[asterisk-users] Strange Invite issue

David White David.White at watchguard.com
Fri Apr 30 15:09:05 CDT 2010


in the SIP/2.0 180 Ringing, the SDP shows:

a=sendonly

this is "hold" by rfc 3264.  then when the other end picks up, a new SDP is probably sent with 

a=sendrecv

I believe your server is acting correctly.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com on behalf of Tarek Sawah
Sent: Fri 4/30/2010 12:11 PM
To: Asterisk Users
Subject: Re: [asterisk-users] Strange Invite issue
 

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


      -- Executing [0020100324519 at a2billing:1] DeadAGI("SIP/58169-ac47fda0", "a2billing.php|1") in new stack
      -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
      -- 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 at 195.X.Y.Z SIP/2.0
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport
From: "58169" <sip:58169 at 100.X.Y.Z>;tag=as00522e07
To: <sip:20100324519 at 195.X.Y.Z>
Contact: <sip:58169 at 100.X.Y.Z>
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 30 Apr 2010 18:52:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 267


v=0
o=root 12516 12516 IN IP4 100.X.Y.Z
s=session
c=IN IP4 100.X.Y.Z
t=0 0
m=audio 13984 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---    -- Called PROVIDER1/20100324519
  
<--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 100 Trying
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060
From: "58169" <sip:58169 at 100.X.Y.Z>;tag=as00522e07
To: <sip:20100324519 at 195.X.Y.Z>;tag=gK02b3c8db
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
CSeq: 102 INVITE
Content-Length: 0



<------------->
  --- (7 headers 0 lines) ---
  
<--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060
From: "58169" <sip:58169 at 100.X.Y.Z>;tag=as00522e07
To: <sip:20100324519 at 195.X.Y.Z>;tag=gK02b3c8db
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
CSeq: 102 INVITE
Contact: <sip:20100324519 at 195.X.Y.Z:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length:  260
Content-Disposition: session; handling=required
Content-Type: application/sdp


v=0
o=Sonus_UAC 10183 6645 IN IP4 195.X.Y.Z
s=SIP Media Capabilities
c=IN IP4 195.219.240.5
t=0 0
m=audio 15846 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendonly
a=maxptime:20

<------------->
  --- (11 headers 12 lines) ---
  Found RTP audio format 18
  Found RTP audio format 101
  Peer audio RTP is at port 195.219.240.5:15846
  Found audio description format G729 for ID 18
  Found audio description format telephone-event for ID 101
  Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
  Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  Peer audio RTP is at port 195.219.240.5:15846
      -- SIP/PROVIDER1-1fd586a0 is ringing
      -- Call on SIP/PROVIDER1-1fd586a0 placed on hold
      -- Started music on hold, class 'default', on SIP/58169-ac47fda0
      -- SIP/PROVIDER1-1fd586a0 is making progress passing it to SIP/58169-ac47fda0
  sip show channels
  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
  
<--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060
From: "58169" <sip:58169 at 100.X.Y.Z>;tag=as00522e07
To: <sip:20100324519 at 195.X.Y.Z>;tag=gK02b3c8db
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
CSeq: 102 INVITE
Contact: <sip:20100324519 at 195.X.Y.Z:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 0



<------------->
  --- (9 headers 0 lines) ---
      -- SIP/PROVIDER1-1fd586a0 is ringing 





-- Tarek Sawah 

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP


USA: +1 347 562 2308






> Date: Thu, 29 Apr 2010 16:52:24 +0100
> From: list-asterisk at skycomuk.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Strange Invite issue
> 
> Can you post a sip debug
> 
> Tarek Sawah wrote:
>> Greetings List.
>> 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.. 
>> this is happening only with this provide although i have 3 other providers i route calls through.. 
>> can anyone explain what is going on?
>> 
>> --
>> Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308
>> 
>> 
>> 
>> 
>>  		 	   		  
>> _________________________________________________________________
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> 
> 
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