange Invite issue<BR>
<BR>
<BR>
Before posting let me mention that this doesn't happen with ALL destination on this provider.. some destination doesn't face this problem .. but this is a sample call<BR>
<BR>
<BR>
[K -- Executing [0020100324519@a2billing:1] [1;36;40mDeadAGI[0;37;40m("[1;35;40mSIP/58169-ac47fda0[0;37;40m", "[1;35;40ma2billing.php|1[0;37;40m") in new stack<BR>
[K -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php<BR>
-- AGI Script Executing Application: (Dial) Options: (SIP/PROVIDER1/20100324519|60|HL(166986000:61000:30000)) -- Limit Data for this call: > timelimit = 166986000 > play_warning = 61000 > play_to_caller = yes > play_to_callee = no > warning_freq = 30000 > start_sound = (null) > warning_sound = timeleft > end_sound = (null)Audio is at 100.X.Y.Z port 13984Adding codec 0x100 (g729) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 195.X.Y.Z:5060:INVITE sip:20100324519@195.X.Y.Z SIP/2.0<BR>
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport<BR>
From: "58169" <sip:58169@100.X.Y.Z>;tag=as00522e07<BR>
To: <sip:20100324519@195.X.Y.Z><BR>
Contact: <sip:58169@100.X.Y.Z><BR>
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba@100.X.Y.Z<BR>
CSeq: 102 INVITE<BR>
User-Agent: Asterisk PBX<BR>
Max-Forwards: 70<BR>
Date: Fri, 30 Apr 2010 18:52:23 GMT<BR>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>
Supported: replaces<BR>
Content-Type: application/sdp<BR>
Content-Length: 267<BR>
<BR>
<BR>
v=0<BR>
o=root 12516 12516 IN IP4 100.X.Y.Z<BR>
s=session<BR>
c=IN IP4 100.X.Y.Z<BR>
t=0 0<BR>
m=audio 13984 RTP/AVP 18 101<BR>
a=rtpmap:18 G729/8000<BR>
a=fmtp:18 annexb=no<BR>
a=rtpmap:101 telephone-event/8000<BR>
a=fmtp:101 0-16<BR>
a=silenceSupp:off - - - -<BR>
a=ptime:20<BR>
a=sendrecv<BR>
<BR>
--- -- Called PROVIDER1/20100324519<BR>
[K <BR>
<--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 100 Trying<BR>
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060<BR>
From: "58169" <sip:58169@100.X.Y.Z>;tag=as00522e07<BR>
To: <sip:20100324519@195.X.Y.Z>;tag=gK02b3c8db<BR>
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba@100.X.Y.Z<BR>
CSeq: 102 INVITE<BR>
Content-Length: 0<BR>
<BR>
<BR>
<BR>
<-------------><BR>
[K --- (7 headers 0 lines) ---<BR>
[K <BR>
<--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 180 Ringing<BR>
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060<BR>
From: "58169" <sip:58169@100.X.Y.Z>;tag=as00522e07<BR>
To: <sip:20100324519@195.X.Y.Z>;tag=gK02b3c8db<BR>
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba@100.X.Y.Z<BR>
CSeq: 102 INVITE<BR>
Contact: <sip:20100324519@195.X.Y.Z:5060><BR>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH<BR>
Content-Length: 260<BR>
Content-Disposition: session; handling=required<BR>
Content-Type: application/sdp<BR>
<BR>
<BR>
v=0<BR>
o=Sonus_UAC 10183 6645 IN IP4 195.X.Y.Z<BR>
s=SIP Media Capabilities<BR>
c=IN IP4 195.219.240.5<BR>
t=0 0<BR>
m=audio 15846 RTP/AVP 18 101<BR>
a=rtpmap:18 G729/8000<BR>
a=fmtp:18 annexb=no<BR>
a=rtpmap:101 telephone-event/8000<BR>
a=fmtp:101 0-15<BR>
a=sendonly<BR>
a=maxptime:20<BR>
<BR>
<-------------><BR>
[K --- (11 headers 12 lines) ---<BR>
[K Found RTP audio format 18<BR>
[K Found RTP audio format 101<BR>
[K Peer audio RTP is at port 195.219.240.5:15846<BR>
[K Found audio description format G729 for ID 18<BR>
[K Found audio description format telephone-event for ID 101<BR>
[K Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)<BR>
[K Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<BR>
[K Peer audio RTP is at port 195.219.240.5:15846<BR>
[K -- SIP/PROVIDER1-1fd586a0 is ringing<BR>
[K -- Call on SIP/PROVIDER1-1fd586a0 placed on hold<BR>
[K -- Started music on hold, class 'default', on SIP/58169-ac47fda0<BR>
[K -- SIP/PROVIDER1-1fd586a0 is making progress passing it to SIP/58169-ac47fda0<BR>
[K sip show channels<BR>
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 195.X.Y.Z 2010032451 7f169cce700 00102/00000 0x100 (g729) Yes Init: INVITE 78.184.197.119 58169 AC8455D8edd 00101/160518 0x4 (ulaw) No Rx: INVITE 2 active SIP channels<BR>
[K <BR>
<--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 180 Ringing<BR>
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060<BR>
From: "58169" <sip:58169@100.X.Y.Z>;tag=as00522e07<BR>
To: <sip:20100324519@195.X.Y.Z>;tag=gK02b3c8db<BR>
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba@100.X.Y.Z<BR>
CSeq: 102 INVITE<BR>
Contact: <sip:20100324519@195.X.Y.Z:5060><BR>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH<BR>
Content-Length: 0<BR>
<BR>
<BR>
<BR>
<-------------><BR>
[K --- (9 headers 0 lines) ---<BR>
[K -- SIP/PROVIDER1-1fd586a0 is ringing <BR>
<BR>
<BR>
<BR>
<BR>
<BR>
-- Tarek Sawah <BR>
<BR>
Integrated Digital Systems<BR>
<BR>
CCNA, MCSE, RHCE, VoIP<BR>
<BR>
<BR>
USA: +1 347 562 2308<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
> Date: Thu, 29 Apr 2010 16:52:24 +0100<BR>
> From: list-asterisk@skycomuk.com<BR>
> To: asterisk-users@lists.digium.com<BR>
> Subject: Re: [asterisk-users] Strange Invite issue<BR>
><BR>
> Can you post a sip debug<BR>
><BR>
> Tarek Sawah wrote:<BR>
>> Greetings List.<BR>
>> I'm facing a strange issue with one of my providers.. after sending an INVITE request my server places the call on hold.. until the call is answered..<BR>
>> this is happening only with this provide although i have 3 other providers i route calls through..<BR>
>> can anyone explain what is going on?<BR>
>><BR>
>> --<BR>
>> Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308<BR>
>><BR>
>><BR>
>><BR>
>><BR>
>> <BR>
>> _________________________________________________________________<BR>
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