Summary: | ASTERISK-07396: Wrong Caller ID after Attended Transfer | ||
Reporter: | Raphael Vallazza (endian) | Labels: | |
Date Opened: | 2006-07-25 10:47:34 | Date Closed: | 2006-07-26 10:19:40 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/Transfers |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) astersik-atxfer.txt ( 1) Attended_Xfer1.txt | |
Description: | We have the following problem with Snom phones (but i don't think it's Snom related): 1) Phone B recieves a call from Phone A 2) Phone B puts Phone A on hold and calls Phone C 3) Phone C answers and sees the Caller ID of Phone B 4) Phone B now just does a hangup to pass Phone A -> Phone C 5) Phone C still sees Phone B as Caller ID (instead of Phone A) At point 5 it should show the Caller ID of Phone A instead of Phone B. I don't think this is a correct behaviour so i contacted the Snom Support and they stated that it's an Asterisk related problem because it works with SER. They also sent me a SIP Traces of SER, i've attached them. Maybe i'm doing something wrong... | ||
Comments: | By: Serge Vecher (serge-v) 2006-07-25 10:50:17 As per bug guidelines, you need to attach a SIP debug trace illustrating the problem. Please do the following: 1) Prepare test environment (reduce the ammount of unrelated traffic on the server); 2) Make sure your logger.conf has the following line: console => notice,warning,error,debug 3) restart Asterik. 4) Enable SIP transaction logging with the following CLI commands: set debug 4 set verbose 4 sip debug 5) Save complete console log to file and _attach_ said file to the bug. By: Joshua C. Colp (jcolp) 2006-07-25 11:15:36 This is just the way things work. Phone B is the phone that called Phone C, not Phone A. Now unless one could update callerid after a call has been established... then you could change it but we don't support doing that. By: Raphael Vallazza (endian) 2006-07-25 11:35:38 I understand that that's the way things work, to be honest i don't have a problem with it, but our users do... :) Everyone expects to see the original caller number, because all the time they hear from the caller "why are you asking my number, you should see it on your display" :-D Well, i've attached the asterisk log. It's a call from 10 -> 11 then then 11 -> 13 and after hangup 10 -> 13. The Snom support guy said that this problem is related to SIP REFER messages not being forwarded... Sorry that i didn't attach the log earlier. By: Joshua C. Colp (jcolp) 2006-07-25 12:59:52 Asterisk is not a SIP proxy, so it doesn't "forward" packets - thus why this entire situation is occuring. Each call is kept separate from the other, so when you do an attended transfer and finish it - the other side doesn't know it occured. Sorry, but that's the way it works and will continue to work... Asterisk has to remain a back to back user agent in order to keep being protocol independent. By: Serge Vecher (serge-v) 2006-07-25 13:09:12 interesting line from debug ... Jul 24 11:54:05 DEBUG[3885]: chan_sip.c:1030 parse_sip_options: Found no match for SIP option: callerid (Please file bug report!) By: Joshua C. Colp (jcolp) 2006-07-25 13:16:29 'tis a SIP option we don't support, a cool thing would be trying to track down whether it has an RFC or not... and what the option is for exactly. By: Raphael Vallazza (endian) 2006-07-25 15:05:55 Hmmm, ok, i understand :-/ The problem is that many users complain that and i also understand theri disappointment... is there any possible hack, workaround, whatever that could solve the problem? By: Joshua C. Colp (jcolp) 2006-07-25 15:25:08 I can't think of any off the top of my head, sorry. By: Serge Vecher (serge-v) 2006-07-25 15:29:53 you may want to consider posting a bounty for this support on wiki. By: Joshua C. Colp (jcolp) 2006-07-26 10:19:39 And this bug comes to a close since it's not really a bug... if you do end up finding something endian, post back so that we can potentially help others who want this same thing. Thanks! |