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I cannot make call between Tandberg C60 using Sip protocol

Answered Question
Aug 4th, 2014
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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  

Correct Answer by Wayne DeNardi about 3 years 2 weeks ago

You can't just dial SIP with an IP address only - that's not a valid SIP format.  It must be an [email protected] or [email protected] (ie [email protected]) format.

In your example above, you should be able to dial "[email protected]" or "[email protected]" - both of these are valid SIP addresses.  The straight IP address 10.88.17.5 is not.

PS - As mentioned by Patrick, you really should upgrade your software on your endpoints to a newer version.  There's a Security Advisory that should assist you getting the new software, even without an active service contract: http://tools.cisco.com/security/center/content/CiscoSecurityAdvisory/cisco-sa-20140605-openssl

Wayne
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Patrick Sparkman Mon, 08/04/2014 - 08:35
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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?

Patrick Sparkman Mon, 08/04/2014 - 14:12
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saheerulladan Tue, 08/05/2014 - 06:12
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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 "[email protected]"  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 Tue, 08/05/2014 - 06:25
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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 Tue, 08/05/2014 - 06:50
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Hi,

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

 

Displaying 1 search for this Search ID
 
Patrick Sparkman Tue, 08/05/2014 - 07:01
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Provide a screen shot of your search rules.

Patrick Sparkman Tue, 08/05/2014 - 09:45
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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 [email protected]?  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?

Patrick Sparkman Tue, 08/05/2014 - 12:40
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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 Tue, 08/05/2014 - 15:45
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Hi , I checked with Cisco Jabber video then call connecting and works well with "[email protected]" 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 Tue, 08/05/2014 - 16:52
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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 [email protected] 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 [email protected] for everything.

saheerulladan Sun, 08/10/2014 - 01:14
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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.

Correct Answer
Wayne DeNardi Tue, 08/05/2014 - 18:56
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You can't just dial SIP with an IP address only - that's not a valid SIP format.  It must be an [email protected] or [email protected] (ie [email protected]) format.

In your example above, you should be able to dial "[email protected]" or "[email protected]" - both of these are valid SIP addresses.  The straight IP address 10.88.17.5 is not.

PS - As mentioned by Patrick, you really should upgrade your software on your endpoints to a newer version.  There's a Security Advisory that should assist you getting the new software, even without an active service contract: http://tools.cisco.com/security/center/content/CiscoSecurityAdvisory/cisco-sa-20140605-openssl

Wayne
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