Audiocodes SIP Gateway to IP Office v6.0(18)

[DBrewsky]

I am trying to setup calls between an Audiocodes MP124 (FXS) to a station on an IPO 500 v2 6.0(18).

I have built an incoming call route (6699) setup on the IPO to ring a x201. Also have a SIP trunk setup, Line 17.

In the Audiocodes, when 6699 is dialed I have it pointed to the IP Office IP address.

I get a failure. And need a bit of guidance.

Here is the SysMon trace of the call, and also attaching the IPO configuration.

70275mS SIP Rx: UDP 192.168.56.3:5060 -> 192.168.50.23:5060
INVITE sip:6699@192.168.50.23;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.56.3;branch=z9hG4bKac185493194
Max-Forwards: 70
From: "3792" <sip:3792@copper-state.net>;tag=1c185489868
To: <sip:6699@192.168.50.23;user=phone>
Call-ID: 1854894762012201016425@192.168.56.3
CSeq: 1 INVITE
Contact: <sip:3792@192.168.56.3:5060>
Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.6.00A.016.002
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 252

v=0
o=AudiocodesGW 185476457 185476372 IN IP4 192.168.56.3
s=Phone-Call
c=IN IP4 192.168.56.3
t=0 0
m=audio 6230 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:10
a=sendrecv
70278mS SIP Call Rx: 17
INVITE sip:6699@192.168.50.23;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.56.3;branch=z9hG4bKac185493194
Max-Forwards: 70
From: "3792" <sip:3792@copper-state.net>;tag=1c185489868
To: <sip:6699@192.168.50.23;user=phone>
Call-ID: 1854894762012201016425@192.168.56.3
CSeq: 1 INVITE
Contact: <sip:3792@192.168.56.3:5060>
Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.6.00A.016.002
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 252

v=0
o=AudiocodesGW 185476457 185476372 IN IP4 192.168.56.3
s=Phone-Call
c=IN IP4 192.168.56.3
t=0 0
m=audio 6230 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:10
a=sendrecv
70279mS Sip: License, Valid 1, Available 5, Consumed 0
70281mS SIP Call Tx: 17
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.56.3;branch=z9hG4bKac185493194
From: "3792" <sip:3792@copper-state.net>;tag=1c185489868
To: <sip:6699@192.168.50.23;user=phone>;tag=797be2db6bbc548f
Call-ID: 1854894762012201016425@192.168.56.3
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

70282mS SIP Tx: UDP 192.168.50.23:5060 -> 192.168.56.3:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.56.3;branch=z9hG4bKac185493194
From: "3792" <sip:3792@copper-state.net>;tag=1c185489868
To: <sip:6699@192.168.50.23;user=phone>;tag=797be2db6bbc548f
Call-ID: 1854894762012201016425@192.168.56.3
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

70288mS SIP Call Tx: 17
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.56.3;branch=z9hG4bKac185493194
From: "3792" <sip:3792@copper-state.net>;tag=1c185489868
To: <sip:6699@192.168.50.23;user=phone>;tag=797be2db6bbc548f
Call-ID: 1854894762012201016425@192.168.56.3
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

70288mS SIP Tx: UDP 192.168.50.23:5060 -> 192.168.56.3:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.56.3;branch=z9hG4bKac185493194
From: "3792" <sip:3792@copper-state.net>;tag=1c185489868
To: <sip:6699@192.168.50.23;user=phone>;tag=797be2db6bbc548f
Call-ID: 1854894762012201016425@192.168.56.3
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

70346mS SIP Rx: UDP 192.168.56.3:5060 -> 192.168.50.23:5060
ACK sip:6699@192.168.50.23;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.56.3;branch=z9hG4bKac185493194
Max-Forwards: 70
From: "3792" <sip:3792@copper-state.net>;tag=1c185489868
To: <sip:6699@192.168.50.23;user=phone>;tag=797be2db6bbc548f
Call-ID: 1854894762012201016425@192.168.56.3
CSeq: 1 ACK
Contact: <sip:3792@192.168.56.3:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.6.00A.016.002
Content-Length: 0

70349mS SIP Call Rx: 17
ACK sip:6699@192.168.50.23;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.56.3;branch=z9hG4bKac185493194
Max-Forwards: 70
From: "3792" <sip:3792@copper-state.net>;tag=1c185489868
To: <sip:6699@192.168.50.23;user=phone>;tag=797be2db6bbc548f
Call-ID: 1854894762012201016425@192.168.56.3
CSeq: 1 ACK
Contact: <sip:3792@192.168.56.3:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.6.00A.016.002
Content-Length: 0

--DB

http://support.csctech.us

Remote Support Specialist

 

[DBrewsky]

I got it to work.. Here is the configuration..

SIP Line Number: 17 (or whatever you choose)
ITSP Domain Name: <blank>
ITSP IP Address: <IP address of the Audiocodes Gateway>
Preficx: <blank>
National Prefix: 0 <default>
Country Code: <blank>
International Prefix: 00 <default>
Send Caller ID: P Asserted ID
Registration Required: <unchecked>
In Service: <checked>
Use Tel URI: <checked>
Chock OOS: <unchecked>
Call Routing Method: Request URI
Layer 4 Protocol: UDP
Send Port: 5060
Use Netowrk topology: LAN 1
Listen Port: 5060 <default>

I setup and Incoming Call Route with 6699 (digits dialed from the Audiocodes MP124 that is directed to the IP Office IP address) with a destination of a valid user.

Tested and worked perfectly.

Hope this helps someone else!

--DB

http://support.csctech.us

Remote Support Specialist

 

[intrigrant]

It is better practice to make it act as a SIP phone instead of a SIP line, then the user is registerd to the IPO Gatekeeper. It is more secure as a SIP trunk and the user can transfer calls to any other station.

 

[DBrewsky]

Gotcha, but our end result is connecting this to a full SIP PBX to allow station to station dialing. I will be able to test this tomorrow.

And since our gateway on the SIP PBX is an Audiocodes Mediant 1000, I will be able to setup DIDs to route like SIP trunks from the Mediant to the IPO.

Basically we're doing this to setup DIDs for remote users who want to access features from their cell phones.

--DB

http://support.csctech.us

Remote Support Specialist

 

[tlpeter]

You only need one DDI for options from cell phones.
The money spend on the mediant could have been spend on voimail pro.
That will give features to cell phones.
There is no need for sip trunks for this.
Or do you only have analog trunks?

When you pay peanuts, you get monkeys!

honey, i fried the IP Office !!!

Sarcasm, it's only one of the services I offer.

 

[DBrewsky]

This IPO is a demo kit. We already have Mediant gateways in our office, so I was utlizing that to connect trunks to our IPO. As for VM Pro, how do you mean? We want to use a cell phone for transferring calls via the IPO. It's all for demonstration purposes. Our sales folks were lead to believe we can have mobile IPO users use their cell phones for internal communications and IPO feature usage.

--DB

http://support.csctech.us

Remote Support Specialist

 

[intrigrant]

They are right, we do the same here but not with a Mediant.
I was planning to use a Mediant 600 with BRI myself here but my Mediant was selfdestructing after a few days continous powered up, basically there came fire out of the back of the unit fans.
Is that typical behaviour of the Mediant?
I like a good fire but this was a bit disapointing.

 

[tlpeter]

It was a very small fire
My opinion: burn them all


When you pay peanuts, you get monkeys!

honey, i fried the IP Office !!!

Sarcasm, it's only one of the services I offer.