I have an Elastic PBX (extremely similar to trixbox) that I am having an issue with. I cannot get the incoming routes to work correctly. If I setup an any/any route it works fine once I try to configure by DID it stops working. I need some help redoing the trunks and incoming routes.
I am willing to pay please respond with how much you will charge.
Youre probably not matching the pattern your provider is sending to the inputs you are putting in.
You'll need to watch the asterisk log during an incoming call to see how theyre presenting the DID.
You might have put in 12223334444 while the provider is sending 3334444 and so on...
check that
Well I just checked teh call log "/var/log/asterisk/cdr-csv" It doe snot show a DID that is being called. I tried 2 different DID's that I have for 2 different cities. I get the following.
"","3035780587","101","from-did-direct","""3035780587"" ","SIP/from-voicemeup_trunk-00000022","SIP/101-00000023","Dial","SIP/101,15,tr","2010-08-01 13:51:54","2010-08-01 13:52:04","2010-08-01 13:52:11",17,7,"ANSWERED","DOCUMENTATION","1280670714.34",""
"","13035780587","101","from-did-direct","""13035780587"" ","SIP/from-didww5-00000024","SIP/101-00000025","Dial","SIP/101,15,tr","2010-08-01 13:53:27","2010-08-01 13:53:33","2010-08-01 13:53:41",14,8,"ANSWERED","DOCUMENTATION","1280670807.36",""
I even did a capture for the call this was the output (note I called from 3035780587 and was trying to call 3039973260) I cant see the did i tried to call anywhere listed. Is the DID number NOT being passed for some reason?
Asterisk 1.6.2.8, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': == Found
[0;37m[1;30m == [0mParsing '/etc/asterisk/extconfig.conf': [1;30m == [0mFound
[0mConnected to Asterisk 1.6.2.8 currently running on pbx (pid = 3050)
pbx*CLI>
Verbosity was 3 and is now 20
[Kpbx*CLI>
Really destroying SIP dialog '3191e7915cddebd9777e2eb5298d42ca@82.103.139.232' Method: REGISTER
[Kpbx*CLI>
== Manager 'admin' logged on from 127.0.0.1
[Kpbx*CLI>
== Manager 'admin' logged on from 127.0.0.1
[Kpbx*CLI>
== Manager 'admin' logged off from 127.0.0.1
[Kpbx*CLI>
== Manager 'admin' logged off from 127.0.0.1
[Kpbx*CLI>
> doing dnsmgr_lookup for 'callcentric.com'
[Kpbx*CLI>
> ast_get_srv: SRV lookup for '_sip._UDP.callcentric.com' mapped to host alpha6.callcentric.com, port 5080
[Kpbx*CLI>
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 204.11.192.36:5060:
REGISTER sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 82.103.139.232:5060;branch=z9hG4bK1fc17425;rport
Max-Forwards: 70
From:
To:
Call-ID: 3191e7915cddebd9777e2eb5298d42ca@82.103.139.232
CSeq: 110 REGISTER
User-Agent: Asterisk PBX 1.6.2.8
Authorization: Digest username="17772162662", realm="callcentric.com", algorithm=MD5, uri="sip:callcentric.com", nonce="8531c9796798857a31f6df50225ba532", response="22923970fa376d36102a733ee1746804"
Expires: 120
Contact:
Content-Length: 0
---
[Kpbx*CLI>
SIP/2.0 200 Ok
v: SIP/2.0/UDP 82.103.139.232:5060;branch=z9hG4bK1fc17425;rport=5060
f:
t:
i: 3191e7915cddebd9777e2eb5298d42ca@82.103.139.232
CSeq: 110 REGISTER
m:
l: 0
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '3191e7915cddebd9777e2eb5298d42ca@82.103.139.232' in 32000 ms (Method: REGISTER)
[Kpbx*CLI>
INVITE sip:pbx.allnetsol.com SIP/2.0
Via: SIP/2.0/UDP 204.11.194.38:5060;branch=z9hG4bK7f0b1b7c;rport
From: "13035780587"
To:
Contact:
Call-ID: 6d1dde106b57054042e640077e91b253@204.11.194.38
CSeq: 102 INVITE
User-Agent: DIDWW
Max-Forwards: 70
Remote-Party-ID: "13035780587"
Date: Sat, 07 Aug 2010 16:08:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 411
v=0
o=root 20949 20949 IN IP4 204.11.194.38
s=session
c=IN IP4 204.11.194.38
t=0 0
m=audio 17226 RTP/AVP 0 8 18 4 111 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (15 headers 19 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 204.11.194.38 : 5060 (no NAT)
Using INVITE request as basis request - 6d1dde106b57054042e640077e91b253@204.11.194.38
Found peer 'from-didww5' for '13035780587' from 204.11.194.38:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 111
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format G726-32 for ID 111
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 204.11.194.38:17226
Looking for s in from-trunk (domain pbx.allnetsol.com)
list_route: hop:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.194.38:5060;branch=z9hG4bK7f0b1b7c;received=204.11.194.38;rport=5060
From: "13035780587"
To:
Call-ID: 6d1dde106b57054042e640077e91b253@204.11.194.38
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact:
Content-Length: 0
-- Executing [s@from-trunk:1] [1;36mNoOp[0m("[1;35mSIP/from-didww5-00000001[0m", "[1;35mNo DID or CID Match[0m") in new stack
-- Executing [s@from-trunk:2] [1;36mAnswer[0m("[1;35mSIP/from-didww5-00000001[0m", "[1;35m[0m") in new stack
Audio is at 82.103.139.232 port 12642
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.194.38:5060;branch=z9hG4bK7f0b1b7c;received=204.11.194.38;rport=5060
From: "13035780587"
To:
Call-ID: 6d1dde106b57054042e640077e91b253@204.11.194.38
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 925506540 925506540 IN IP4 82.103.139.232
s=Asterisk PBX 1.6.2.8
c=IN IP4 82.103.139.232
t=0 0
m=audio 12642 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[Kpbx*CLI>
ACK sip:s@82.103.139.232 SIP/2.0
Via: SIP/2.0/UDP 204.11.194.38:5060;branch=z9hG4bK7cfe66ce;rport
From: "13035780587"
To:
Contact:
Call-ID: 6d1dde106b57054042e640077e91b253@204.11.194.38
CSeq: 102 ACK
User-Agent: DIDWW
Max-Forwards: 70
Remote-Party-ID: "13035780587"
Content-Length: 0
--- (11 headers 0 lines) ---
[Kpbx*CLI>
-- Executing [s@from-trunk:3] [1;36mWait[0m("[1;35mSIP/from-didww5-00000001[0m", "[1;35m2[0m") in new stack
[Kpbx*CLI>
-- Executing [s@from-trunk:4] [1;36mPlayback[0m("[1;35mSIP/from-didww5-00000001[0m", "[1;35mss-noservice[0m") in new stack
--
[Kpbx*CLI>
-- Executing [s@from-trunk:5] [1;36mSayAlpha[0m("[1;35mSIP/from-didww5-00000001[0m", "[1;35m[0m") in new stack
-- Executing [s@from-trunk:6] [1;36mHangup[0m("[1;35mSIP/from-didww5-00000001[0m", "[1;35m[0m") in new stack
== Spawn extension (from-trunk, s, 6) exited non-zero on 'SIP/from-didww5-00000001'
-- Executing [h@from-trunk:1] [1;36mHangup[0m("[1;35mSIP/from-didww5-00000001[0m", "[1;35m[0m") in new stack
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/from-didww5-00000001'
Scheduling destruction of SIP dialog '6d1dde106b57054042e640077e91b253@204.11.194.38' in 32000 ms (Method: ACK)
set_destination: Parsing
set_destination: set destination to 204.11.194.38, port 5060
Reliably Transmitting (no NAT) to 204.11.194.38:5060:
BYE sip:13035780587@204.11.194.38 SIP/2.0
Via: SIP/2.0/UDP 82.103.139.232:5060;branch=z9hG4bK21cdcaa8;rport
Max-Forwards: 70
From:
To: "13035780587"
Call-ID: 6d1dde106b57054042e640077e91b253@204.11.194.38
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.8
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Kpbx*CLI>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.103.139.232:5060;branch=z9hG4bK21cdcaa8;received=82.103.139.232;rport=5060
From:
To: "13035780587"
Call-ID: 6d1dde106b57054042e640077e91b253@204.11.194.38
CSeq: 102 BYE
User-Agent: DIDWW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '6d1dde106b57054042e640077e91b253@204.11.194.38' Method: ACK
[Kpbx*CLI>
-- Remote UNIX connection
-- Remote UNIX connection disconnected
[Kpbx*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).
[0m
You are using callcentric...
Did you read/follow this?
http://www.callcentric.com/support/device/did_trixbox
Callcentric has a DID service, but in your case you are using it for outbound only... ok
i have never worked with didww, so i would suggest you contact their tech support.
From their website
"Can I Use 3rd Party SIP Services such as Callcentric.com, Sipgate.com or Phonegnome.com?
DID World Wide allows DIDs to be mapped to a number of 3rd party VoIP providers. The selection of the VoIP provider is performed by using our on-line configuration interface. For more information see How can I Use 3-rd Party SIP Services Such as Callcentric.com, Sipgate.com or Phonegnome.com? "
Are you doing this? Routing your calls through callcentric?
It seems their website in the customer section should have a config helper
It looks like from your trace that when your ddi I'd rung the number you have called is not presented
I would check your configuration at
DIDWW it seems calls are just being forwarded to the ip address of your trixbox no ddi is being passed.
You have no any/any destination setup so the call is rejected as an invalid number
DIDWW is a new one to me, but what I can gather about their SIP handoffs is they don't send any DID information. It isn't a traditional SIP trunk. It looks like you can specify a device in their online configuration tool, so it would place a call like:
DID 303-999-9990 goes to your extension 201, so when someone calls that DID, it sends it to 201@youripaddress. Is this not correct? If this is what they do, you won't get any DID information. You ARE getting an ID in the header.
How many DID's are you getting from them? You could do something like:
[custom-get-did-from-sip]
exten => s,1,Noop(Fixing DID using information from SIP TO header)
exten => s,n,Set(pseudodid=${SIP_HEADER(To)})
exten => s,n,Set(pseudodid=${CUT(pseudodid,@,1)})
exten => s,n,Set(pseudodid=${CUT(pseudodid,:,2)})
exten => s,n,Goto(from-trunk,${pseudodid},1)
Try a catch-all again and call in. Put yourself on hold and call back in. Look at the trace and see if the next call comes in on port 5061 instead of 5060. We need to determine how they are sending your calls to you. Some providers will send calls in this way.
Obviously your trunk configuration says "context=from-trunk", right? Also, did you follow their guide? As your PBX is very close to Trixbox, I'm sure their special required config would apply. It looks like you have to build a trunk for each of their servers...
http://www.didww.com/Knowledgebase/how_to_set_asteriskhome_trixbo...
Member Since:
2009-05-21