search for: go2cal

Any got experience with these? I couldn't fint anything in any postings... it seems they have a h.323 on voip01.go2call.com and a sip on sip01.go2call.com I have tried to register with some of the same as I use for nikotel, but Asterisk does not want to register. I've tried to use both the user name (ingvald) and the PIN code 440.... as authentication. ---from sip.conf---- register => 440686267684:x...

...br> AGI Tx >> agi_context: default<br> AGI Tx >> agi_extension: 01236337388<br> AGI Tx >> agi_priority: 1<br> AGI Tx >> agi_enhanced: 0.0<br> AGI Tx >> agi_accountcode: <br> AGI Tx >> <br> AGI Rx << exec dial "sip/go2call/551236337388|60|TtS(3600)"<br> &nbsp;&nbsp;&nbsp; -- AGI Script Executing Application: (dial) Options: (sip/go2call/551236337388|60|TtS(3600))<br> &nbsp;&nbsp;&nbsp; -- Setting call duration limit to 3600 seconds.<br> &nbsp;&nbsp;&nbsp; --...

...in buying the hardware until I get the networking sorted. Somehow I need to TUNNEL throught the router and firewall? I was looking at <http://en.wikipedia.org/wiki/STUN> as one way of doing this. The Grandstream FAQ explains how to do this: '5. How do I setup my Grandstream Phone for go2call network? typical configuration is: SIP Server: voip01.go2call.com Outbound proxy: (Should leave it blank, because it's a GW) User ID: xxxxx (your Go2Call PIN number) Authentication ID: same as your User ID Password: xxxxxxx (Your Go2Call password) NAT Traversal: YES (WITHOUT setting the STUN s...

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the Go2Call side, not the SJPhone cos I can dial from SJPHone t...

Hi Everybody, I'm trying to make my asterisk dial a international call from a SER request of it. My ambient is like this. [Clients]--[SER]--[Asterisk]--[Go2Call] Client: My SIP clients. SER: My REGISTRAR/Proxy Server Asterisk: All other services(Voicemail,musiconhold etc) and also acting as an UAC dialing International Calls, because SER doesn't do that sending username, password and host. Go2Call: My International Sip Provider. For my Client Loc...

...lls to each other through my SER Server, that's to say, non external or international calls. I would like my clients to make external and international calls through my server but for that they must authenticate at another server to have a valid VoIP phone number. I want them to authenticate at Go2Call VoIP Server to make internatinoal and external calls, but for that i need my ser to forward every call begginning with "1" to Asterisk and Asterisk to auth at Go2Call Server sending username and password (Funcionality that ser doesn't have) ending the VoIP tunnel and making the call...

...t I can still connect my sip phone to my server but it doesn't give me an outside line after I press 1. Have anyone implemented this solution or know what I may be doing wrong ?? My configurations are following below: Extensions.conf exten => 1,1,Dial(SIP/<username>:<password>@go2call,30,rT) exten => 2,1,Playback(tt-weasels) exten => 2,2,Hangup() exten => 3,1,Playback(tt-weasels) Sip.conf [go2call] context = go2call username=<username> secret=<password> auth=md5 type=friend host=<go2callhost> -- Felipe Martins TEP Solution &...

...E100P card from Digium, I already have a basic asterisk setup up & running. My application is the following : I want to accept incoming calls from the PSTN to Asterisk, and without asking anything of the client just pass them immediately to a call gateway in USA, actually we are planning to use Go2Call.com Currently the company I work with already runs a number of callcentres where a client can walk in, pick up a phone & use a calling card, they are routed through hardware boxes made by Quintum (tenor A800) so i'm thinking of something in a similar vein but taking calls off the PSTN. Any...

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? O...

...w 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 Dial("SIP/217.168.168.49:5060", "SIP/723@216.52.153.207") in new stack -- Called 723@216.52.153.207 WARNING[1240577216]: File channel.c, Line 1711 (ast_channel_make_compatible): No path to translate from SIP/216.52.153.207-2e12(1) to SIP/217....

...alls. I have SER working with mysql without problems, all my internal users autenticate at SER and then if any number begins with a "1", Ser forwards the call to asterisk. Asterisk takes the forward and act as a Sip Client to make the call. I need to do that because My VoIP numbers are at Go2call, and I need asterisk to authenticate at go2call server to take line signal, to complete incoming/outgoing calls. At this moment none of them are working. When I try to dial number 13125899691 I receive the following Error: . . . Looking for 13125899691 in OUTGOING Reliably Transmitting (no NAT):...

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to be using it .. connecti...

...e, asterisk crashes the machine after trying to read the SIP.conf (Crashes to the extent that the machine freezes .. ) What I would *like* the system to do is as follows : for now, just take an input call from a softphone and route it through to an internet calling gateaway (I have an account with Go2Call) in such a way that I can play around with the scripts & work out how to bill it .. in future I'd like to route calls from a number of H323 calling gateways in different locations to pass through a central node & bill everything before forwarding the calls to a gateway in USA. If any...

Hi all, i use the DISA app for giving the user a trunk after a authentication through PGSQL as follows .... auth via PGSQL exten => s,1,DISA,no-password|test I think the user is now in context "test" and he could dial any number if the extension-conf in "test" is for example exten s,1,Dial,OH323/<myip> But if the user dial one digit the call build up

I'm configuring my SER to forward calls based in extension. Cause I would like my ASTERISK to do international calls. How could I make ASterisk do international calls ?? I must pass the host (Go2Call), username and password to get the call up, but I don't know how. I'm trying to find a extension command that like Dial, does the call but passing username, password and host for authentication. Is there a way to do that ? Thanks in Advance. -- Felipe Martins Linux System Administrato...