search for: ast_indicate

Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound & get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered SIP/10.10.2.250-9903 Looking at channel.c, I can see that this means that 'condition' is neither of 'AST_CONTROL_PROGRESS' or 'AST_CONTROL_PROCEEDING'. Presumably it's 'AST...

...system. i am writing a prepaid application to incorporate into the asterisk PBX. however, after searching the database for the user's pin number, the dial application on my dialplan does not work properly. it gives me the error saying... Sep 20 15:47:29 WARNING[1184048960]: channel.c:1404 ast_indicate: Unable to handle indication 3 for 'SIP/3307-8a8f' -- SIP/3308-05d9 answered SIP/3307-8a8f -- Attempting native bridge of SIP/3307-8a8f and SIP/3308-05d9 can anyone tell me what i'm doing wrong? here's a piece of my extensions.conf file that does the pin number varificatio...

.... Is it possible to "transfer" the call in the dialplan or how can I solve this? It doesn't help to use the r option in Dial(SIP/104|30|r). This is the output from asterisk where indication 3 according to the source code is ringing: May 21 11:03:10 WARNING[12188]: channel.c:2049 ast_indicate: Unable to handle indication 3 for 'SIP/XXXXXX-c52d' /urban

Hi, try to setup asterisk as an ISDN2H323-Gateway. The only problem i have after establishing a call is, that Audio works only from IP to ISDN-Phone but not from ISDN to IP-Phone. A known problem ??? Thanks in advance Michael i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus) and oh323-0.6.0 Here are my config's ############## # modem.conf # ##############

...112422428|60|HS(63840)) -- Setting call duration limit to 63840 seconds. -- Called YYYY#XX112422428@XX.103.19.91/XX112422428 -- H323/XX.103.19.91 is making progress passing it to SIP/XX.121.81.26-0815e650 -- H323/XX.103.19.91 is ringing Mar 10 22:26:29 WARNING[20146]: channel.c:1476 ast_indicate: Unable to handle indication 3 for 'SIP/XX.121.81.26-0815e650' -- H323/XX.103.19.91 answered SIP/XX.121.81.26-0815e650 >>>>>. no audio. My H.323 end point is a CISCO g/w. Can anyone help me to understand the meaning of the above warning ??? Any tips greatly appreciated...

...ed 87874586 Jun 8 00:11:31 DEBUG[21733]: rtp.c:472 ast_rtp_read: RTP NAT: Using address 10.17.43.53:8000 Jun 8 00:11:32 DEBUG[21735]: chan_h323.c:1218 progress: Received ALERT/PROGRESS message for self-generated tones -- H323/87874586-11 is ringing Jun 8 00:11:32 DEBUG[21733]: channel.c:1562 ast_indicate: Driver for channel 'SIP/6817-6399' does not support indication 3, emulating it Jun 8 00:11:32 DEBUG[21733]: channel.c:1681 ast_prod: Prodding channel 'SIP/6817-6399' -- H323/87874586-11 is ringing -- H323/87874586-11 answered SIP/6817-6399 Jun 8 00:11:38 DEBUG[21733]: cha...

...om UNKN to G729A Oct 4 00:53:41 DEBUG[1146877376]: rtp.c:438 ast_rtp_read: RTP NAT: Using address 68.2.178.157:16410 Oct 4 00:53:46 DEBUG[1089555136]: chan_zap.c:1179 zt_enable_ec: Enabled echo cancellation on channel 1 -- Zap/1-1 is ringing Oct 4 00:53:46 WARNING[1146877376]: channel.c:1441 ast_indicate: Unable to handle indication 3 for 'SIP/6568543197-86c2' Oct 4 00:54:01 DEBUG[1089555136]: chan_zap.c:1163 zt_enable_ec: Echo cancellation already on -- Zap/1-1 answered SIP/6568543197-86c2 Oct 4 00:54:02 DEBUG[1083546560]: chan_sip.c:823 __sip_ack: Stopping retransmission on 'e73...

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --

...w stack -- Called g1/6152437 Aug 2 11:04:17 DEBUG[557083]: rtp.c:1166 ast_rtp_write: Ooh, format changed from UNKN to ULAW Aug 2 11:04:22 DEBUG[262160]: chan_zap.c:1186 zt_enable_ec: Enabled echo cancellation on channel 1 -- Zap/1-1 is ringing Aug 2 11:04:22 DEBUG[557083]: channel.c:1436 ast_indicate: Driver for channel 'SIP/4001-40ee' does not support indication 3, emulating it Aug 2 11:04:22 DEBUG[557083]: channel.c:1551 ast_prod: Prodding channel 'SIP/4001-40ee' Aug 2 11:04:22 DEBUG[557083]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals Aug 2 11:0...

...0f200, delta=-1) at astobj2.c:229 #9 0x0809c1e9 in ast_closestream (f=0xb1f0f200) at file.c:902 #10 0x00b03422 in local_ast_moh_stop (chan=0xb2150fd0) at res_musiconhold.c:1058 #11 0x00a6e510 in sip_indicate (ast=0xb2150fd0, condition=17, data=0x0, datalen=0) at chan_sip.c:4049 #12 0x08081512 in ast_indicate_data (chan=0xb2150fd0, _condition=17, data=0x0, datalen=0) at channel.c:2530 #13 0x08081728 in ast_indicate (chan=0xb2150fd0, condition=17) at channel.c:2475 #14 0x0097fd33 in agent_new (p=0x950fef0, state=0) at chan_agent.c:1139 #15 0x009837cf in agent_request (type=0xb55fc9b4 "Agent",...

...Line 521 (__sip_ack): Stopping retransmission on '7a655bae-29f17562-197726d5@62.212.12.21' of Response 26024: Not Found Jun 5 01:55:37 DEBUG[1236360496]: File pbx.c, Line 1116 (pbx_extension_helper): Launching 'Ringing' Jun 5 01:55:37 DEBUG[1236360496]: File channel.c, Line 1163 (ast_indicate): Driver for channel 'SIP/5010-d3c4' does not support indication 3, emulating it Jun 5 01:55:37 DEBUG[1236360496]: File channel.c, Line 747 (ast_activate_generator): ast_activate_generator Jun 5 01:55:37 DEBUG[1236360496]: File channel.c, Line 1381 (ast_set_write_format): Set channel SIP/...

Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know

...at ALAW Aug 26 15:54:17 DEBUG[-1260983376]: rtp.c:1156 ast_rtp_write: Ooh, format changed from UNKN to ALAW Aug 26 15:54:17 DEBUG[-1248367696]: chan_zap.c:1179 zt_enable_ec: No echocancellation requested -- Zap/17-1 is ringing Urgent handler Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1395 ast_indicate: Driver for channel 'SIP/2000-e12c' does not support indication 3, emulating it Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1510 ast_prod: Prodding channel 'SIP/2000-e12c' Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set channel SIP/2000-e12c to writ...