Hello,
I have also problems with my IAX tel.
When I call other numbers then mine it works.
But when I call mine it does not want to work, can somebody help me?
-- Executing Dial("SIP/1011-2161",
"IAX2/greentone:xxxx@iaxtel.com/17009016976@iaxtel") in new stack
-- Called greentone:xxxx@iaxtel.com/17009016976@iaxtel
-- Call accepted by 12.37.165.130 (format GSM)
-- Format for call is GSM
NOTICE[147466]: File chan_iax2.c, Line 4060 (socket_read): Rejected connect
attempt from 12.37.165.130
WARNING[147466]: File chan_iax2.c, Line 4545 (socket_read): Received mini frame
before first full voice fr
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I would like to say also that ealier it was working.
Today I updated * from CVS and now I have this problem
Hello,
I have also problems with my IAX tel.
When I call other numbers then mine it works.
But when I call mine it does not want to work, can somebody help me?
-- Executing Dial("SIP/1011-2161",
"IAX2/greentone:xxxx@iaxtel.com/17009016976@iaxtel") in new stack
-- Called greentone:xxxx@iaxtel.com/17009016976@iaxtel
-- Call accepted by 12.37.165.130 (format GSM)
-- Format for call is GSM
NOTICE[147466]: File chan_iax2.c, Line 4060 (socket_read): Rejected connect
attempt from 12.37.165.130
WARNING[147466]: File chan_iax2.c, Line 4545 (socket_read): Received mini
frame before first full voice fr
-------------- next part --------------
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URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20031023/d5182633/attachment.htm
Hello, I'm having problems connecting to other * boxes through IAXTel. I've seen this addressed in the list archives, and other places on the web, but haven't seen that anyone has come up with a solution. I'm dialing in to my Asterisk server using DISA, authenticating OK, then attempting to dial out and keep getting "IAX2/69.73.19.178:4569/8 stopped sounds" and it hangs up. I've tried switching codecs as I saw someone suggest, but get the same result. This happens with 1-700 numbers as well as 18XX numbers that I know work properly, so I don't think it's a misconfiguration on the receiving server's side as some have suggested. I've found people posting about this several times over the past year and a half, so I imagine the problem is pretty common, and is something misconfigured on my side, or some kind of common conflict. Any ideas? PF
Hi, I tried to add the IAXTel config to my asterisk, so i can dial free numbers inside the US from my SIP softphone (X-lite), everything seems to be working, but the sound quality is terrible, the other side sounds like a "digitized" voice, and the voice is cut, i cant hear a full word, I tried using FWD IAX interface, and no problem there, it works great. Now, although this is in a testing phase, i wanted to know if i am missing something, or IAXTel is just problematic . I am "dialing" from Israel, over a E1 line, dont know exactly how much of my E1 reaches the US, but should be sufficent for one session (for which FWD works fine with) Any help appriciated. Marco.