similar to: 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx000001] exten => 101,1,Set(__TRANSFER_CONTEXT=pbx000001) exten => 101,n,Dial(SIP/pozitel/37129238254,45,t) exten => 102,1,Dial(SIP/12345,60) so, when user calls ext

On 12/19/2014 09:42 AM, Rusty Newton wrote: > On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1",

Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:

I've got a confbridge set up which works if dialed locally: -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack -- <DAHDI/1-1> Playing

Hi Everyone, Like always, here are IPs from China that try to hack an Asterisk server. Can someone please explain what is happening or what the hacker is trying to reach: 02/10/2010 11:10 SIP/113.105.152.51-000000fb sip "sip" <sip> s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-000000fe sip "sip" <sip> s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-000000fc

I have two internal analogue extensions off a TDM400P. If the first is busy, I'd like to ring the second. So: [incoming] exten =>s,1,Answer() exten =>s,n,Dial(${mainline},60) exten =>s,n,ExecIf($["${DIALSTATUS}" = "BUSY"]?Dial(${secondline},30)) But it doesn't work because * first tries Call Waiting on the main line. Here I dial out: -- Starting

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==

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. --

On 12/20/2014 03:22 PM, sean darcy wrote: > On 12/19/2014 09:42 AM, Rusty Newton wrote: >> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >>> I've got a confbridge set up which works if dialed locally: >>> >>> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >>> --

On 12/21/2014 04:42 AM, Patrick Beaumont wrote: > Have you enabled DTMF logging and seen the DTMF codes being recognised by > Asterisk? I had a bunch of soft phones that I had to change to using ?sip > info? for the DTMF signalling as the RFC signalling was not always being > recognised. This would cause transfers to appear as if the user had not > dialled any digits. > > >

On 12/21/2014 11:09 AM, sean darcy wrote: > On 12/21/2014 04:42 AM, Patrick Beaumont wrote: >> Have you enabled DTMF logging and seen the DTMF codes being recognised by >> Asterisk? I had a bunch of soft phones that I had to change to using ?sip >> info? for the DTMF signalling as the RFC signalling was not always being >> recognised. This would cause transfers to appear

10.9.0. I'm trying to have a setup where hitting # sends the called party to the confbridge. I've set GOTO_ON_BLINDXFR: CLI> dialplan show globals ......... GOTO_ON_BLINDXFR=tel-incoming^confbridge^1 (Also tried tel-incoming,confbridge,1 and using | ) but it doesn't work: Dial("DAHDI/1-1", "DAHDI/4/xxxyyyzzzz,,tT") in new stack -- Called

On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: > I've got a confbridge set up which works if dialed locally: > > -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack > -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack > -- Executing [266 at internal:3]

http://www.imapwiki.org/ImapTest/ <http://www.imapwiki.org/ImapTest/> I am doing imap testing using imaptest scripts. But, i am unable to append message. APPEND INBOX "15-Jun-2015 05:30:05 -0700" "From:abc at gmail.com" "Subject: test. HI this is msg". This command i am using in my testing script. Command executed properly. But, it is not appending mail

Hi all! :) I see that this is partially covered in the mailing list archive but at the moment I can''t find a straight & working answer. I have an imq device with dynamically attacched classes/qdiscs/filters. There is a hashing filter that maps the last octet of an user''s IP address to a class (and associated qdisc). The "empty" filter looks like this: filter

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

Hi all, I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Thanks and best regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101118/97cba039/attachment.htm

The Asterisk Development Team has announced the release of Asterisk 1.6.2.15. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.15 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.15. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.15 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hi, We had 2 asterisk 1.4 connected together in iax, all was fine. One of them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38 When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But calling from 1.6.2 to 1.4 give a bad audio to calling party (words are cutted, you can't understand the words). On callee party it's still good. We replace