Summary:ASTERISK-08817: Going from hold to unhold status fails with Uniden phones
Reporter:geisj (geisj)Labels:
Date Opened:2007-02-15 14:04:34.000-0600Date Closed:2007-06-19 08:31:19
Versions:Frequency of
Environment:Attachments:( 0) call_drop.txt
( 1) verbose.txt
( 2) verbose1.2.txt
( 3) verbose1.4.txt
Description:I am using UIP200 phones however I dont think the phone is the
issue as this worked in 1.2.X

I call ext 210, another person calls ext 210.
210 flashes and accepts the other call and I'm on hold.
210 then flashes back to me and the other person hangs up.
I am speaking with 210 again. After a few seconds the entire call is dropped.

Comments:By: Joshua C. Colp (jcolp) 2007-02-15 14:24:18.000-0600

Please attach a sip debug and complete console output. Thanks.

By: geisj (geisj) 2007-02-15 15:48:07.000-0600

I upload the console output.

Call came from 550 to 528.
second call came from 526 to 528.
flashed to answer, then flashed back, then 526 hungup.
short time later the 550 - 528 call is dropped.


By: Joshua C. Colp (jcolp) 2007-02-16 14:38:52.000-0600

I need the full console output and sip debug, not just the part you uploaded. I have to be able to piece together the entire SIP dialog and calls.

By: geisj (geisj) 2007-02-17 07:28:17.000-0600

Can you tell me the command(s) you want me to run.
I thought I did what you were asking for.


By: Serge Vecher (serge-v) 2007-02-19 11:20:49.000-0600

These will do:
1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterisk with the following command:
  'asterisk -Tvvvvvdddddngc | tee /tmp/verbosedebug.txt'
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Reproduce the problem
6) Trim startup information and attach verbosedebug.txt to the issue.

By: geisj (geisj) 2007-02-22 16:29:18.000-0600

I did the commands as requested.

526 called 528, answered
550 called 528, FLASHED and answered
528 flashed back, 550 hangs up

after a short time the call is dropped between 528 and 526.


By: Joshua C. Colp (jcolp) 2007-02-22 16:56:59.000-0600

What happens if you set canreinvite to no? It looks like one of the devices involved can't handle reinviting back to Asterisk when MOH kicks in (which should be proper to do).

By: geisj (geisj) 2007-02-23 07:40:22.000-0600

I set all 3 phones involved to canreinvite=no
stopped asterisk
started asterisk
did the exact same calling and after a few
seconds it still hangs up the call.


By: Serge Vecher (serge-v) 2007-02-23 09:03:23.000-0600

any changes with 1.4 checked out from svn?

By: geisj (geisj) 2007-02-23 10:04:53.000-0600

No this is straight 1.4.0


By: Serge Vecher (serge-v) 2007-02-23 10:14:30.000-0600

can you please checkout 1.4 from svn and test?

By: geisj (geisj) 2007-03-05 07:16:29.000-0600

I just tried with 1.4.1 and the same thing is happening.


By: geisj (geisj) 2007-03-08 12:02:30.000-0600

What else can I do to provide information.
This issue is becoming a problem as I am dropping my calls from customers.


By: Serge Vecher (serge-v) 2007-03-08 12:19:30.000-0600

ok, since this worked in 1.2.x, please provide the log with instructions above from whichever latest release it worked with. Then attach a new log for 1.4.1

By: geisj (geisj) 2007-03-10 11:12:53.000-0600

ok, I have wentback to 1.2 and verififed it works. I uploaded that part.
Then I went back to 1.4 and after a few seconds 10 or so it does hangup
after flashing back to the first call.

I was using 1.2.13
I was using 1.4.1

let me know,


By: Olle Johansson (oej) 2007-03-14 12:14:12

We're sending a 200 OK on the re-invite, the phone does not answer with an ACK, so finally we hang up the call. Need to analyze this a bit more.

By: geisj (geisj) 2007-03-19 09:48:52

Anything new on this... I just accidentally did it again and dropped my call.


By: Serge Vecher (serge-v) 2007-03-19 09:53:02

geisj: following oej's analysis, the phone must ack a 200 ok from Asterisk as per the standard. You may want to let Uniden know.

By: Olle Johansson (oej) 2007-05-16 03:48:26

Any new log files? Please test with latest 1.4 from svn, since there has been a few changes for hold/unhold.

By: Olle Johansson (oej) 2007-05-18 05:29:35

No answer from reporter. Assume either bug in phone firmware or problem fixed with recent hold changes.

By: geisj (geisj) 2007-05-18 06:33:07

I have been travelling - sorry I didnt reply...

I can't easily try the SVN as it is on a production system and I travel alot.

Uniden actually told me there will be no changes to the firmware as
the team is no longer together.

All I know is it works under 1.2 and not 1.4.


By: Olle Johansson (oej) 2007-05-18 06:53:57

Well, according to the logs, there is a bug in your uniden phones. Nothing we can fix. Asterisk has improved SIP compatibility for every release and this is propably what you discover.

By: Joshua C. Colp (jcolp) 2007-06-19 08:31:18

Per oej's comment this appears to be a bug in the SIP implementation of the phone.