search for: sip_ind

...and that if they hear the call originates, the lines are E1 in alcatel pbx. When a asteris user call to 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://li...

Hello, I've a problem. I've asterisk 1.6.0.5 version. And I've created 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...

...Executing [s at Capture-Fax:2] ReceiveFAX("SIP/173-b53023e8", "/var/spool/asterisk/fax/20090921_1806.tif") in new stack -- Channel 'SIP/173-b53023e8' receiving fax '/var/spool/asterisk/fax/20090921_1806.tif' [Sep 21 18:06:37] WARNING[5149]: chan_sip.c:5561 sip_indicate: Don't know how to indicate condition 19 -- Channel 'SIP/173-b53023e8' fax session '7' started Is there some list of these conditions? Should I care? sean

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

...eb2a in filestream_destructor (arg=0xb1f0f200) at file.c:340 #8 0x0806e412 in ao2_ref (user_data=0xb1f0f200, 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=...