Archives : November-2014
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there,Im trying use asterisk to connect against mgcp endpoints, but i would like to use realtime mgcp gw/endpoint configuration, but i cannot find any documentation about howcan I configure realtime mgcp gw configuration. Anyone here can guide me..
Im banging my head last few days trying to get volume level on channel. So far, I tried using sox with no luck.sox file.wav -r 1 file.datgave me volume levels but not before Asterisk finish call recording.What I really need is to detect volume le..
Howdy,Im trying to re-write my voicemail check extension.I formerly used the SIPPEER function to get the mailbox for a peer with${SIPPEER(${peer},mailbox)}Is there a way to do this with PJSIP now that Ive converted over?I see a function PJSIP_ENDPO..
Hi I have installed Asterisk 11.13.1 on Fedora running in VirtualBox. The VB network interface is configured to use NAT. The host machine is Windows 7 and is connected to a SIP server using a VPN connection. I have configured ‘externaddr’, ‘localnet..
group and thank you for the attention.Im using Asterisk 11.12 running on Ubuntu Server 12.04We have an issue with channels remaining open after a SIP peer unregisters.It seems that if the peer goes away before manually hanging up a call, the chan..
I came thru ISDN UUI (User-User Information) protocol which is defined in this RFC – http://www.ietf.org/id/draft-ietf-cuss-sip-uui-17.txtBut I dont understand how to use this with Asterisk. Any idea would be much appreciated.Than..
I have 5 Asterisk servers all using mysql realtime to store queue log information.There is 1 out of 5 servers which stores the data in 4 columns : data1 –> data 5.All other servers store data in 1 column data with the data seperated by pipe.I see..
First I am new to PBX so i might be doing something fundamentally wrong… That being said I got a FreePBX 32bit stable 6.12.65.I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: Note ..
All, I am using asterisk-11.12.0 version and I am trying to setup secure call(TLS + SRTP) between two extensions and while making a call, I got following error*CLI> == Using SIP RTP CoS mark 5– Executing [6004@from-office:1] Dial(SIP/6003-00000000,SIP/6004,..
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