search for: builtin_atxfer

...isk says the dialed number is incorrect or non existing. We tried with blind transfers and the same problem. This is the Asterisk CLI log when making a call transfer: -- <DAHDI/19-1> Playing 'pbx-transfer.gsm' (language 'es') [Oct 19 09:00:21] WARNING[18521]: features.c:2319 builtin_atxfer: No digits dialed for atxfer. -- <DAHDI/19-1> Playing 'pbx-invalid.gsm' (language 'es') -- <DAHDI/19-1> Playing 'pbx-transfer.gsm' (language 'es') [Oct 19 09:00:50] WARNING[18521]: features.c:2319 builtin_atxfer: No digits dialed for atxfer. -...

...fer with 3 SIP phones. I noticed about 10% of my transfers make the call drop and I get this on my log: Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback: Failed to write frame -- Playing 'beep' (language 'it') Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer: Failed to play transfer sound! Moreover, every time I try to transfer from called phone to a third phone I get this message: -- SIP/5-082a9f78 answered Local/12 at inbound_sip-f8de,2 Jul 5 13:02:40 NOTICE[24701]: res_features.c:1171 ast_feature_request_and_dial: Don't know what to do...

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: luned? 30 maggio 2005 18.40 A:

...ing -- SIP/2391-d264 answered SIP/2339-4e1d -- Attempting native bridge of SIP/2339-4e1d and SIP/2391-d264 -- Started music on hold, class 'default', on SIP/2339-4e1d -- Playing 'pbx-transfer' (language 'it') Jun 7 15:11:41 WARNING[13701]: res_features.c:881 builtin_atxfer: Did not read data. -- Playing 'beeperr' (language 'it') == Spawn extension (intern, 2391, 1) exited non-zero on 'SIP/2339-4e1d' -- Stopped music on hold on SIP/2339-4e1d Jun 7 15:11:48 WARNING[13701]: channel.c:902 ast_hangup: Hard hangup called by thread 327699...

Hi, We are having a problem that is preventing users from using *2 to manage an attended transfer. After dialling *2, asterisk places the call on hold, but you can only dial one digit before it times out, and the cli says:- [2011-06-29 18:33:55] WARNING[29877]: res_features.c:938 builtin_atxfer: Did not read data. There is already an issue in JIRA: https://issues.asterisk.org/jira/browse/ASTERISK-16927 And loads of people on forums with this problem, but no one has an answer. Have anyone made a workaround for this? Thanks Dan -------------- next part -------------- An HTML attachment...

The procedure that will do asterisk is very nice ;) but whe it was available ? Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why. Here my sip.conf for the phone, can u say me if there is somethingh wrong ? [2391] type=friend username=2391 secret=2391 language=it host=dynamic context=intern dtmfmode=rfc2833

Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to