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Sip Trunk no outgoing

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Tinos

Technical User
Nov 5, 2014
713
BW
Hi all.
I have a sip trunk between a 3300 and Matrix Eternity. The 3300 can receive calls from the Matrix but cant make outgoing calls to the Matrix. When you dial a number going to the Matrix the IP phone displays the number then reverts the display to the idle screen. No tone or message shows why.
It is as if the call is never sent to the trunk.
What could be wrong?
 
Do a CCS trace and a SIP TAIL ON and see what it tells you.

Your digits dialled, route, digit mod, CoR and CoS on the trunk is all sound I suppose?
 
2017-07-11 15:42:14 SSP App->SSP
Event Type : MakeCall
Call Object: 0 Session: 0x105cb7a8 App: 2 Unique: 10000043
Realm : Local_150, User Name :
To: sip:301@150.36.42.90
From: "Kanoonya R" <sip:3005@172.18.174.130>
RequestUri: sip:301@150.36.42.90
Contact: "Kanoonya R" <sip:3005@172.18.174.130>
Asserted Identity List:
String List : "Kanoonya R" <sip:3005@172.18.174.130>

SessTimer: 90 MaxFwds: 26
Name : Require, Value : 100rel
Type: application/sdp
Length : 293
v=0
o=- 4470 4470 IN IP4 172.18.174.151
s=-
c=IN IP4 172.18.174.151
t=0 0
m=audio 50384 RTP/AVP 102 8 0 18 121 101
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:121 L16/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
 
There are 1 SIP link(s) programmed.
Link Matrix state is IN SERVICE (link down count 1)
 
Link said:
Matrix state is IN SERVICE (link down count 1)]

(link down count 1) surely indicates that the link is down and therefore out of service. You maybe need to do a sip trace to see what that says rather than a CCS

SIP TAIL <TAIL_FLAG>= on and make a call and that will output to the screen (set to =off to turn off).
 
I am able to recieve via the same trunk. How then is it that it is down?
What is the full sip trace command?
 
I would try and run a wireshark trace and see what message you receive when trying to call the Matrix.
 
Sip Tail ON gives the below:
2017-07-11 15:42:14 SSP App->SSP
Event Type : MakeCall
Call Object: 0 Session: 0x105cb7a8 App: 2 Unique: 10000043
Realm : Local_150, User Name :
To: sip:301@150.36.42.90
From: "Kanoonya R" <sip:3005@172.18.174.130>
RequestUri: sip:301@150.36.42.90
Contact: "Kanoonya R" <sip:3005@172.18.174.130>
Asserted Identity List:
String List : "Kanoonya R" <sip:3005@172.18.174.130>

SessTimer: 90 MaxFwds: 26
Name : Require, Value : 100rel
Type: application/sdp
Length : 293
v=0
o=- 4470 4470 IN IP4 172.18.174.151
s=-
c=IN IP4 172.18.174.151
t=0 0
m=audio 50384 RTP/AVP 102 8 0 18 121 101
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:121 L16/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
 
SIP TCPDUMP ON
Trunk & Line side
Starts TCPDUMP (capturing signaling packets) for SIP signaling on MiVoice Business.
Captured data is saved into files and placed in the /Vmail folder. All SIP signaling files in the /Vmail folder have names in the format of WS_xxxxxx.pcap. A maximum of 20 files of approximately 55-Mbyte size can be created by MiVoice Business. To collect captured data, manually ftp all WS_xxxxxx.pcap files in the /Vmail folder and use the Wireshark application to view them.
NOTE: Once retrieved, it is strongly recommended to delete all WS_*.pcap files.

Followed by SIP TCPDUMP OFF when you have finished.(before downloading the files).

But I suspect your call is stopping at the end of the bit it is showing. And the link is going down when the call fails. But not absolutely sure.
 
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