Im facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package).Please note: PJSIP is a free and open source multimedia communication libr..
Using a T.38 termination, you can receive faxes using g729, but not while using alaw. If thats the case, your Asterisk installation will need to detect the Fax Tones so as to make the decision the incoming call is a fax and then switch to the fax extensi..
Is there a way to have ALSA accept more than one incoming call? I have asterisk running a box with an audio source input. So the incoming call just connects the audio feed. Issue is I want at times to source that feed to more than one call. Can I..
Currently my asterisk system pickup incoming call after 3 or 4 rings.
How can I ask it to answer the call on the first ring? I put
immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no
Thanks in advance 🙂
Hi Have an asterisk. Setup a couple of friends. Sip.conf – http://pastebin.com/zUgiYbBi Trying to make incoming call, and have such error(cli output) http://pastebin.com/zFfgYcNR NOTICE: chan_sip.c:23316 handle_request_invite: Call from RMT20 (192.168.8.1:50..
My question is so complex and I try to explain well. We have a customer that he wants limits incoming calls to his extensions to only one. Thats not complicated with GROUPCOUNT, DEVICE_STATE or SIPPEER with curcalls option.But the problem is when ..
Does asterisk 1.8 or 10 provides any events or Dialplan application to detect MCID (Malicious Caller Identification) in an incoming call?
Please provide any sample Dialplan if possible.
Thanks & Regards,
We are facing an issue with asterisk in the case of call-Transfer scenarios. Our requirement is to identify whether an incoming call is a fresh incoming call or a Transferred call from some other clients. We have a setup, where in the asterisk1.6 ..
Dear Moderator, I need an assistance for below problem: 1. there are time incoming call from extension 100 that I dont create, how to know where is the origin of this extension? 2. when I pick up the call, there is no sound at all, this is very iritat..
Is it possible to distribute QUEUE information among several Asterisk nodes in a multimaster or load balancing setup? I havent tried this yet but if one uses realtime with a clustered multimaster database and the queue agents/members are fixed SIP chann..