similar to: Need Help with SIP/H323.

I have a remote installation that connects via IAX from my office pbx. When I call an extension on the remote pbx, after the dial period, the call is terminated. Nothing I do in configuration of that extension seems to matter: -- Executing NoOp("IAX2/netconcepts@nnn.nnn.nnn.nnn:4569-5", ""Dial 710"") in new stack -- Executing

I have been working on integrating some FXS ports into my dial plan delivered via a channel bank and testing with an analog handset. The receptionist is on Extension 700. All other SIP phones are 7XX. >From a SIP phone I can dial 700 and all other extensions. >From the analog handset I can dial any other extension but not the 700 number. Weird? Yep. The CLI does not show any dialing when I

An application I am running provides a dial tone to my users, read 9 digits, checks whether or not the called party number should be allowed and then dials out using overlap dialing on a pri channel. I.e. exten => _XXXXXXXXX,1,AGI(pm-check-destination.agi) exten => _XXXXXXXXX,2,Dial,Zap/g1/BYEXTENSION|60|CH The AGI-Skript takes about 0.3 to 0.5 seconds (it does a number of rather complex

Hello all, I have an extension like this in my test box: exten => _XXXXXXXX,1,Answer() exten => _XXXXXXXX,n,Playtones(350/100,0/750) exten => _XXXXXXXX,n,Dial(SIP/${EXTEN}) The idea is that occasionally the SIP peer might take a few seconds to connect the call, and this provides an in progress sound instead of silence. I don't want to provide a "ring" tone followed by a

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

Sir, i am using goautodial server , bri card is showing ok but when i try to call that showing below , This configuration is in doing in dubai , so kindly help me how can connet the call from this , what is my mistake is in this :::chan-dahdi.conf [channels] #include dahdi-channels.conf language=en context=default usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=yes

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

Hello! How can one select outgoing MSN when dialing out from ttyI-interfaces? I have successfully done this with CAPI e.g... exten => _XXXXX.,2,Dial,CAPI/60:bBYEXTENSION ...in extensions.conf. Currently correponding for my ISDN modem interface is... exten => _XXXXX.,2,Dial(Modem/g1:${EXTEN}) ...but this selects only MSN of outgoing group g1 for dialout MSN number. I also tried to

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

>any feedback would be awsome, the idea is to fill in the 30 channels of >the E1 all at the same time and see how stable it can be > > ----- Original Message ----- > From: "Miguel Cavazos" <miguel@cavazos.com.mx> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Sent: Wednesday,

Hi all, I have configured Asterisk to call to PSTN phone from our IP phone, But I am unable to call my IP phone from a PSTN phone (If I called any number between 21494350 and 21494399, the card should route my call to my IP phone, IF my configuration was correct). I have done my research and gathered bits and pieces of information (which are vague by the way) but still cannot call my IP phone from

Hi list users I?m trying to do an incluye statement from the Database In my dialplan I have different contexts that defines common services for example ---------------------------------------------------------------------------- ----------------------- [basic_services] exten => 100,1,VoicemailMain() exten => 600,1,Playback(demo-echotest) exten => 600,2,Echo() exten

It looks like you have the first extry of the [incoming] context in extensions.conf commented out Hi all, I am resending this message, so far no one has helped me with this incoming call issue. there is no problem with outbound call but there is no inbound call to my sip phone. the only message I get when I call from pstn is "unable to create local channel for call forward to

I don?t know if the last message was with content. So, I sent again. I have installed a Digium card TE210P and unicall for use MFC/R2. I think that it?s all right but I can?t make and receive calls. I?m using asterisk 2.1 with the patch made by Jos? P. Leit?o and the follow libs: libsupertone-0.0.2 libunicall-0.0.3 libmfcr2-0.0.3 zaptel 2.1 My number is 34318300. The Telco send me only 8300.

Hi To dial an outside line i have to dial 0. I want to have that when we dial local numbers, that is we are in the 08 area, I don't want to have to dial 08, how to set this up in asterisk 1.6? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460

I can't seem to make the fax detection work. Here's an excerpt from zapata.conf: signalling=fxs_ks group=0 context => guestaccess channel => 47-48 and from extensions.conf: [guestaccess] include => incomingmain [incomingmain] exten => s,1,Dial,Zap/1&Zap/9&Zap/10&Zap/11|24 exten => s,2,Voicemail,u7000 exten =>

Hi all, I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028. This IP phone is registered in an Asterisk PBX and I've a problem when I dialing internal number. If I dial an internal number, like for example 102, the IP phone takes 35 seconds to send the number to Asterisk; here below the debug output 192.168.0.75: first 192.168.0.75: 1 192.168.0.75: 10 192.168.0.75: 102

I've buyed SIP traffic from platinumcalling.com.au. in sip.conf i have: [platinum] context=default type=peer username=MYUSERNAME secret=MYPASSWORD host=sip.platinumcalling.com.au in my extension.conf: [default] exten => _XXXXX.,1,Dial,SIP/${EXTEN}@platinum|60|r When i try to call a number i got: Got SIP response 404 "Not Found" back from 203.30.19.164 SIP/platinum-67f2 is

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

Hello there I'm quite a newbie in the IP Telephony area. I'm playing a little around with a setup with one linux box with a e100 p card installed, which functions as an Asterisk gateway and a oh323 GK(Gnu Gatekeeper). I have two h323 phones, Welltech WellGate 1501 and 3502. So far I've managed to get the two IP phones and Asterisk connected to the GK. I can place calls from one