Gunnar Þ. Gestsson
2004-Nov-04 03:31 UTC
[Asterisk-Users] Voicemail, Cisco and H.323 problems
Hello. I have a Cisco IP Phone 7940 connected to Cisco CallManager. I have a Cisco 2600 Router running H.323 Gatekeeper I have Asterisk server running OpenH323 that registers to this Gatekeeper. I have a Cisco CallManager that registers to this Gatekeeper. The problems I face are: 1. Calling from Cisco IP Phone to Asterisk. -- Executing Dial("H323/ip$10.169.208.3:4791/194", "IAX2/4555564| 20| r") in new stack -- Called 4555564 -- Call accepted by 82.221.53.53 (format ULAW) -- Format for call is ULAW -- IAX2/4555564/1 is ringing -- Nobody picked up in 20000 ms -- Hungup 'IAX2/4555564/1' -- Executing VoiceMail("H323/ip$10.169.208.3:4791/194", "u4555564") in new stack If I answer this call with IAX Phone on my Windows Desktop then audio is not transcoded in Asterisk between codecs. I can see that both phones are sending Asterisk udp packets but no packets are sent from Asterisk to neither phones. Is there not supposed to be a transcoder in Asterisk to transcode GSM <-> uLaw ? If I don't pick up the call the VoiceMail answers. I can't hear anything in my Cisco Phone when voicemail answers. If I dial VoiceMailMain I will not hear anything in my Cisco IP Phone either. 2. Calling from Asterisk to Cisco IP Phone. exten => _45570XX,1,Dial(H323/${EXTEN}) ; With Gatekeeper this should be the Dial Command, right ? exten => _45570XX,2,Hangup -- Executing Dial("IAX2/4555564@4555564/2", "H323/4557040") in new stack Nov 4 09:40:28 NOTICE[360469]: chan_h323.c:464 oh323_call: h323_make_call failed(H323/4557040) -- Couldn't call 4557040 == Everyone is busy/congested at this time -- Executing Hangup("IAX2/4555564@4555564/2", "") in new stack == Spawn extension (default, 4557040, 2) exited non-zero on 'IAX2/4555564@4555564/2' -- Hungup 'IAX2/4555564@4555564/2' 3. Asterisk does not answer all h.323 calls. tcpdump of CallManager (10.169.208.3) calling Asterisk (10.169.208.11) but Asterisk is not answering. no entries found ? Asterisk log or console to explain why the call is not answered. 09:50:03.501931 10.169.208.11.h323hostcall > 10.169.208.3.1301: P 179:225(46) ack 232 win 6432 (DF) 09:50:03.502074 10.169.208.3.1301 > 10.169.208.11.h323hostcall: R 1926726275:1926726275(0) win 0 [tos 0x68] 09:50:03.502863 10.169.208.11.h323hostcall > 10.169.208.3.1301: R 225:225(0) ack 232 win 6432 (DF) 09:50:04.275719 arp who-has 10.169.208.3 tell 10.169.212.64 09:50:04.298929 10.169.208.3.xtel > 10.169.208.11.h323hostcall: S 1930294417:1930294417(0) win 65535 <mss 1460,nop,nop,sackOK> [tos 0x68] 09:50:04.298947 10.169.208.11.h323hostcall > 10.169.208.3.xtel: S 2316371676:2316371676(0) ack 1930294418 win 5840 <mss 1460,nop,nop,sackOK> (DF) 09:50:04.299044 10.169.208.3.xtel > 10.169.208.11.h323hostcall: . ack 1 win 65535 [tos 0x68] 09:50:04.301532 10.169.208.3.xtel > 10.169.208.11.h323hostcall: P 1:182(181) ack 1 win 65535 [tos 0x68] 09:50:04.301559 10.169.208.11.h323hostcall > 10.169.208.3.xtel: . ack 182 win 6432 (DF) 09:50:04.304664 10.169.208.11.h323hostcall > 10.169.208.3.xtel: P 1:179(178) ack 182 win 6432 (DF) 09:50:04.493788 10.169.208.3.xtel > 10.169.208.11.h323hostcall: . ack 179 win 65357 [tos 0x68] 09:50:06.986005 arp who-has 10.169.208.3 tell 10.169.212.39 09:50:08.048699 arp who-has 10.169.208.3 tell 10.169.212.34 09:50:11.604300 arp who-has 10.169.208.3 tell 10.169.212.69 09:50:16.342221 10.169.208.11.h323hostcall > 10.169.208.3.xtel: P 179:225(46) ack 182 win 6432 (DF) 09:50:16.343182 10.169.208.11.h323hostcall > 10.169.208.3.xtel: F 225:225(0) ack 182 win 6432 (DF) 09:50:16.343317 10.169.208.3.xtel > 10.169.208.11.h323hostcall: . ack 226 win 65311 [tos 0x68] 09:50:16.347658 10.169.208.3.xtel > 10.169.208.11.h323hostcall: F 182:182(0) ack 226 win 65311 [tos 0x68] 09:50:16.347670 10.169.208.11.h323hostcall > 10.169.208.3.xtel: . ack 183 win 6432 (DF) [tos 0x68] 09:50:20.875796 arp who-has 10.169.212.98 tell 10.169.208.3 4. Changing Asterisk configuration. If I change Asterisk configuration and execute the reload command no H.323 communication will work. I am forced to stop Asterisk service, clear the endpoint from the Gatekeeper and restart Asterisk. Regards, Gunnar Gestsson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041104/bc320081/attachment.htm