Inbound not working

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Inbound not working

Postby udfxrookie » Fri Jan 27, 2012 1:26 pm

I use Xcast as my main carrier and I use DIDLogic for my inbound line... in ViciDial I create a carrier for didlogic, have it not active and just create my inbound route.... I know with OSDial it's supposed to be super user friendly... but having issues. I created a new carrier with the same settings:

Code: Select all
[didlogic-trunk]
host=sip.didlogic.com
username=*****
secret=didlogic
type=friend
insecure=port,invite
context=trunkinbound

Registration:
Code: Select all
*****:didlogic@sip.didlogic.com/12034471212

Then I click add DID
for the number I put: 12034471212
and point it to the Ingroup I created which is checked within my main campaign and every user in that campaign has inbound for that group check... however now when I call the number I get pure dead air... any idea what it could be (hard to check asterisk live with 30 agents on... looks like the matrix lol)

OSDial Version: 2.2.9.087/1414-394
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Posts: 69
Joined: Thu Feb 10, 2011 10:11 am

Re: Inbound not working

Postby fadmin » Fri Jan 27, 2012 2:47 pm

I'd remove the one as I'm sure the carrier is not sending you 11 digits.
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Re: Inbound not working

Postby udfxrookie » Mon Jan 30, 2012 1:13 pm

tried dropping the one... finally got the room off for a debug:

Code: Select all
<------------->
[Jan 30 12:11:12]
<--- SIP read from UDP:178.63.100.24:5060 --->
INVITE sip:2034478807@108.9.164.52 SIP/2.0
Via: SIP/2.0/UDP 178.63.100.24:5060;branch=z9hG4bK7fb69dba;rport
Max-Forwards: 70
From: "17278155871" <sip:17278155871@178.63.100.24>;tag=as64e336b0
To: <sip:2034478807@108.9.164.52>
Contact: <sip:17278155871@178.63.100.24>
Call-ID: 08f20d1f08145799393f498739240323@178.63.100.24
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze2
Date: Mon, 30 Jan 2012 17:11:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 318

v=0
o=root 494051477 494051477 IN IP4 178.63.100.24
s=Asterisk PBX 1.6.2.9-2+squeeze2
c=IN IP4 178.63.100.24
t=0 0
m=audio 18452 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
[Jan 30 12:11:12] --- (14 headers 14 lines) ---
[Jan 30 12:11:12]   == Using SIP RTP CoS mark 5
[Jan 30 12:11:12] Sending to 178.63.100.24 : 5060 (no NAT)
[Jan 30 12:11:12] Using INVITE request as basis request - 08f20d1f08145799393f498739240323@178.63.100.24
[Jan 30 12:11:12] Found peer 'didlogic-trunk' for '17278155871' from 178.63.100.24:5060
[Jan 30 12:11:12] Found RTP audio format 0
[Jan 30 12:11:12] Found RTP audio format 8
[Jan 30 12:11:12] Found RTP audio format 18
[Jan 30 12:11:12] Found RTP audio format 101
[Jan 30 12:11:12] Found audio description format PCMU for ID 0
[Jan 30 12:11:12] Found audio description format PCMA for ID 8
[Jan 30 12:11:12] Found audio description format G729 for ID 18
[Jan 30 12:11:12] Found audio description format telephone-event for ID 101
[Jan 30 12:11:12] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Jan 30 12:11:12] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jan 30 12:11:12] Peer audio RTP is at port 178.63.100.24:18452
[Jan 30 12:11:12] Looking for 2034478807 in trunkinbound (domain 108.9.164.52)
[Jan 30 12:11:12]
<--- Reliably Transmitting (no NAT) to 178.63.100.24:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 178.63.100.24:5060;branch=z9hG4bK7fb69dba;received=178.63.100.24;rport=5060
From: "17278155871" <sip:17278155871@178.63.100.24>;tag=as64e336b0
To: <sip:2034478807@108.9.164.52>;tag=as00496d7d
Call-ID: 08f20d1f08145799393f498739240323@178.63.100.24
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.18-75
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Jan 30 12:11:12] NOTICE[3561]: chan_sip.c:20660 handle_request_invite: Call from '43282' to extension '2034478807' rejected because extension not found in context 'trunkinbound'.
[Jan 30 12:11:12] Scheduling destruction of SIP dialog '08f20d1f08145799393f498739240323@178.63.100.24' in 32000 ms (Method: INVITE)
[Jan 30 12:11:12]
<--- SIP read from UDP:178.63.100.24:5060 --->
ACK sip:2034478807@108.9.164.52 SIP/2.0
Via: SIP/2.0/UDP 178.63.100.24:5060;branch=z9hG4bK7fb69dba;rport
Max-Forwards: 70
From: "17278155871" <sip:17278155871@178.63.100.24>;tag=as64e336b0
To: <sip:2034478807@108.9.164.52>;tag=as00496d7d
Contact: <sip:17278155871@178.63.100.24>
Call-ID: 08f20d1f08145799393f498739240323@178.63.100.24
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze2
Content-Length: 0


<------------->
[Jan 30 12:11:12] --- (10 headers 0 lines) ---
[Jan 30 12:11:12] Really destroying SIP dialog '08f20d1f08145799393f498739240323@178.63.100.24' Method: ACK
static-108-9-164-52*CLI>
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