SIP error 481

[G3261]

my outbound fax calls to pstn doesn't work...please see call flow below

fax server--SIP trunk--cucm---mgcp gw--pstn

Internal calls are working fine.

Invite-100 trying--183 session in progress--UPDATE--481 call doesn't exist

I would appreciate if someone can get back to me.

***if you see ++ before a number that just a typo

[Nithin Eluvathingal]

Internal calls are working fine. Internal calls from phone to Fax server? What about incoming call from PSTN to FAX ?

From: sip:4031112222@171.20.71.210:5060;tag=BAC7A25B7D10
To: sip:+14032223345@192.168.175.252:5060
Call-ID: 336274AE-4547-4417-B4B6-18B380950ED8@171.20.71.210
CSeq: 101 INVITE

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 171.20.71.210:5060;branch=z9hG4bK-5C2583E84A69
From: sip:4031112222@171.20.71.210:5060;tag=BAC7A25B7D10
To: sip:+14032223345@192.168.175.252:5060;tag=167043904~8a8ec789-a441-442a-a33a-7f83cbca7314-96467356

SIP/2.0 481 Call Does Not Exist
Via: SIP/2.0/UDP 192.168.175.252:5060;branch=z9hG4bKebb25f68e49a17
From: sip:++14032223345@192.168.175.252:5060;tag=167043904~8a8ec789-a441-442a-a33a-7f83cbca7314-96467356 >> header is in other direction. 
To: sip:4031112222@171.20.71.210:5060;tag=BAC7A25B7D10

[G3261]

Internal Calls----one eFax to this eFax server

Incoming from PSTN to FAX---works

[G3261]

192.168.175.252--CUCM call manager

171.20.71.210---fax server (this ip is also configured on CUCM SIP Trunk) as a destination address port 5060

fax from 4031112222

fax to 14032223345

2nd cucm is -192.168.175.220

I hope that helps.

[Nithin Eluvathingal]

Since only outgoing Fax are not working, Does the fax sever trunk has access to CSS which can call outside numbers ?

What protocol you use for fax,   t.38 or Pass through ?

[G3261]

yes...it does.

SIP Trunk CSS (Inbound calls) contains PT for route pattern

[Nithin Eluvathingal]

The default configuration for Unified CM SIP trunks is to use Delayed Offer, if your Fax use early offer you need to make your sip trunk for early offer.

Check MTP required on the SIP trunk configuration page or if you are on CUCM 8.5 or later go to the SIP Profile (Device > Device Settings > SIP Profile) and check "Early Offer support for voice and video calls (insert MTP if needed)".

[G3261]

will do....let you know little later this week. Thanks