Archives : October-2014
Home » 2014 (paged 22)
all,I hope to find a solution with the help of the list, Im trying to get the status of my extensions with ejabberd , the idea is to visualize my users ejabberd incoming calls or missed.Im testing with my operator extension with this code but only ..
A question on channel originating (call files and AMI Originate):How can I change the CALLERID(num) var (because of the E1 provider needs), but having another númber (the original one) stored on the clidCDR field on the database?A channel agnostic solut..
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Someone reported me that from a PBX on which someone gained fraudulent access, he could observe hundreds of calls to the same destination number.For curiositys sake, Im wondering why would this happen (dialing the same number over and over) ?Some spec..
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..