I am using Elastix 2.5 stable with asterisk 11.13.0. I
have connected to my voip provider via sip trunk and outgoing calls are
working. Incoming calls are also working but just for “any DID/any CID"
and “DID/any CID”. However, I tried to make incoming route for” DID/CID"
and “any DID/CID” and I got error “Received an unknown call with DID
set to myDID” with message “The number you have dialed is not in
service”. Why asterisk can not match the CID? (I tried with
allowquest=yes but without success )Am I missing something? Below is my
trunk configuration:
humm,
in that case I probably can’t suggest much else than doing
asterisk -vvvvvvvvr # not really sure how many 'v's, but 5 minimum to get very verbose output
from the ssh console, and using ngrep to catch transmissions on port 5060 in another console; it can reveal what exactly is your provider sending to you that you need to tell Asterisk/FreePBX to match to.
(the screen program/package helps a lot with this)
(it helped me when I had issues like that)
Here are prints from asterisk console when call is placed and not placed. Can I use tcpdum instead of ngrep I have some problems with installation, if you had experience with tcpdump. Thx
Call is not placed
DID/CID
any DID/CID
Executing [myDID@from-trunk:1] Set(“SIP/myprovider-out-0000009b”, “__FROM_DID=myDID”) in new stack
– Executing [myDID@from-trunk:2] NoOp(“SIP/myprovider-out-0000009b”, “Received an unknown call with DID set to myDID”) in new stack
– Executing [myDID@from-trunk:3] Goto(“SIP/myprovider-out-0000009b”, “s,a2”) in new stack
– Goto (from-trunk,s,2)
– Executing [s@from-trunk:2] Answer(“SIP/myprovider-out-0000009b”, “”) in new stack
– Executing [s@from-trunk:3] Wait(“SIP/myprovider-out-0000009b”, “2”) in new stack
> 0x2013fff0 – Probation passed - setting RTP source address to sip.myprovider.com:11252
– Executing [s@from-trunk:4] Playback(“SIP/myprovider-out-0000009b”, “ss-noservice”) in new stack
– <SIP/myprovider-out-0000009b> Playing ‘ss-noservice.gsm’ (language ‘en’)
== Spawn extension (from-trunk, s, 4) exited non-zero on ‘SIP/myprovider-out-0000009b’
– Executing [h@from-trunk:1] Macro(“SIP/myprovider-out-0000009b”, “hangupcall,”) in new stack
Call is placed
any DID/ any CID
DID/ any CID
– Executing [myDID@from-trunk:1] Set(“SIP/myprovider-out-0000009e”, “__FROM_DID=myDID”) in new stack
– Executing [myDID@from-trunk:2] Goto(“SIP/myprovider-out-0000009e”, “s,1”) in new stack
– Goto (from-trunk,s,1)
– Executing [s@from-trunk:1] ExecIf(“SIP/myprovider-out-0000009e”, “0?Set(__FROM_DID=s)”) in new stack
– After macros are checked call is placed
Do you have the inward route configured for the DID in question ? (Extension Routes (fee)free module needs to be used)
Can you indicate in the log excerpts, where the DID is a number (say use the 555 555 555 there) and where it is a string/literal name ?
(my story is, that my SIP provider was setting the DID to … the SIP username so I had to set up the inward routes to match the /literal/ string received… towards which the route module was all like, huh, what? but it worked in the end)
Other than that,
ngrep (if from source) needs libpcap and libpcap-devel.
(excerpt from my .bash-history below)
wget http://prdownloads.sourceforge.net/ngrep/ngrep-1.45.tar.bz2?download
# tar jxf ngrep-1.45.tar.bz2 # like, that was /real/ cool...
tar jxf ngrep-1.45.tar.bz2\?download
cd ngrep-1.45
./configure
yum install libpcap-devel
./configure
make
make clean
yum install libpcap
yum install libpcap-devel
Off course, but I do not know what they have changed exactly, some configuration on trunk at their side, I just complained to them that I can not figure out where is the problem. I think they configured DID number transfer on their side or something like that:)