Summary:	ASTERISK-16888: CONNECTEDLINE is not working at all if do not introduce a simple SIP call
Reporter:	Loic Didelot (voipgate)	Labels:
Date Opened:	2010-10-28 06:38:41	Date Closed:	2011-06-07 14:04:54
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Functions/func_callerid
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	I am testing CONNECTEDLINE on asterisk 1.8.0. My phones AASTRA and Polycom support CALLERID updates on outbound calls. What I try to achieve is to rewrite the name on the display of the phone on outbound calls. Here is what I tried and which is not working. I used tshark to confirm that asterisk is not sending any packets with a P-Asserted header. Set("SIP/mixdev1_601-00000091", "CONNECTEDLINE(num)=3133731337") Set("SIP/mixdev1_601-00000091", "CONNECTEDLINE(name)=HUMBALA TEST") Dial("SIP/mixdev1_601-00000091", "SIP/VOIPmixdev1-4/20333000") Here is another example that did not work either. Set("SIP/mixdev1_601-00000091", "CONNECTEDLINE(num)=3133731337") Set("SIP/mixdev1_601-00000091", "CONNECTEDLINE(name)=HUMBALA TEST") Echo() ****** ADDITIONAL INFORMATION ****** While playing around, I found a way to make callerid updates work by modifying the dialplan and by introducing a "dummy" dial line. I am not sure if this helps to find the problem. Set("SIP/mixdev1_601-00000091", "CONNECTEDLINE(num)=3133731337") Set("SIP/mixdev1_601-00000091", "CONNECTEDLINE(name)=HUMBALA TEST") Dial("SIP/mixdev1_601-00000091", "SIP/mixdev1_442,1")    <<< NEW DUMMY LINE Dial("SIP/mixdev1_601-00000091", "SIP/VOIPmixdev1-4/20333000") I have not clue why it works when dialing a registered peer before dialing my SIP trunk. Dialing an nonexistent peer or a peer that is not registered does not help to make CONNECTEDLINE work.
Comments:	By: Richard Mudgett (rmudgett) 2010-10-28 09:47:45 There is documentation on manipulating party ID information in the doc/tex directory.  It is included in the asterisk.pdf file generated by "make pdf". You may not be using the inhibit options correctly.  They are described in the documentation.  See the asterisk.pdf example 19.3.2. By: Loic Didelot (voipgate) 2010-10-28 10:03:16 Come-on...  Did you read and try to understand my example?   Here is my dialplan that I tried now: exten => 55555,1,NOOP(TTTT) exten => 55555,n,Set(CONNECTEDLINE(num,i)=3133731337) exten => 55555,n,Set(CONNECTEDLINE(name,i)=Leet everywhere) exten => 55555,n,Set(CONNECTEDLINE(num-pres,i)=allowed) exten => 55555,n,Set(CONNECTEDLINE(name-pres,i)=allowed) ;;exten => 55555,n,Dial(SIP/mixdev1_333,1) exten => 55555,n,Dial(SIP/VOIPmixdev1-4/20333000) It does not work!!!!! But it works if I activate the commented line. By: Richard Mudgett (rmudgett) 2010-10-28 11:59:10 19.3.2   Straightforward dial through exten  => 1000,1,NoOp ; The  CONNECTEDLINE information is sent when the call is answered. exten  => 1000,n,Set(CONNECTEDLINE(name,i)="Company Name") exten  => 1000,n,Set(CONNECTEDLINE(name-pres,i)=allowed) exten  => 1000,n,Set(CONNECTEDLINE(num,i)=5551212) exten  => 1000,n,Set(CONNECTEDLINE(num-pres)=allowed) ; The  I option prevents overwriting the CONNECTEDLINE information ; set  above when the call is answered. exten  => 1000,n,Dial(SIP/1000,20,I) exten  => 1000,n,Hangup You need the I option in the Dial to prevent the answering side from overwriting your preset information. By: Loic Didelot (voipgate) 2010-10-28 15:47:39 Not every call ends in the DIAL application. Asterisk can do much more like Meetme. By: Richard Mudgett (rmudgett) 2010-10-28 18:43:11 The MeetMe application is not going to overwrite your preset CONNECTEDLINE information when it answers the call because it does not have CONNECTEDLINE information.  In fact, the CONNECTEDLINE function was initially created for use with applications like MeetMe and Playback. The Dial and Queue applications CAN overwrite your preset CONNECTEDLINE information when the called party answers.  The Dial and Queue 'I' option was created to block the CONNECTEDLINE update when the called party answers. See issue ASTERISK-8584 By: Loic Didelot (voipgate) 2010-10-29 00:51:13 I give up. I tell you that it does not work but you do not seem to believe it. Well you can connect to my server or you can close the issue if you still  think its a configuration issue. By: Paul Belanger (pabelanger) 2010-10-29 07:46:35 You'll need to provide something, aside from it does not work.  A debug log (see below) would be a good first step. --- We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt By: Loic Didelot (voipgate) 2010-10-29 09:32:00 I am wiling to pay for the support I need and maybe I got something special in my setup. My server is public so I can give you access and you can try yourself. So I guess it takes about 1-2 hours of debugging. If its a configuration mistake on my side then I pay 100$ per hour. I have provided enough information to reproduce the problem and I provided a strange way to eliminate the problem. So, if its a bug you just put it on your todo an you will fix it otherwise I pay. I do not think that you will accept my offer so your choice to close the ticket. By: Loic Didelot (voipgate) 2010-10-29 10:51:07 It just does not work for the ECHO application. So basically my test was flawed, but thats something that you have to know to do the right tests. Please close this bug report. Not sure if the fact that it does not work for Echo is a bug, but I think its not important.