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Summary:ASTERISK-09581: Progess tone not passed.
Reporter:seal (seal)Labels:
Date Opened:2007-06-04 05:11:40Date Closed:2011-06-07 14:08:10
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
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Issues:
Environment:Attachments:
Description:Hi, i have this scenario:

City E1 trunks <-> Cisco AS5350 <-SIP-> Asterisk 1.2.18 <-SIP-> Softphone XLITE or Linksys SPA 9XX

When im trying to call to outside im not getting progress inband generated by my operator but only ringing signal generated by phone.

AS send progress inband:

<-- SIP read from 192.168.XXX.XXX:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.XXX.XXX:5060;branch=z9hG4bK39eff666;rport
From: "XXXXX XXXXX" <sip:4XX@192.168.XXX.XXX>;tag=as5a1e5d7b
To: <sip:060xxxxxxx@192.168.XXX.XXX>;tag=140EDF90-202C
Date: Mon, 04 Jun 2007 09:46:00 GMT
Call-ID: 3d448eb86e9e0383193ab8321b348040@192.168.XXX.XXX
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

<-- SIP read from 192.168.3.9:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.XXX.XXX:5060;branch=z9hG4bK39eff666;rport
From: "XXXXX XXXXX" <sip:455@192.168.XXX.XXX>;tag=as5a1e5d7b
To: <sip:060xxxxxxx@192.168.XXX.XXX>;tag=140EDF90-202C
Date: Mon, 04 Jun 2007 09:46:00 GMT
Call-ID: 3d448eb86e9e0383193ab8321b348040@192.168.XXX.XXX
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:0602806423@192.168.XXX.XXX:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 199

v=0
o=CiscoSystemsSIP-GW-UserAgent 2846 9147 IN IP4 192.168.3.9
s=SIP Call
c=IN IP4 192.168.3.9
t=0 0
m=audio 18814 RTP/AVP 18
c=IN IP4 192.168.3.9
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no

Found RTP audio format 18
Peer audio RTP is at port 192.168.3.9:18814
Found description format G729
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
   -- SIP/cisco-081ff4b0 is making progress passing it to SIP/4XX-b6a219e8

ASTERISK PASSESS RINGING TO PHONE:

-- Called cisco/0602806423
Transmitting (NAT) to 192.168.XXX.XXX:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.XXX.XXX:5060;branch=z9hG4bK-690cf669;received=192.168.XXX.XXX
From: "XXX" <sip:4XX@192.168.XXX.XXX>;tag=21dd70682d9caf7do0
To: <sip:0060xxxxxxx@192.168.XXX.XXX>;tag=as3ea0d3c5
Call-ID: 9225d704-d5101991@192.168.XXX.XXX
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0060xxxxxxx@192.168.XXX.XXX>
Content-Length: 0

I looked with Ethereal AS sends to Asterisk Progress sound...

Changing progressinband in sip.conf to any value changes nothing.

Comments:By: Joshua C. Colp (jcolp) 2007-06-04 06:20:30

Can you please provide your Dial line? I suspect you have the 'r' argument which tells it to send ringing regardless.
By: seal (seal) 2007-06-04 06:39:35

Yes, that was my mistake :( Removing "r" helps. Sory.

You can close this.
By: Joshua C. Colp (jcolp) 2007-06-04 06:43:25

Closed as it was a configuration issue.