Im new to this list. Im trying to configure my Asterisk to have user access their email. SO far users can leave voicemail but they cant access voicemail. As you can see I had sip.conf and extensions.conf below. Please advice how to access configure extensions.c..
all, I got three asterisk-machines, two of them acting as proxies. On one machine (sles11sp1) i got iritating messages about not bing able to find codecs and other stuff, so i thought it might be time for an update: Stupid! I went originally from a alm..
Via: SIP/2.0/UDP 10.11.22.161:10000;branch=z9hG4bK-a860600ex0dx0a From: Jian Gao ;tag=7e9c4091bfc704bco0x0dx0a To: Jian Gao x0dx0a Call-ID: firstname.lastname@example.org CSeq: 48998 REGISTERx0dx0a Max-Forwards: 70x0dx0a Contact: Jian Gao ;expires=60x0d..
I use voip.ms and have no issues using IAX and Asterisk 1.4.xx Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall Their on line config samples just work! Suggest you check your firewall and your configs, and above all post s..
I have a general question about SIP access for nonregistered users. I would like to make it possible for basically anybody to make a SIP call to my asterisk without having to have a user account, but in a specific context. So that e.g. somebody co..
Hy all of you,
I try to access to the IRC canal without success. Where should I
subscribe to be allowed to chat?
Using the AMI how do I tell if a call is on hold? I use the core show channels concise command to get a listing of the channels. Basically I want to use the AMI to transfer a call(which I can do) but if the person has more than one call going how..
Virendra, It may be problem for rtp packet port forwarding if u can dial through DID number. You need to open rtp port range in firewal. e.g. 10,000 to 20,000 port. please, write how can you dial call mobile or other devices. e.g. DID number, PRI num..