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I cannot make call between Tandberg C60 using Sip protocol

saheerulladan
Level 1
Level 1

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  

1 Accepted Solution

Accepted Solutions

You can't just dial SIP with an IP address only - that's not a valid SIP format.  It must be an alias@domain or alias@ipaddress (ie alias@10.10.10.10) format.

In your example above, you should be able to dial "std322.10@10.88.17.5" or "std322.11@10.88.17.55" - 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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Please remember to rate responses and to mark your question as answered if appropriate.

Wayne
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Please remember to mark helpful responses and to set your question as answered if appropriate.

View solution in original post

15 Replies 15

Patrick Sparkman
VIP Alumni
VIP Alumni

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?

Hi Patrick,

 

Thanks for your reply.

 

please see the attached .

 

 

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.

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

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?
 

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
 

Provide a screen shot of your search rules.

please find the attached

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?

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.

 

 

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.

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.

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.

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.

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