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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. --

Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the "transferer". Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like

...ndesired). We like to keep the channel up and do something else in the dialplan. As I'm not too familiar with the asterisk internals I have problems to interprete the sources, but I think it might be in the method: static int changrab_exec in Line 131 which is res = ast_bridge_call(chan, newchan, &config); ast_hangup(newchan); I have no clue if ast_bridge_call blocks until some statechange happens and ast_hangup hangs up which would lead to the undesired behavior!? Maybe someone can help me understanding some basics about these asterisk functions!? -- Knud A. M?ller

I released app_changrab.c lastnight really late... It includes a way to hijack a channel and originate calls from the CLI. /b --- Keep Your Friends Close, But Your Enemies Even Closer...

...chan_sip.c:11888 sip_dump_history: 008. Invite New call: 842ada2a-77b3dff at 192.168.xx.xx [Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 009. AuthOK Auth challenge succesful for 293 [Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 010. NewChan Channel SIP/293-00000002 - from 842ada2a-77b3dff at 192.168.xx.xx [Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 011. TxResp SIP/2.0 / 102 INVITE - 100 Trying [Mar 26 14:36:57] DEBUG[12001]: chan_sip.c:11888 sip_dump_history: 012. TxResp SIP/2.0 / 1...