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I am using Asterisk version 11.5. My problem is the following: I literally cannot make any outbound call. That's because, whatever I dial in any client, the sip_domain part gets ignored, it is interpreted as an internal asterisk domain call:

endpoint@sip_domain

It doesn't matter what I dial, even if I dial endpoint@blablablabla the call gets redirected to my asterisk server. What might it be? If you need any other info, tell me.

Thanks in advance.

EDIT - sip.conf:

[general]
context=from-sip
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=h264
allow=h263
allow=t140
rtcachefriends=yes
rtupdate=yes
srvlookup=yes
videosupport=yes
textsupport=yes
directmedia=no
icesuport=true
;nat=force_rport,comedia
accept_outofcall_message=yes
outofcall_message_context=sip-messaging
auth_message_requests=yes

EDIT - sip debug logs

In the following files you can find debug logs of a successful call (external to internal) and a failed call (internal to external): http://sdrv.ms/1axKGli .

Hari
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1 Answers1

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If you post your concealed sip.conf it would beneficial to assist you. Also, connect to asterisk in verbose mode asterisk -vvvvr or enable sip debugging sip set debug on might tell you a bit more about the problem. Also with this type of question it would help if you posted the SIP header information as well -

Try setting allowexternaldomains=yes

note: performing a dig on the domain t-meeting.se for the SRV record yields no answer, have you tried calling another number on a sip server that supports SRV lookup?

It looks like your dialplan needs some modification. I would consider taking a look at this link. Towards the bottom they define a macro that can accomplish what you are looking for. However in short you need something like the following in extensions.conf. You will probably need to alter this to work with realtime but this is the jist: 

exten => _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
exten => _.,n,HangUp()

[macro-uri-dial]
exten => s,n,NoOp(Calling as SIP address: ${ARG1})
exten => s,n,Dial(SIP/${ARG1},60)
Shapeshifter
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  • Added more details to question. – Hari Aug 23 '13 at 21:08
  • Sorry, what about your dialplan? Could you post relevant lines from that as well? I will see what I can find! – Shapeshifter Aug 26 '13 at 15:40
  • Also, is this your entire sip.conf? You do not have outboundproxy set or anything else? Also I would like to see even more information, having the relevant extensions from extensions.conf would also be useful. There are a lot of variables that need to be considered. – Shapeshifter Aug 26 '13 at 16:31
  • Regarding your note: even if i call a random number like this: 01@blablablablabla.com , the message is the same: extension not found in context... The same with an ekiga.net account. I'll add a link to my sample extensions.conf and extensions table in db (i am using realtime) – Hari Aug 26 '13 at 19:23
  • Skydrive link has updated data with extensions.conf and extension.table (table data). – Hari Aug 26 '13 at 19:35
  • No problem! I am glad you got it working, fun isn't it !? – Shapeshifter Aug 30 '13 at 15:02