listi have an issue with my dahdi_channels.confin span 1 when i use it like below i can do my outband calls without issue; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)group=0,11context=from-pstn switchtype = euroisdn signalling = pri_cpe chan..
all, Is there any way of originating calls in future without using call files?We have 2 servers (1 active at a time). If we use call files with modification date in future, on the 1st server and it dies and, we activate the second server but we l..
Using Ubuntu Server 12.04 and Asterisk 11.2.1.Im getting the following error when trying to start asterisk:(Syslog) kernel: [ 1032.713864] asterisk trap invalid opcode ip:7fc272923076 sp:7fff928cf1b0 error:0 in codec_ilbc.so[7fc272921000+e000..
all im using ooh323.so module for my h323 connections and it works fine. i just have problem with loading and unloading module. you know, ooh323module doesnt support reload command. it means, if ooh323 module is loaded and i reconfigure my h323 chann..
I keep getting the following message whenever an AMI call is made:asteriskjava.manager.internal.EventBuilderImpl.buildEvent(EventBuilderIm pl.java:296) No event class registered for event type localbridge, Tried adding an event listener. Anyone k..
Ive got a Digium Wildcard TDM410P with one POTS line and three extensions. One of the extensions is connected to a fax modem. This kind of works, but theres a gotcha. If I set faxdetect=incoming in chan_dahdi.conf, then incoming faxes do get routed..
Hello!Just readhttp://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINEtried on dahdi, it works, i.e. if I call Asterisk user from my pbx connected phone I see what I set in Set(CONNECTEDLINE(name)But if I call the same user over h323 ( no mat..
Is there a limit to the number of parked calls Asterisk can handle?Th..
short question : does Asterisk reserve RTP ports for every IP-phone that is being called ?If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ?I guess Asterisk sends in the SIP INV..
HiWe have come across a situation where we are loosing synch of party 1 & party 2 voice in call recording. Here is the scenario Party 1 initiate a call to Party 2 using AMI commands When both calls are connected, we bridge these 2 calls. Then we st..