search for: moshiach

Displaying 8 results from an estimated 8 matches for "moshiach".

2015 Jun 17
1
Channels stuck on CONFBRIDGE_INFO
...ing asterisk) - the problem disappeared. So my conclusion is that this is probably some kind of a deadlock with CONFBRIDGE_INFO function. I've tried to look for bug reports, but didn't find anything similar. Is this a known issue? We're using Asterisk 11.9. Best regards -- ???? NOW! Moshiach is coming very soon, prepare yourself! ??? ?????? ?????? ?????? ??? ????? ????? ???! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150617/350948a7/attachment.html>
2013 Aug 11
1
SIP trunk and congestion handling
..., but actually the java application gets OriginateResponseEvent with failure reason code 8. This behavior is hard to reproduce. I was trying with my own phone number and then i get the expected reason code 3, but i constantly get this situation running our customer's campaigns. -- ???? NOW! Moshiach is coming very soon, prepare yourself! ??? ?????? ?????? ?????? ??? ????? ????? ???! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130811/6035897b/attachment.htm>
2015 Mar 02
4
Problems with the voice quality under load
...ical server into several VMs and they claim that this will improve the total capacity. In my own experience, this did not work very well and seems like the visualization actually made the quality worse. Do you have any advice for me (other than purchasing more servers ;-) ? Thanks! -- ???? NOW! Moshiach is coming very soon, prepare yourself! ??? ?????? ?????? ?????? ??? ????? ????? ???! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150302/2cba9b4d/attachment.html>
2013 Aug 22
2
How to get the original SIP result code
...nnel. Is there any way that my AMI application can get the original SIP response of the failed Originate action? I'm using Asterisk 1.8.22 and slightly tweaked asterisk-java ( https://blogs.reucon.com/asterisk-java/) 1.0.0. -- ????? ?????? ???? ???? ???? ?????? ??????? ?????????! ???? NOW! Moshiach is coming very soon, prepare yourself! ??? ?????? ?????? ?????? ??? ????? ????? ???! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130822/9a6c1020/attachment.htm>
2013 Jun 03
1
DAHDI 2.6 and OPENVOX cards
...led and configured it. It recognizes my card, but after loading dahdi and the config, the card is not functional - it shows green lights even when no E1 cable connected, and dahdi_test gives 0.0% all the time so looks like it doesn't generate interrupts. Thanks for your help :-) -- ???? NOW! Moshiach is coming very soon, prepare yourself! ??? ?????? ?????? ?????? ??? ????? ????? ???! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130603/f67cefdc/attachment.htm>
2014 Jan 01
1
Get data from the SDPof SIP INVITE message
...oks like it was abandoned or forgotten. The patch is against older version of chan_sip and i don't know how to apply it against the current version. I'm not enough familiar with chan_sip internals. Is there any way to do this with the current version of Asterisk? Thanks in advance! -- Moshiach is coming very soon, prepare yourself! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140101/602b4542/attachment.html>
2013 Jun 11
2
A problem with IAX2
...switch, there is no problems with the network. No packet loss. There's already bug report present with very similar issue, but it is "suspended" and, like stated there, the problem is very hard to reproduce. See: https://issues.asterisk.org/jira/browse/ASTERISK-21762 -- ???? NOW! Moshiach is coming very soon, prepare yourself! ??? ?????? ?????? ?????? ??? ????? ????? ???! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130611/33ea9b4c/attachment.htm>
2013 Jun 19
3
Handoff dial control to dialplan after AMI Originate
Hello, I'd like to use the AMI interface to originate a call to a context in a dialplan, and handoff the dial control to the context. Whenever I execute the below action, the recipient does ring, but when I answer it dials the recipient again. I believe this is because once answered the system is going to execute the Context/Exten/Prio in the Originate action? Action: Originate Channel: