Archives : September-2014
Its the first time i try to configure an ISDN card with dahdi, so my experience is very poor (be kind ;))My problem is with dahdi_genconf, when i start it it says:/usr/sbin/dahdi_span_assignments: Missing/sys/bus/dahdi_devices/devices (DAHDI driver unloaded?)Comm..
All,I have one asterisks server and 3 client (im using voip sip client for my handphone). Ive configured sip.conf and extension.conf with 3 user different. And nothing wrong with them, i could make them to make a call too.what i want to ask is, i ..
——=_NextPart_001_0003_01CFDB1B.E4F030A0Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: 7bit, My client has Asterisk based telephony system. He needs to add the intercom feature in his telephones. He has 300 concurrent users w..
All,I am trying to record the call using MixMonitor. exten=>_XXXX,n,MixMonitor(${EXTEN}.wav,b)What i want to do is-when first time a call is made to some number say 1100, a new file(1100.wav) is created. When call is made 2nd or 3rd time, no new f..
hi: when using chan_sip, I can use set SIP_CODEC in dialplan to change the codec of endpoint. this method didnt work with pjsip in asterisk12/13. I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER. according to the description, it seems ..
I am using Asterisk 12 svn, from today, and after a few thousand calls, I run out of ports. This happens eith PJSIOP and regular old SIP. I think it is RTP related. Any idea how can I troblshoot this. It happened teh same with Asterisk 11. On the ot..
Guys, I have recently moved my database servers to a different database cluster that runs on haproxy.Every minute or so I get the following error in asterisk MySQL RealTime: Ping failed (2006).Trying an explicit reconnect The strange thing is if I..
allI am using Asterisk 12.4.0 on debian 7.6 x64I experience some troubles with some specific calls, so I want to dig into this as deep as possibleI run these CLI commands: > sip set history on (answer: SIP History Recording Enabled) > sip set debug p..
Hey everyone!Way back in the murky days of 2010, the Asterisk project released the first version of Asterisk 1.8. This was the first release following a new policy of alternating Long Term Support (LTS) releases and Standard releases. As the first ..
I have 2 Asterisk systems and a unique scenario where I need to play a particular tone on Asterisk1 and identify the same tone on Asterisk2. Following is my call flow, Asterisk1(Plays audiofile1,Wait for 2 Sec,Plays a tone,Plays audiofile2) ->PSTN..