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Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf file: [3002] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Allow codecs in order of preference allow=ilbc

Hi... I just started working with Asterisk and found something that looks like an error, but i want to be sure, so that's why I'm asking you. When i make a call from "A" to "B" (both SIP clients), I don't see the name of the called party in the phone that initiated the call, just the dialed number. I made an ethereal trace and found out, that there is no name

Hi everyone, I'm currently trying to play a little with WebRTC using sipml5 client and Asterisk trunk (370821) It seems the the WebRTC implementation for Asterisk 11 is already available in the trunk? Am I right? http://lists.digium.com/pipermail/asterisk-dev/2012-July/055940.html I'm having trouble to even register to my Asterisk server using sipml5 client. The only reference to

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100