| Summary: | ASTERISK-08803: Error at call from h323 to SIP | ||
| Reporter: | dmoreno (dmoreno) | Labels: | |
| Date Opened: | 2007-02-14 06:19:44.000-0600 | Date Closed: | 2007-02-14 09:13:54.000-0600 |
| Priority: | Blocker | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_h323 |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Platform: Ubuntu Dapper (Linux 2.6.15-27-686 #1 SMP PREEMPT i686 GNU/Linux) Asterisk 1.4.0 release Asterisk configuration as sip gateway with gnugk as h323 gatekeeper (all in the same host -> 10.10.10.5). There is no nat in this scenario. Some config files: * h323.conf (complete file): [general] port = 1730 bindaddr = 0.0.0.0 allow=all gatekeeper = 10.10.10.5 AllowGKRouted = yes AcceptAnonymous = yes context=from-h323 [astgw] type=h323 context=from-h323 prefix=11,22,33 * sip.conf (only h323 references): [from-h323] exten => _.,1,NoOp(Incoming Call from h323 protocol ${CALLERID} for ${EXTEN}) exten => 11,1,NoOp(Call to lolo) exten => 11,n,Dial(SIP/lolo) exten => 22,1,NoOp(Call to pera) exten => 22,n,Dial(SIP/pera) exten => 33,1,NoOp(Call to diego) exten => 33,n,Dial(SIP/diego) * gnugk.ini (default config file with only one addon): [RasSrv::PermanentEndpoints] 10.10.10.5=ASTERISK;diego Now, I call from a h323 sotfphone (Polycom PVX 8.0.0;10.10.10.6), dialing 33 extension, to a SIP softphone (Ekiga-2.0.1 registered as sip user diego;). Then, the SIP phone rings and when I pick the phone the call is finished with this error: [Feb 12 17:34:16] ERROR[6311]: chan_h323.c:1848 external_rtp_create: Unable to find call ip$10.10.10.6:3236/13438(13438) ERROR: on_external_rtp_create failure ****** ADDITIONAL INFORMATION ****** I attach a log file with this options: core set verbose 3 core set debug 2 h323 set debug h323 set trace 5 | ||
| Comments: | |||