Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFF..
Hi. Anyone know how to fix this problem below. Im add codecs AMR-NB and AMR-WB and I get error in AMR-NB about this Found unknown media description format AMR for ID, a search about this on google and I cant find any solution about this. Tha..
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..
I have a Grandstream HT-502 device connected to my Asterisk PBX.It is configured not to place anonymous calls, and from my mostly layman reading of the invitation that the device sends, it should not be anonymous.However, the Asterisk PBX sends an anonym..
My VSP uses Asterisk to which Im connected with an ATA. When I receive an inbound call the invite includes the following… v=0 o=root 32218 32218 IN IP4 18.104.22.168 s=session c=IN IP4 22.214.171.124 t=0 0 m=audio 16864 RTP/AVP 18 101 a=rtpmap:18 G729/8..
all, I have a problem of Music on Hold on AsteriskNow system, based on Asterisk 126.96.36.199 with FreePBX 188.8.131.52 On another system, when we press the HOLD button on the phone, the phone sends an INVITE with a=sendonly in the SDP, and we get an OK and ..
Hi I request your help because i dont have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364xxxx (official number) 081169xxxx (Nddi Number) When i receive a c..
Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no luck. I have incoming calls working, but no outgoing. If get outgoing working then incoming dont work. I have s..
On 01/22/11 22:04, Da Rock wrote: > On 01/22/11 20:00, Da Rock wrote: >> On 01/21/11 20:28, Da Rock wrote: >>> On 01/21/11 03:19, Tom Rymes wrote: >>>> On 01/19/2011 10:34 PM, Da Rock wrote: >>>> >>>>> WARNING: chan_sip.c:19069 handle_response_invi..
All, I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am noticing a delay calling in and out via the FXO, but calls to local extension are ok. What i noticed when i used ngrep is that, it sends invite but got no response from the serv..