I cannot make call between Tandberg C60 using Sip protocol

[saheerulladan]

Hi There,

I tried to make call between Tandberg C60 codecs  with both ends SIP , unfortunately the call couldn't connect. Its not showing the "connecting" message either.

Then I changed in the Codec c60 (both sides c60) protocol to H323 , then call connects and works well.

Please any idea.

Thanks.

Regards,

Saheer

==========

VCS ver: X7.2.1

TANDBERG Codec C60;

Codec1: TC4.2.4.296355

Codec2:TC4.2.4.296355  

[Patrick Sparkman]

First off, have you thought about upgrading your codecs, as there are some very notable issues with the older versions of TC software running OpenSSL.  This might also fix your issue too.

In regards to your issue, can we see a search history from your VCS of a failed SIP call?

[saheerulladan]

Hi Patrick,

Thanks for your reply.

please see the attached .

[Patrick Sparkman]

Why don't you dial the other endpoint's SIP address "alias@domain", instead of IP address?

Where are each of the two endpoints registered, ex: Control or Expressway?

Do you have the other part of the search history, since it goes over the traversal link to the Expressway.

[saheerulladan]

We installed this codecs in meeting rooms which are equipped in Audio Visual rack with AMX controller and touch panel and so on. We us AMX touch panel to "dial". when i dialled with "alias@domain"  it shows nothing. so i dialled with ip then i can see the messaged "connecting". after 1 or 2 minutes it got disconnected.

Endpoints are registered in control. i.e. Cisco Telepresence Video Communication Server Control.

Thanks

[Patrick Sparkman]

Try upgrading the codecs and see what results you get, after all they're outdated, since the newest software for them is TC7.x.

When you try to do a SIP call, do you see anything on your VCS, either call history or search history of the call attempt?
 

[saheerulladan]

Hi,

see the below , search history for the call i made;

Displaying 1 search for this Search ID

Search (252) State: Completed Found: False Reason: Temporarily Not Available Type: SIP (INVITE) CallSerial Number: 7813a438-1ca4-11e4-8fec-0010f324928a Tag: 7813a5f0-1ca4-11e4-b51a-0010f324928a Source (1) Authenticated: True Aliases (1) Alias (1) Type: Url Origin: Unknown Value: Std322.11@ads.pnu.edu.sa Zone (1) Name: DefaultSubZone Type: Local Path (1) Hop (1) Address: 10.88.17.55:5061 Destination (1) Alias (1) Type: Url Origin: Unknown Value: sip:10.88.17.5 StartTime: 2014-08-05 16:29:04 Duration: 84.57 SubSearch (1) Type: Transforms Action: Not Transformed ResultAlias (1) Type: Url Origin: Unknown Value: 10.88.17.5 SubSearch (1) Type: Admin Policy Action: Proxy ResultAlias (1) Type: Url Origin: Unknown Value: 10.88.17.5 SubSearch (1) Type: FindMe Action: Proxy ResultAlias (1) Type: Url Origin: Unknown Value: 10.88.17.5 SubSearch (1) Type: Search Rules SearchRule (1) Name: LocalZoneMatch_AnyIP Zone (1) Name: LocalZone Type: Local Protocol: SIP Found: False Reason: Not Found StartTime: 2014-08-05 16:29:04 Duration: 0.01 Gatekeeper (1) Address: 172.17.17.1:0 Alias (1) Type: Url Origin: Unknown Value: 10.88.17.5 Zone (2) Name: LocalZone Type: Local Protocol: H323 Found: False Reason: Not Found StartTime: 2014-08-05 16:29:04 Duration: 0 Gatekeeper (1) Address: 172.17.17.1:0 Alias (1) Type: Url Origin: Unknown Value: 10.88.17.5 SearchRule (2) Name: VCSE_AnyIP Zone (1) Name: VCSE Type: TraversalClient Protocol: SIP Found: False Reason: Temporarily Not Available StartTime: 2014-08-05 16:29:04 Duration: 42.31 Gatekeeper (1) Address: 172.27.1.1:7001 Alias (1) Type: Url Origin: Unknown Value: 10.88.17.5 Zone (2) Name: VCSE Type: TraversalClient Protocol: H323 Found: True StartTime: 2014-08-05 16:29:46 Duration: 0.02 Gatekeeper (1) Address: 172.27.1.1:6001 Alias (1) Type: Url Origin: Unknown Value: 10.88.17.5

 

[Patrick Sparkman]

Provide a screen shot of your search rules.

[saheerulladan]

please find the attached

[Patrick Sparkman]

I see almost all of your search rules are regex, can you try to add the following with a priority of 1 and see what happens?

Protocle:  Any
Source:  Any
Mode:  Any
Pattern String:  Any
Target:  LocalZone

That will match anything alias that would be registered to your VCS.

You keep providing search rules from when you're trying to dial an IP via SIP, but what if you dial using alias@domain?  I know you said it doesn't show connecting on the screen, but does the VCS show any result of a search or call attempt?

Also, have you been able to upgrade the two endpoints and see if that helps or has a different behavior?

[saheerulladan]

I cannot see any call log or search history when dial "alias@domain".

please the attachment , about the search rule. can you advise what i mention in all the columns when i create new search rule. coz i cannot see anything pattern there.

[Patrick Sparkman]

Sorry, the pattern is for regex, either case if you're not seeing any kind of history of the call on the VCS, it sounds like the call isn't even getting to the VCS, so it wouldn't matter to create a new search rule to try.  Double check the codec's SIP registration to the VCS, and it's configuration.  If you upload the codec's logs and a copy of the status and config, might be able to take a look on the codec's side.

[saheerulladan]

Hi , I checked with Cisco Jabber video then call connecting and works well with "alias@domain" through sip to sip

is it possible to dial sip with "ip" only? any limitation on VCS?

It may take time to upgrade firmware.

please see the attached search rule for "any ip " already configured. do i need to make any changes for sip to sip dial in VCS?

both end points are registered with VCS , please see the attached pic.

:)

Tnx.

[Patrick Sparkman]

As long as you have the appropriate search rules in place that will match the SIP URI, you should be fine on the VCS.  One thing I though of is the VCS has a locate feature that you can test your search rules, that can help determine if your VCS is configured correctly.  I'm not logged into a VCS, but it's under Maintenance > Tools > Locate, or something of the like.  If the search works with SIP to SIP with alias@domain that it's probably on the endpoint end.

Regarding IP dialing with SIP, it's possible, though we don't do it since we use alias@domain for everything.

[saheerulladan]

Hi All,

I put system unit name same as sip URI , now i can call sip to sip with URI. 

Thanks for your valuable answers.