| Summary: | ASTERISK-03020: Missing CLID on H.323? | ||
| Reporter: | tdriscoll (tdriscoll) | Labels: | |
| Date Opened: | 2004-12-16 23:17:21.000-0600 | Date Closed: | 2011-06-07 14:05:23 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Is caller ID acting up for anyone? After last nights update I can't receive Caller ID from my H323 Gateway. Under trace it shows up but is not delivered to my sip phones ****** ADDITIONAL INFORMATION ****** *CLI> == New H.323 Connection created. --Received SETUP message -- Setting up Call -- Call token: [ip$192.168.1.99:1617/26562] -- Calling party name: [BCM35 Office] -- Calling party number: [9722981193] -- Called party name: [5001] -- Called party number: [5001] =-= In OnAnswerCall for call 26562 - Progress Indicator: 0 - Inserting PI of 0 into ALERTING message -- Executing Dial("H323/ip$192.168.1.99:1617/26562", "SIP/5001|20") in new stack -- Called 5001 -- Started logical channel: sending G.711-uLaw-64k -- channelsOpen = 1 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 192.168.1.55 -- remotePort: 51000 -- ExternalIpAddress: 192.168.1.2 -- ExternalPort: 28044 -- Started logical channel: receiving G.711-uLaw-64k -- channelsOpen = 2 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 192.168.1.55 -- remotePort: 51000 -- ExternalIpAddress: 192.168.1.2 -- ExternalPort: 28044 -- SIP/5001-c377 is ringing Sending alerting -- SIP/5001-c377 answered H323/ip$192.168.1.99:1617/26562 Answering call ip$192.168.1.99:1617/26562 -- Transmitting RFC2833 on payload 101 channelsOpen = 1 ExternalRTPChannel Destroyed =-= In OnConnectionEstablished for call 26562 -- Connection Established with "BCM35 Office (E164:9722981193) [192.168.1.99]" channelsOpen = 0 ExternalRTPChannel Destroyed -- Transmitting RFC2833 on payload 101 -- Started logical channel: sending G.711-uLaw-64k -- channelsOpen = 1 =-= In OnConnectionEstablished for call 26562 -- Connection Established with "BCM35 Office (E164:9722981193) [192.168.1.99]" -- Started logical channel: receiving G.711-uLaw-64k -- channelsOpen = 2 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 192.168.1.55 -- remotePort: 51000 -- ExternalIpAddress: 192.168.1.2 -- ExternalPort: 28044 MyH323_ExternalRTPChannel::OnReceivedAckPDU -- remoteIpAddress: 192.168.1.55 -- remotePort: 51000 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 192.168.1.55 -- remotePort: 51000 -- ExternalIpAddress: 192.168.1.2 -- ExternalPort: 28044 -- ClearCall: Request to clear call with token ip$192.168.1.99:1617/26562, cause 3 -- Sending RELEASE COMPLETE -- Received RELEASE COMPLETE message... -- ClearCall: Request to clear call with token ip$192.168.1.99:1617/26562, cause 3 -- ClearCall: Request to clear call with token ip$192.168.1.99:1617/26562, cause 7 channelsOpen = 1 channelsOpen = 0 ExternalRTPChannel Destroyed ExternalRTPChannel Destroyed -- BCM35 Office (E164:9722981193) [192.168.1.99] has cleared the call == H.323 Connection deleted. == Spawn extension (sip, 5001, 1) exited non-zero on 'H323/ip$192.168.1.99:1617/26562' -- Saved useragent "X-Lite release 1103m" for peer 5001 | ||
| Comments: | By: Brian West (bkw918) 2004-12-17 00:01:43.000-0600 So help please provide more info such as version and such... By: tdriscoll (tdriscoll) 2004-12-17 00:14:25.000-0600 Ooops Sorry about that. Asterisk CVS-HEAD-12/16/04-23:54:17 H323 gateway is a Nortel BCM3.5 using I2004 IP phones. Asterisk uses snom 200's and xlite. When a snom or a xlite phone calls from asterisk to the BCM the proper caller id is displayed but when a bcm phone calls the sip phone on asterisk the caller id displays asterisk not the calling set. In debug the information is present but not sent to the receiving devices. By: Mark Spencer (markster) 2004-12-17 03:31:47.000-0600 Please try to use a complete and fully descriptive title in the future. By: tdriscoll (tdriscoll) 2004-12-17 08:43:59.000-0600 my apologies - I will do better in the future. By: Brian West (bkw918) 2004-12-17 10:10:21.000-0600 do this. in [general] in sip.conf do context=NO-SUCH-CONTEXT reload Call again. I suspect one of your phones isn't doing the right thing (tm) Intended results you might see is the phone will not work. bkw By: tdriscoll (tdriscoll) 2004-12-17 21:01:09.000-0600 That is a good thought. But it doesn't appear to be the case. I also have a stable 1.0. Jeremy convinced me of doing that so that I always have a reference point. I used the same config files and phones. CLID works fine on it. How do I get more information to you all. Debug level 4 ? any ideas on capturng it to a file. I am new to debugging. Question: Since h323 has been completely rewritten do I need to treat incoming calls differently than before. In prior versions I didn't have to do anything but accept the call. I don't have h323 extensions. I just use h323 to connect to other gateways such as nortel's bcm or audiocodes media paks. May I barrow someones h323.conf file please x out the IP's edited on: 12-17-04 21:09 edited on: 12-17-04 21:11 By: jerjer (jerjer) 2004-12-17 23:11:26.000-0600 Fixed in cvs -head. By: Digium Subversion (svnbot) 2008-01-15 15:16:58.000-0600 Repository: asterisk Revision: 4473 U trunk/channels/chan_h323.c ------------------------------------------------------------------------ r4473 | jeremy | 2008-01-15 15:16:58 -0600 (Tue, 15 Jan 2008) | 2 lines Fix incoming caller*id. Bug ASTERISK-3020 ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=4473 | ||