Archives : February-2017
Home » 2017 (paged 38)
Sharing to the groupDigium Announced the Call for Speakers is Now Open for AstriCon 2017The 14th annual Asterisk user conference and expo to be held Oct. 3-5 atthe Omni Orlando Resort at ChampionsGate in Orlando, Fla. – Digium blog post: http://blogs.digium.com/2017/02/02/astricon-2017-october-orlan..
Asterisk Users.I have an issue with receiving fax on my Asterisk/SIP channel. I keep getting timeout under T.38 negotiation – see http://pastebin.com/6eCe26YMAny help would be greatly appreciated..
Ive got a 13.13.1 system using PJSIP stack on debian Jessie. It runs from 50 to 100 simultaneous calls (so 100 to 200PJSIP channels)all day long. From time to time, roughly meaning once a month, it segfaultswith lines(from dmesg -T output) like this:asterisk[116..
Asterisk Users,Hope you all doing fine!I am working with a quite complex dialplan, and Ive come to some situations where it makes some nasty use of pre-bridge handlers. The pre-bridge handlers wiki (https://wiki.asterisk.org/wiki/display/AST/Pre-Bridge+Handle..
there;2 linux boxes and Windows all report an error and the archive is not extractable.Wget reports the size as follows:2017-02-14 08:36:21 (7.29 MB/s) – ‘asterisk-14-current.tar.gz’ saved[40653605/40653605]It starts un-tarring but then….asterisk-14.3.0/bridges/bridge_native_rt..
Everyone,I am dealing with a problem for now and its really annoying.I want to hangup calls from AGI but it seems that my AGI is not rejecting the calls properly.{$agi->verbose(number-not-in-service);$agi->exec(Congestion,1);$agi->hangup();exit;}w..
I have one: YGW30 1FXS,1FXO SIP ATA unit it was made by company Yuxin I think they are no longer in business.I forgot the default user name / password for log-in.Does anybody know what was the default login or have..
all, I have a strange issue, with a some kind complicate architecture… A router of our internet provider is in front of another bintec rs353j router, at which my freepbx installation is located. However, NAT etc. seems to work fine. BUT: Someth..
all,can someone help? I have CentOS 6.8 trying to install asterisk 14.3.0-rc2on it with options as stated below -./configure –with-crypto –with-ssl –with-srtp=/usr/local/lib–with-jansson=/ –with-pjproject-bundledwhen I tried to run make menusele..
ALLgotsmall questioni use call-limit=1 on peersbutcall limit is not workingifuseris notregistered on PBX and making callsso the main question is — how to Disallow CALLS without register..