Summary:	ASTERISK-10693: unable to perform attended transfer of incoming call from mISDN
Reporter:	Antonio Gallo (agx)	Labels:
Date Opened:	2007-11-06 08:16:46.000-0600	Date Closed:	2007-11-20 08:21:31.000-0600
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) problem-attended-transfer.log
Description:	After talked with Blitzrage about bug 11085 on #asterisk-bugs we agreed on opening a new bug. Its always reproducible when there is an incoming call from mISDN. If instead i generate the call from a SIP or IAX2 phone it does not happen. The flow involves all GXP2000 phones with firmware 1.1.2.27, also tried 1.1.1.14. The bug involes asterisk 1.4.13 also tried with svn-1.4. mISDN used is 1.1.6 also tried to downgrade to 1.1.5. The flow is like this: 1. SIP/12 answer call from mISDN on gxp's Line1 2. SIP/12 pick Line2 and call SIP/11 (mISDN get MOH) 3. SIP/11 answer SIP/12 and say its ok to talk with misdn people 4. SIP/12 pick line1 (SIP/12 get MOH) and say that the other people is free 5. SIP/12 hit TRANSFER button and press LINE2 and here happen the problem: - 90% of the time SIP/12 get a message onto the screen "TRANSFER CANCELLED" and is unable to repick LINE2 - 10% of the time the call is transferred but is one-way-audio I'll attacch full debug output.
Comments:	By: Antonio Gallo (agx) 2007-11-06 08:27:28.000-0600 sorry uploaded the uncompressed log file :-( from my little understanding of the SIP: the relevant part should be the BYE before "No Such Call" response By: Joshua C. Colp (jcolp) 2007-11-06 10:34:15.000-0600 I see both a SIP issue in here (that I'm working on) plus a jitterbuffer issue. By: Antonio Gallo (agx) 2007-11-16 02:23:06.000-0600 I tried with mISDN 1.1.7 also it didn't worked. If this is a SIP problem is there a previous version that is not affected? I wanna help with resolving this problem. By: Antonio Gallo (agx) 2007-11-16 08:44:08.000-0600 the jitter buffer problem causing the 1-way-audio in 10% of the cases is caused by the adaptive settings. I changed sip.conf and misdn.conf to use fixed buffer: ;JITTER BUFFER CONFIGURATION jbenable = yes jbforce = no ;jbimpl = adaptive <=== reason of the problem jbimpl = fixed ;jbmaxsize=200 jblog = no I still have the SIP transfer issue. On GXP2000 LINE2 get stuck in "HOLD" position and there is no operator to recover it. Line2 is on hold forever until it hangups. By: Joshua C. Colp (jcolp) 2007-11-19 08:46:46.000-0600 Okay, what version of Asterisk did you last test the SIP portion with? If it is not the latest please update and attach the output with sip debug. By: Antonio Gallo (agx) 2007-11-19 08:51:23.000-0600 Latest testing was done with 1.4.13 Retesting with 1.4.14 ASAP By: Antonio Gallo (agx) 2007-11-20 06:33:10.000-0600 working on it, i did a few test and seems ok after i: - upgraded 1.4.13 -> 1.4.14 - switched in sip.conf "directrtpsetup" from Yes to No Give me 1 hour of more testing, i'm just waiting the people with that i usually did those testing to be back By: Antonio Gallo (agx) 2007-11-20 07:28:48.000-0600 triple-checked it with different phones (snom, gxp2000, idefisk, etc.) its solved now :) By: Joshua C. Colp (jcolp) 2007-11-20 08:21:30.000-0600 Fixed in latest version.