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...t; 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 & New Technologies Mundivox Communications fmartins@mundivox.com Site: www.mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844

....org/wiki-Asterisk+config+sip.conf http://www.voip-info.org/wiki-Asterisk+SIP+channels http://www.voip-info.org/wiki-Asterisk+introduction Has anybody have this Scenario, that could give me a hand ? It's really starting to piss me off ... Thanks in Advance Best Regards. -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins@mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

Hi I am having alot of difficulty connecting to SIP providers (I am trying 3) and can't seem to find anything similar in the wiki or on the lists.....I can receive inbound calls fine however on placing an outbound call the calling phone never gets 'connected' but 2 way audio is passed for about 20secs before some sort of timeout. Anything suggestions as to what I could try

...ware I could use to do so. I'd also like to know how to make CDR with asterisk, once my Clients authenticate at SER, is it a problem for asterisk to generate logs for CDR ??? Any help will be very aprecciated. Best Regards. -- Felipe Martins Linux System Administrator Tep Solution Provider Mundivox Communications Rua Lauro Muller, 116/Sala 505 RJ - Brasil - 22290-906 Tel.: 55 21 3820-8839 Fax.: 55 21 3820-8844

...d 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 Administrator Tep Solution Provider Mundivox Communications Rua Lauro Muller, 116/Sala 505 RJ - Brasil - 22290-906 Tel.: 55 21 3820-8839 Fax.: 55 21 3820-8844

...erisk -r , sip debug), when I press 2, which was the key I choose to forward the request to Asterisk. I don't know whatelse to do. I'm sure I'm missing some configuration. Can anybody help me ? Thanks in Advance. -- Felipe Martins Linux System Administrator Tep Solution Provider Mundivox Communications Rua Lauro Muller, 116/Sala 505 RJ - Brasil - 22290-906 Tel.: 55 21 3820-8839 Fax.: 55 21 3820-8844

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Hi everybody, I have a SER Server (Sip Proxy / REGISTRAR) and a Asterisk Server (PSTN and other services). I've got some clients that make calls 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