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Summary:ASTERISK-15429: Asterisk does not send "183 Session Progress" when dialing through a dahdi analog line
Reporter:frawd (frawd)Labels:
Date Opened:2010-01-14 09:28:49.000-0600Date Closed:2010-06-01 20:22:41
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) asterisk_vvv_ddd_log.txt
Description:Scenario is a simple bridge between a SIP phone and an FXO line with polarityswitch options turned on (and working). While the PSTN destination hasn't answered yet, the SIP phone keeps silent and does not generate a ring tone.

In 1.4 versions of Asterisk, after dialing the digits in the FXO, Asterisk was sending a "183 Session Progress" back to the phone so it could generate a progress ring.

In 1.6.2.1-rc1, it does not send it, probably related to some strange failure to read frames:
[Jan 14 16:13:00] DEBUG[10593]: audiohook.c:248 audiohook_read_frame_both: Failed to get 160 samples from read factory 0xa71d70


****** ADDITIONAL INFORMATION ******

Attached is a verbose+debug+sip debug log showing the issue (before the destination answers).
Comments:By: frawd (frawd) 2010-02-02 16:16:47.000-0600

Please change to asterisk version 1.6.2.1 as rc1 doesn't exist anymore!
By: frawd (frawd) 2010-02-12 10:39:28.000-0600

Still there in 1.6.2.3, I'm beginning to think that it is normal and that the behavior of 1.4 was wrong. Analog lines don't send progress Ringing so it is quite logical for * not to forward anything to the phone.

Please confirm this and close the bug if necessary.

As a workaround, if I dial through an Analog line, I place the "r" option to Dial.
By: David Woolley (davidw) 2010-04-23 10:24:17

I think this may duplicate DAHLIN-186

In any case, if there is a problem, it is dahdi (driver or channel) related, not SIP related.
By: David Woolley (davidw) 2010-04-23 10:39:27

On the other hand, the UPGRADE.txt for 1.6.2 says that 183 Progress is not sent by default, so maybe it is a non-bug in chan_sip!
By: frawd (frawd) 2010-04-23 10:50:14

Agreed for it being a dahdi issue, as it works ok in SIP to SIP or SIP to DAHDI-PRI situations.

You can close this one and refer to DAHLIN-186
By: frawd (frawd) 2010-04-23 10:53:59

On the other hand, DAHLIN-186 refers to dahdi 2.3.0 and I use dahdi 2.2.1
By: frawd (frawd) 2010-04-23 11:06:41

Ok, after reading the UPGRADE.txt, it appears this is a total non-bug as you mentioned.

It works perfectly by adding a Progress() before my Dial command.

This bug can really be closed.
By: frawd (frawd) 2010-04-23 11:06:51

Thanks by the way.
By: Paul Belanger (pabelanger) 2010-06-01 20:22:40

Thanks for your comments. This does not appear to be a bug report and we are closing it. We appreciate the difficulties you are facing, but it would make more sense to raise your question in the support tracker, http://www.asterisk.org/support