SIP trunk no longer working in R6 1

[DennyHeack]

Hello,

Situation:
IP500v2 6.0(8)
SIP trunk Uconnect (netherlands)
This trunk does not require any authorization credentials.
It is connected directly to LAN2(WAN)

Before, (R5 and lower) this trunk worked instantly.
As of R6 i can only place outbound calls. Inbound calls trigger no system monitor events. In R5 the polling (monitor trace below) did not occur. I've been at it for 2 days now and am out of ideas. Any one?



709172mS SIP Reg/Opt Tx: 17
OPTIONS sip:Unknown@82.148.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 90.145.xxx.xxx:5060;rport;branch=z9hG4bK41a2c207cd72f292effc568e4d068347
From: <sip:Unknown@90.145.xxx.xxx>;tag=bf63defcb7bb738a
To: <sip:Unknown@82.148.xxx.xxx>
Call-ID: 503446c333f1b634328d3458b6308bce@90.145.55.20
CSeq: 1279667066 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

709173mS SIP Tx: UDP 90.145.xxx.xxx:5060 -> 82.148.xxx.xxx:5060
OPTIONS sip:Unknown@82.148.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 90.145.xxx.xxx:5060;rport;branch=z9hG4bK41a2c207cd72f292effc568e4d068347
From: <sip:Unknown@90.145.xxx.xxx>;tag=bf63defcb7bb738a
To: <sip:Unknown@82.148.xxx.xxx>
Call-ID: 503446c333f1b634328d3458b6308bce@90.145.xxx.xxx
CSeq: 1279667066 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

709196mS SIP Rx: UDP 82.148.xxx.xxx:5060 -> 90.145.xxx.xxx:5060
SIP/2.0 406 Method not acceptable
Via: SIP/2.0/UDP 90.145.xxx.xxx:5060;rport=5060;branch=z9hG4bK41a2c207cd72f292effc568e4d068347
From: <sip:Unknown@90.145.xxx.xxx>;tag=bf63defcb7bb738a
To: <sip:Unknown@82.148.xxx.xxx>;tag=fe398036172e03a294d29679da00e1f2.28bb
Call-ID: 503446c333f1b634328d3458b6308bce@90.145.xxx.xxx
CSeq: 1279667066 OPTIONS
Server: OpenSIPS XS 1.4.5
Content-Length: 0

709198mS SIP Reg/Opt Rx: 17
SIP/2.0 406 Method not acceptable
Via: SIP/2.0/UDP 90.145.xxx.xxx:5060;rport=5060;branch=z9hG4bK41a2c207cd72f292effc568e4d068347
From: <sip:Unknown@90.145.xxx.xxx>;tag=bf63defcb7bb738a
To: <sip:Unknown@82.148.xxx.xxx>;tag=fe398036172e03a294d29679da00e1f2.28bb
Call-ID: 503446c333f1b634328d3458b6308bce@90.145.xxx.xxx
CSeq: 1279667066 OPTIONS
Server: OpenSIPS XS 1.4.5
Content-Length: 0

 

[hairlessupportmonkey]

i would make a backup of your config.
delete your trunk
re-create it

then post some more traces after that.

Maybe the upgrade messed up the trunk config.

 

[TheSmash]

You need to make sure that the INVITE and TO fields match. If they don't you need to modify the settings on the SIP trunk.

See Call Routing Method on the SIP Trunk configuration.

That should sort you.

My name is Mike but everyone calls me The Smash...

 

[DennyHeack]

I already deleted the config en recreated the siptrunk.

I have a ipo500 v1 & ipo500 v2. Same trunk, same config (except for the new options ofcourse) v1 works, v2 doesnt.

@TheSmash
I'll give it a try...

 

[TheSmash]

There is a change in the SIP handling for incoming calls between V5 and V6.

I didn't ask you to reconfigure it, but merely to check the invite packet. Does the INVITE line match the TO Line?

If not, then adjust the Call Routing Method on the SIP trunk.

My name is Mike but everyone calls me The Smash...

 

[intrigrant]

We can help you bust that cost some money and we are in the netherlands.
There are only three distris in NL and we are the only one with a real helpdesk.
I think to know what is wrong but i have to make a living as well so only paid support for the Benelux, sorry.

 

[DennyHeack]

Call routing on sip tab is for checking against incoming call route. On inbound calls there are no sip messages in monitor. There is no sip traffic coming at me!

I think it's like this:

IPO sends out some OPTION poll (trace@topicstart) which is rejected by ITSP. This poll is not affected by the new OOS setting on sip tab(!). ITSP doesnt send calls to ipo because of this poll error. I have a working 500v1 and a Cisco CME which dont send this 'poll'. How do I alter or disable this 'poll'?

 

[TheSmash]

Have a look at this example... (Sorry intrigrant if I am doing you out of a sale!)

SIP Rx: UDP 193.203.210.38:5060 -> 192.168.42.85:5060
INVITE sip:901526@192.168.42.85:5060;transport=UDP SIP/2.0 <-- This is the Request URI, number being presented is 901526
Via: SIP/2.0/UDP 193.203.210.38:5060;branch=z9hG4bK-sYn-0-9bff6b128daf
Max-Forwards: 8
Contact: <sip:193.203.210.38:5060>
From: <sip:01707299900@voip.co.uk;user=dialstring>;tag=25dd0408cd
To: <sip:01133228628@voip.co.uk;user=dialstring> <-- This is the To field, number being presented is 01133228628
Call-ID: 43d7c106-0bf9-11df-a0a2-3993852fea92
CSeq: 1726121442 INVITE
Expires: 120
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL
Content-Type: application/sdp
Content-Length: 556

My name is Mike but everyone calls me The Smash...

 

[DennyHeack]

That's odd. Just asked my distri with support desk yesterday. Couldnt help me.
Assuming you're correct, I spoke 2 u or one of your colleges.

I can't see an INVITE an TO header in the option 'poll'

Here's a sip outbound call:

3052528mS SIP Call Tx: 17
INVITE sip:010xxxxxx@82.148.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 90.145.xxx.xxx:5060;rport;branch=z9hG4bKd3f3a47162a78214dcf031582b116b6c
From: "TestGebruiker2" <sip:313671xxxxx@90.145.xxx.xxx>;tag=e06692f60895b813
To: <sip:010xxxxxx@82.148.xxx.xxx>
Call-ID: 084273758abbfe07be78ca8fa38e6d54@90.145.xxx.xxx
CSeq: 1575178306 INVITE
Contact: "TestGebruiker2" <sip:902@90.145.xxx.xxx:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Content-Length: 273

v=0
o=UserA 3502507665 88005349 IN IP4 90.145.xxx.xxx
s=Session SDP
c=IN IP4 90.145.xxx.xxx
t=0 0
m=audio 49154 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

3052577mS SIP Rx: UDP 82.148.xxx.xxx:5060 -> 90.145.xxx.xxx:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 90.145.xxx.xxx:5060;rport=5060;branch=z9hG4bKd3f3a47162a78214dcf031582b116b6c
From: "TestGebruiker2" <sip:31367xxxxx@90.145.xxx.xxx>;tag=e06692f60895b813
To: <sip:010xxxxxx@82.148.xxx.xxx>
Call-ID: 084273758abbfe07be78ca8fa38e6d54@90.145.xxx.xxx
CSeq: 1575178306 INVITE
Server: OpenSIPS XS 1.4.5
Content-Length: 0

 

[TheSmash]

I have just re-read your post and you are seeing no Sysmon info for incoming. This suggests that the problem is not with the phone system at all. I'd advise looking at your routing and Firewall.

My name is Mike but everyone calls me The Smash...

 

[DennyHeack]

WAN port is directly connected to bridgerouter. No firewalls in between.
Firewall setting on lan2 is on 'Open Internet' (suggested by distri helpdesk)

I can ping itsp, dns servers and everything else from Lan1&Lan2.
Other ITSP provider (with registration/credentials) works fine.

I've opened up a case with ITSP. Hopefully they'll educate me some more....

 

[TheSmash]

It sounds like your ITSP is not sending the INVITE's to the correct IP Address...

My name is Mike but everyone calls me The Smash...

 

[DennyHeack]

ITSP didnt change anything.
7 seconds after i hung up everything "automagically"
starts working

Still dont understand why R5 worked correctly and R6 didnt.
Oh well...

Thanx everyone!

 

[IPOfficeIreland]

The to invite header is determined in the shortcode

N"@whatever_you_want