similar to: IAX -> SIP Call Help; IAX with G729

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

Hi, Firstly, what I'm trying to do is: * Get asterisk to pick up a SIP call via a DID * Prompt the user * When the user puts in a number, go to IAX.conf and route it according to what I've specified there, i.e Least Cost Routing, etc. I've set-up something similar to what I've found online, but it doesn't work! Asterisk doesn't pick up the call at all..... :( The files

Greetings! to all Now, with some hard time and help from many genurous people's in the list, I have come to this point with my TDM20B card & my teliax's IAX2 account. I hope someone may help me with this issue mentioned below. I have already selected my codec as gms in my iax.conf as well as in teliax's "my account page" but still i have the same error when I attempt

I'm trying to get Asterisk to pass thru calls using the G729 codec. I've got a 7960 phone and my gateway is an AS5400. I got the following messages when debugging SIP (7778881000 is the 7960): WARNING[1872]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/7778881000-2874(4) to SIP/as5400-35c1(256) WARNING[1872]: app_dial.c:1002 dial_exec: Had to drop call

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

Hello, all. I have some problem when testing my IP phone with Windows Messenger. My IP phone supports such codecs as g729, g711. And Windows Messenger supports red, g711, g723 as you know. The problem comes up when testing with this sip.conf file. ([general] context displayed only) =================================================================================== [general] port=5060

Dear Folks, I'm running Asterisk 1.4.31 server, on an Ubuntu 9.10 system. My scenario is simple: connection to the PSTN directly via SIP, using g729 codec, and connection to the softphones (X-lite 3.0 build 56125) trought local network, using ulaw codec. Sometimes, I got messages like: [Jul 1 15:26:16] WARNING[26483]: chan_sip.c:5514 process_sdp: Unsupported SDP media type in offer: image

If a call comes into my Asterisk server on a DiD provided by an ITSP and the dialplan sends that call to another external number throught the same ITSP's network, I don't want the RTP packets to pass through my server once the call is bridged. I have had great success getting this to work using IAX, but I have not been able to get this to work with SIP. The call is bridged OK (media at

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

The setup: Inc SIP Call -> Asterisk 1 -- IAX --> Asterisk 2 -- SIP --> phone (3044) Asterisk 1 shows the following: (1.0.3) -- Executing Goto("SIP/XX.XX.XX.XX-0819f590", "cytel-internal|3044|1") in new stack -- Goto (cytel-internal,3044,1) -- Executing Dial("SIP/XX.XX.XX.XX-0819f590",

Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten => 111, 1, Dial(chan_h323, H323/111@phone2) (in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the

Hi all, I install the last version of Asterisk and I tried to send faxes, but nothing works. Here is my configuration: Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA <----> Analog Fax 2 I tried Analog Fax 2 -> Analog Fax but nothing works!! In the Patton configuration I put G711 and no silence suppression. In asterisk I have

Dear all, I am getting the below errors when using asterisk. I am using sjphone for testing purpose. Below are the setting for sip.conf and extension.conf When i dial the number it rings on the remote telephone. but after ringing 1 time it will disconnect. Can anybody tell me what does this error means and the how to solve this issue. Thanking You, Joel sip.conf [general] context=default

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", "SIP/1014|40|t") in new stack

Am I reading the data below incorrectly, or does it appear that even though I have the directive canreinvite=no set for the two asterisk boxes, they are trying to do a reinvite (which fails) anyway? Is this expected behaviour in this situation? If so, how can I prevent this? ---- Lots of output ---- Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608)