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..
This is a really simple problem that I just cant get to work. There are two Asterisk servers with the following sip user and peer. When a call is attempted, Asterisk is not sending authentication details in response to the 401. Note, if the secret..
Your Server Voipon isnt responding- See if internet is working fine, or your Voipon sip trunk/peer is registered OK? On Fri, Feb 3, 2012 at 5:53 PM, virendra bhatiwrote: > Call is not routing from server to destination. > > > app8*CLI> console dial 00918885268..
While compiling 184.108.40.206, I met the following error: [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_ge..
folks, Im having problems when I try to record my calls using MixMonitor or Monitor. Calls are working well and audio quality is good. But I just cant get recorded audio in one leg with both applications. It happens with internal calls too. As it see..
all, I am using to Xlite to save video voice mail. when i retreive it, then only video show , no voice is here out. Plz tell me where ,i am wrong , and how i can able to see video plus here audio in voice mail box. I did following configuration In Sip.c..
,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [22..
folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations t..
everybody! Lately Ive had experiences that Id like to share with you: I did a some faxing over VOIP during the last two years. Not that much, lets say 1 fax per day on average. The setup is Old analog fax machine < -> Linksys PAP2 ATA < -> Asterisk ..
all, I have a major issue with a codec renegotiation in an asterisk 220.127.116.11 setup, which leads me to ask a general question about asterisk 1.4.X codec negotiation: asterisk can support a re-negotiation of a codec on the fly through a re-Invite? If..