DAHLIN-00171: On call-waiting events, dropped audio on first call then switch to second call when first call ends without 'flash'

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Summary: DAHLIN-00171: On call-waiting events, dropped audio on first call then switch to second call when first call ends without 'flash'

Reporter: Kirk Wolff (kirkawolff) Labels:

Date Opened: 2010-01-18 19:07:07.000-0600 Date Closed: 2019-05-31 09:49:38

Priority: Major Regression? No

Status: Closed/Complete Components: General

Versions: 2.2.0.2 Frequency of
Occurrence

Related
Issues:

Environment: Attachments:

Description: I'm running asterisk 1.6.2 on ubuntu karmic. I was running a asterisk on ubuntu jaunty, and converted configuration from jaunty to karmic. I have a Wildcard TDM400P REV I (according to DAHDI_hardware) with three modules, fxs, fxo, and fxs. All calls are between DAHDI FXS channel 1 and SIP. The fxs channel 1 is configured in /etc/dahdi/system.conf as fxoks and uses the mg2 echo canceler.

When a phone call is taking place (an incoming call is answered from DAHDI/1), and a second call comes in. The callerid tone is sent, and the person on DAHDI/1 can continue to hear the person on the other end (the SIP call), but they cannot hear the person on the DAHDI/1 line. When the first caller hangs up, the DAHDI/1 line is immediately switched to the new incoming phone call without any 'flash' event taking place. The only information related to DAHDI in the asterisk debug output is the following:

VERBOSE[12345] chan_dahdi.c: -- CPE supports Call Waiting Caller*ID. Sending 'FIRST LAST/9515554444'

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

The following is the uncommented contents of chan_dahdi.conf:
----------------------

[trunkgroups]

[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300
usecallerid=yes
hidecalleridname=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
mailbox=200
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

context=internal
echocancel=yes
mailbox=201
signalling=fxo_ks
callerid="Georgette Foreman"<101>
rxgain=0.0
txgain=0.0
channel => 1

context=incoming
mailbox=200
txgain=0
rxgain=5
signalling=fxs_ks
callerid=
channel => 2

context=internal
echocancel=256
mailbox=200
signalling=fxo_ks
callerid="George Foreman"<100>
rxgain=0.0
txgain=0.0
channel => 3

context=incoming
mailbox=200
signalling=fxs_ks
channel => pseudo

Comments: By: Paul FitzPatrick (fatboyfitz) 2010-04-29 11:24:35

I have the same situation - hoping that using "callwaiting=no" in the chan_dahdi.conf will at least work around the issue. As this is my home phone system, I can test potential patches without major disruption.
By: Alec Davis (alecdavis) 2010-06-08 03:35:56

I am experiencing same here with
Asterisk SVN-branch-1.6.2-r268819M
DAHDI SVN-branch-2.2-r8552

initial call between External SIP to internal DAHDI TDM800P FXS port.
2nd call, (call waiting) from PSTN TDM800P FXO port -> FXS port.

After callwaiting tone, symptoms are the same, no audio FXS -> SIP, but audio SIP -> FXS.
Then SIP end hangs up, and somehow FXS port is then connected to the incoming PSTN call on FXO port (automatically?)

Dialplan below for incoming call<pre>[incoming-sip]
exten => 820,1,Progress()
exten => 820,n,Dial(DAHDI/33&DAHDI/34&DAHDI/35)

[incoming-pstn]
exten => s,1,Dial(DAHDI/33&DAHDI/34&DAHDI/35)</pre>

By: Kirk Wolff (kirkawolff) 2010-08-22 00:20:16

Still seeing bad callwaiting behavior with dahdi 1:2.2.1-0ubuntu2. I've re-installed with Ubuntu server 10.04 (after fixing problem with the netjet driver and the TDM400 PCI VID/PID contention issue), have it on an old intel motherboard since the guys in IRC seem to hate VIA chipsets, and I get similarly broken results but not exact. Now:

1) Phone call with someone (call A)
2) Someone 'beeps in', I get the callwaitingcallerid squawk, I'm able to continue the conversation with call A.
3) I ignore the incoming call and continue conversation with call A, hoping call B goes away and leaves a voicemail.
4) After number of rings expires on call B and it is moved to voicemail, outgoing audio is lost on call A.
5) Call A says "Hello, are you still there". I say "Yes, I'm still here." Call A says "I can't hear you, you must have gone away." I say "Shit, you can't hear me."
6) I hang up, turn off callwaiting in my chan_dahdi.conf, restart asterisk and reinitiate call A.
7) I wait a few days and write this note to you guys hoping something will change soon.
By: Alec Davis (alecdavis) 2011-01-09 22:09:13.000-0600

I've added a releation ship to this, as your initial description sounds exactly the same, which has been fixed in ASTERISK-16803

But I notice your note ~126215 is slightly different.

A later version from 1.6.2 branch may resolve your problem.