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Community Manager

Ask the Cisco VIP: Troubleshooting SIP in Cisco Unified communications


Troubleshooting SIP in Cisco Unified communications deployments with Cisco VIP Ayodeji Okanlawon

This is a Q&A Ask the Expert Session continuation from the Live Webcast

Ask your questions on Session Initiation Protocol (SIP) and how it is redefining our UC world.The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.Okanlawon Cisco VIP

Featured Expert

Ayodeji Okanlawon, a Cisco Designated VIP, is the Lead Consultant Engineer for Global Solutions Design and Engineering at Verizon Business. In his past, he has worked at Intact IS, NCS Global, and Schlumberger Information Solutions. His experience includes development of design and deployment of large scale IP telephony projects on Cisco Call Manager platforms, Cisco Voice gateways, Cisco Jabber cloud and on premise solution. His expertise includes SIP solutions, CUBE design and Deployment, Troubleshooting: Voice gateways, CUCM, Unity connection, CUPS. Deji has been awarded the Cisco Designated VIP in 2013 and 2014. Deji holds a Bachelor of Science (BS), Electrical and Electronics Engineering, Second Class Upper from Obafemi Awolowo University.  

According to Deji, “If you want to advance your career, if you’re serious about your skill sets, you’ve got to be in the forums.”  (Read the Interview >>)


We look forward to your participation. This event is open to all, including partners.

* * Remember to use the rating system to let Deji know if you have received an adequate response. * *

Deji might not be able to answer each question due to the high volume expected during this event. This event runs January 13 through January 23, 2015.  Visit this forum often to view responses to your questions and the questions of other community members.

1 ACCEPTED SOLUTION

Accepted Solutions
VIP Mentor

Hi Yannick,You can use OOD

Hi Yannick,

You can use OOD OPTIONS ping for this.

Out-of-dialog OPTIONS message sent to check the status of the SIP Trunk

  • The dial-peer is “busyout” if it does not receive a response within a configurable time period
  •  For an INVITE that matches a “busyout” dial-peer, CUBE sends “503 Service Unavailable”
  •  If there is a secondary dial-peer econfigured, the call will be re-routed the secondary path

You should configure this on the dial-peer facing your ITSP. Example is shown below:

dial-peer voice 100 voip
voice-class sip options-keepalive up-interval 20 down-interval 20 retry 3

 

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32 REPLIES 32

Dear, Is there a method of

Is there a method of letting the CUCM know that the CUBE has lost it's connection towards the SIP provider. We want to be implementing a small BRI for ISDN backup to a CUBE.

My suggestion is that we register that BRI card with MGCP to call manager but then how can CUCM know that the sip trunk is down (provider side) as his connection to CUCM will still be up & running.

Is there a sort of keepalive that CUCM can send towards the provider?

Is there a deployment document written over

Kr,

Yannick Vranckx

VIP Mentor

Hi Yannick,You can use OOD

Hi Yannick,

You can use OOD OPTIONS ping for this.

Out-of-dialog OPTIONS message sent to check the status of the SIP Trunk

  • The dial-peer is “busyout” if it does not receive a response within a configurable time period
  •  For an INVITE that matches a “busyout” dial-peer, CUBE sends “503 Service Unavailable”
  •  If there is a secondary dial-peer econfigured, the call will be re-routed the secondary path

You should configure this on the dial-peer facing your ITSP. Example is shown below:

dial-peer voice 100 voip
voice-class sip options-keepalive up-interval 20 down-interval 20 retry 3

 

Please rate all useful posts
Hall of Fame Community Legend

Hey Deji, Nice work on this

Hey Deji,

 

Nice work on this my friend! Hope all is well with you :)

 

Cheers!

Rob

Hello Ayodeij, Thanks for the

Hello Ayodeij,

Thanks for the answer, i don't have to set anything specific on the trunk towards the CUBE on Call Manager right?

Just set up the OOD ping option on the CUBE and if i then have another gateway in the RL CUCM should know when it's down

Kind Regards,

VIP Mentor

You can enable options ping

You can enable options ping on the sip profile associated with the sip trunk pointing to the cube. With this cucm will know when the trunk to CUBE is down and will reroute the call to the other gateway in the RL

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Frequent Contributor

Hello Deji..  I will be

Hello Deji..  I will be posting questions from the Webcast that we were not able to get to..

 

(derrick shum) Q: is there a way to insert call id Name in the SIP header on outbound calls

VIP Mentor

Yes you can insert call id

Yes you can insert call id name (CNAM) in a SIP header. This is usually found in the From header.

E.g.

From: "Inspector Clouseau" <sip:011270059523@192.11.13.5>;tag=301774~23001403-1e76-4ea5-8dd4-52d6f82b946c-27747658

NB: you will need to configure the display and ascii display on the DN in CUCM for this to be present in the From header which identifies the display name the user wishes to present to other user.

However it is not enough to pass this to the ITSP, your ITSP must offer this service for it to work. The name must be registered to a valid number within their Servers for this to work.

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Frequent Contributor

Q: do you have a recommand

(derrick shum) Q: do you have a recommand tool for troubleshooting SIP call flow in the gateway

VIP Mentor

Derrick,Thank you for joining

Derrick,

Thank you for joining us for the webcast

There are a few tools that you can use. Here are my favourites.

1. TranslatorX (http://translatorx.cisco.com/).. --You can do a whole lot with it.

2. Notepad++ (http://notepad-plus-plus.org/)        ----Helps you to analyse your logs, debugs etc

3. Agent Ransack (http://www.mythicsoft.com/agentransack) --- A powerful search tool to help you find calls within your log files (especially with CUCM logs)

 

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Frequent Contributor

(Björn Engel) Q: I

(Björn Engel) Q: I implemented a SIP Trunk to my gateway- But everytime i start an AD Hoc Conference with external ISDN Participants via CUCM Software Conference Bridge, the first external Participant drops. SDP shows only g711u and the scenario works perfectly with H323.   Any Hints on where to search for what ISDN GW I have in use?

VIP Mentor

Bjorn,Thank you for taking

Bjorn,

Thank you for taking time to join us for the webcast

I will need to look at your logs to understand in detail what is going on.

Can you please send the following:

sh run

debug ccsip message.

CUCM traces

Please including the calling and called number.

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Frequent Contributor

(Mike Amb) Q: One question I

(Mike Amb) Q: One question I have is FAX support under SIP.  If On-Ramp is supported, or if you need to move to a dedicated SIP/Fax solution (rightfax etc)

Frequent Contributor

(Deval Tetar)  Q: Can you

(Deval Tetar)  Q: Can you cover some info on SIP trace study from CUCM

VIP Mentor

Hi Deval,Thank you for taking

Hi Deval,

Thank you for taking time to join us for the webcast.

I have already covered this in the document below. Please have a read and let me know if you have more questions.

https://supportforums.cisco.com/document/113271/understanding-sip-traces

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