Summary:ASTERISK-06793: Problem with '183 Session Progress'
Reporter:Arif-Uz-Zaman (arifzaman)Labels:
Date Opened:2006-04-17 08:44:45Date Closed:2011-06-07 14:02:57
Versions:Frequency of
Environment:Attachments:( 0) debug_with_rtpsip_session_progress_asterisk-
( 1) new_debugsip_session_progress_asterisk-
( 2) sip_session_progress_debug.txt
( 3) sip_session_progress_debugasterisk-
Description:I'm initiating a call from a GRANDSTREAM IP Phone to reach to extensions on a 3rd party VOIP Provider. But when the other end sends back ‘183 Session Progress’ with SDP and next it sends back ‘200 OK’. But from asterisk debug information, I nave got the following line for 183.

Thanks in advance.


But when i'm initiating a call with two asterisk end points, then it works nicely.

version: asterisk-1.0.7
Comments:By: Joshua C. Colp (jcolp) 2006-04-17 10:01:05

Update to 1.2, I'm not going to look into this since it's for 1.0 which is now old and out dated in comparison.

By: Arif-Uz-Zaman (arifzaman) 2006-04-17 22:57:26

i've tried with asterisk- also, but i have got the same problem. so if there is any solution or patch regarding this issue, you are highly appreciated to try with that one.

Please let me know, if you need anything from me.

By: Joshua C. Colp (jcolp) 2006-04-17 22:58:58

Can you give me updated debug using

By: Arif-Uz-Zaman (arifzaman) 2006-04-18 05:04:04

Thanks for your quick response.

The updated debug is uploaded for asterisk- Here, '183 Session Progress’ is forwarder to calling party which is come from called party. But there is no Ring-Back-Tone, what is expected.

pls consider the issue.

By: Arif-Uz-Zaman (arifzaman) 2006-04-18 08:27:12

pls try to check with the new uploaded debug.

File Name: new_debug(sip_session_progress_asterisk-

By: Joshua C. Colp (jcolp) 2006-04-18 09:47:47

If you get a 183 Session Progress, then ringing should be provided inband as audio from the device. Can you do an rtp debug and see if audio is coming from it?

By: Arif-Uz-Zaman (arifzaman) 2006-04-18 22:34:48

here is the debug with rtp.

File Name: debug_with_rtp(sip_session_progress_asterisk-

By: Serge Vecher (serge-v) 2006-04-19 08:31:03

arifzaman: you have forgotten to attach the debug_with_rtp(sip_session_progress_asterisk-

By: Arif-Uz-Zaman (arifzaman) 2006-04-19 22:46:39

sorry for the Inconvenience. pls check it now.

By: Olle Johansson (oej) 2006-04-25 05:18:17

This debug file shows us receiving 183 session progress and sending 183 session progress to the phone, with rtp audio. So where's the problem?

By: Arif-Uz-Zaman (arifzaman) 2006-04-26 01:53:31

here, i'm not geeting any Ring-Back-Tone. so how could i know, whether the called party phone is Ringing or not.

By: Serge Vecher (serge-v) 2006-05-04 16:03:35

Asterisk is acting within RFC specs. If your Grandstream phone does not produce ringing after receiving 183 from Asterisk, then it is a bug within that phone. I suggest that you contact Grandstream for support. This will likely be resolved with a firmware upgrade.

By: Serge Vecher (serge-v) 2006-05-12 11:40:59

User agent issue here, no problem with Asterisk.