SIP negotiation fails

Unanswered Question

Good day.

I have troubles with setting up Cisco2821 for outgoing call via ITSP ICH (part of DeltaThree). Unfortunately, ITSP Support Team answered that they provide full support for PAP2/ATA186 devices only and cannot help with examples/troubleshooting for

another Cisco SIP-compliant devices.

It may be misconfiguration on our side, but i'm not familiar with SIP, and cannot find where is it. Can anyone point right way to solve this issue?

VoIP configuration on our side is:

!

dial-peer voice 100 voip

answer-address 12068556332

destination-pattern .T

session protocol sipv2

session target ipv4:213.x.x.140:5060

dtmf-relay rtp-nte

!

!

sip-ua

authentication username xxxxx password xxxx

retry invite 3

retry response 3

retry bye 3

retry cancel 3

retry register 3

retry options 10

timers trying 1000

timers connect 1000

timers register 1000

sip-server ipv4:213.x.x.140:5060

!

"debug ccsip messages" shown SIP negotiation (not full debug output placed for minimize

size):

---------------

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

Received:

SIP/2.0 100 Trying

Received:

SIP/2.0 407 Proxy Authentication Required

Sent:

ACK sip:[email protected]:5060 SIP/2.0

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

Received:

SIP/2.0 100 Trying

(~30 sec waiting, then)

Sent:

CANCEL sip:[email protected]:5060 SIP/2.0

Received:

SIP/2.0 481 Call Leg/Transaction Does Not Exist

------------

I have this problem too.
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Paolo Bevilacqua Mon, 03/12/2007 - 11:02

Actually it would have been better if you had included the whole debug. The ITSP may be indicating a realm and tha needs to be included in the username configuration.

Paolo Bevilacqua Mon, 03/12/2007 - 13:27

Hi,

Looks to me the router is authenticating correctly and the ITSP is at fault by not sending any result in 30 sec after the "trying" message.

The only think I can think is to try to set the calling number to be the same as authentication username, this you do with "clid network-provided" under "dial-peer".

Paolo, thank you for responce.

Unfortunately, it did not help to solve problem - ITSP proxy still not respond after "trying".

Our multiple ATA's working fine with ITSP. As i think the difference between SIP negotiation at ATA and Cisco2821 is that ATA register at ITSP proxy.

The one way to resolve problem, as i think, can be emulate ATA on 2821, but there are some moments that i dont know how to set.

Each ATA have 2 FXS ports, connected to PBX. 2821 connected to PBX via ISDN E1 trunk (work fine, and 2 Asterisk in remote offices works fine too).

Maybe somewhere are examples how to emulate ATA in that case?

Paolo Bevilacqua Thu, 03/15/2007 - 12:05

You can have the 2801 register as well.

Configure a secondary number on an ephone-dn of your choice with no-reg primary

in sip-ua, configure "registrar" and "credentials".

Configure "no-register" on all "dial-peer pots".

And let's see if it goes better.

Yes, configuring ephone-dn with secondary number helps to register number on ITSP (got OK, 200).

But this didnt help to place call... Still got timeout after "trying".

I tried to compare SIP Invite packets from ATA and from 2821. The differencies that i found are:

- from ATA goes user=phone, from 2821 goes user=calling

- some differencies in list of "Allow" requests.

I havent any ideas why it didnt work... :(

----------

Packet from ATA-186:

[0:0]Tx Msg to 213.137.73.140:5060

INVITE sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP 80.249.238.155:5060

From: Line 0 ;tag=2193509254

To:

Call-ID: [email protected]

CSeq: 1 INVITE

Contact: Line 0

User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A)

Expires: 300

Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER

Content-Length: 263

Content-Type: application/sdp

---------------------

Packet from 2821:

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 195.234.72.4:5060;branch=z9hG4bK2AA16E

Remote-Party-ID: ;party=calling;screen=yes;privacy=off

From: ;tag=9B85BB48-94

To:

Date: Fri, 16 Mar 2007 15:49:55 GMT

Call-ID: [email protected]

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 3521911484-3540783579-2156658712-433194112

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1174060195

Contact:

Expires: 300

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 294

Paolo Bevilacqua Sat, 03/17/2007 - 08:02

Sorry, I do not see a clear reason why the calls from the router are not successful. Time to switch to a different ITSP perhaps.

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