search for: ast_indicate_data

Hi guys, When I try to get ring tones when dialing out with the command Dial(SIP/sipout/${PHONE},15,r), I get the error message indicated in the subject. I've checked my indications.conf file using the sample file provided with asterisk 1.4.10 (the version I'm using) and it's not better. Any idea ? Regards. -- Cyril SCETBON

...analog line the call is ok. Everyone, has been that problem? I change asterisk version 1.4.21 to 1.4.18 but the same problem. I saw the cli [Aug 12 16:15:40] WARNING[2997]: chan_sip.c:3927 sip_indicate: Don't know how to indicate condition 9 [Aug 12 16:15:40] WARNING[2997]: channel.c:2369 ast_indicate_data: Unable to handle indication 9 for 'SIP/4001-0a16f5c0' Anyone can help me.. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090813/2b47e1ff/attachment.htm

...ated callcenter, but agents registers to another SIP server. When agent tries transfer a client to another operator , pressing flash, I get this: [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know how to indicate condition 9 [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data: Unable to handle indication 9 for 'SIP/xxx.xxx.xx-082b9c80' -- Started music on hold, class 'default', on SIP/sip.call.lt-082b9c80 Is it possible to avoid this? I don't want that in this situation (after pressing Flash), asterisk starts Musing On Hold. Thanks for help --...

...After 60 seconds of ringing back, it's supposed to go to [park-dial] t extension as far as I can tell, which it actually does seem to do. However, before the t extension kicks in, the line is dropped with the following error message on the CLI: [Oct 3 08:45:31] WARNING[12621]: channel.c:2616 ast_indicate_data: Unable to handle indication 3 for 'SIP/727-095c0348' [Oct 3 08:46:31] WARNING[11487]: chan_sip.c:12037 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '294bf6652379a770665524cc50a8cfab at xxx.xxx.xxx.xxx. Giving up. -- SIP/717-09570200 is circuit...

...23/(null)-8c76' > >? ???-- Executing [9999 at h323:1] > Dial("OOH323/(null)-3074", > > "Zap/8/604xxxxxxx") in new stack > >? ???-- Called 8/604xxxxxxx > >? ???-- Zap/8-1 is ringing > > [2008-07-02 15:48:55] WARNING[21544]: channel.c:2390 > ast_indicate_data: > > Unable to handle indication 3 for > 'OOH323/(null)-3074' > >? ???-- Zap/8-1 is ringing > >? ???-- Zap/8-1 answered > OOH323/(null)-3074 > > [2008-07-02 15:49:08] WARNING[21544]: > chan_ooh323.c:1053 > > ooh323_indicate: Don't know how to indic...

Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

Hi List. I have a small problem in using the transfer key transfer of IP Phone in Asterisk 1.6, I think I spend some detail in the configuration but can not find. What happens is, when I do a transfer using the Transfer button, the phone, does not play the music on hold, which is waiting on the phone is silent, and I have the same settings on a 1.4 server, and the music plays correctly when

...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", form...

...;, "belllord|SIP/101&IAX2/alanb/201|tolc") in new stack -- Executing [s at macro-belllord:1] Dial("SIP/101-081d1050", "SIP/101&IAX2/alanb/201|10|tr") in new stack -- Called 101 -- Called alanb/201 [Oct 17 16:09:47] WARNING[2836]: channel.c:2634 ast_indicate_data: Unable to handle indication 3 for 'SIP/101-081d1050' -- SIP/101-081d4fc0 is ringing -- Call accepted by 80.XXX.XX.XX (format alaw) -- Format for call is alaw -- IAX2/alanb-3 answered SIP/101-081d1050 [Oct 17 16:09:47] NOTICE[2836]: cdr.c:434 ast_cdr_free: CDR on channe...

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx