Ask the Experts: Configuring and Troubleshooting Cisco SIP CUBE/Gateways and MGCP Gateways

[ciscomoderator]

With Edson Pineiro

Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn and ask questions about configuring and troubleshooting Cisco session initiation protocol (SIP) Cisco Unified Border Element (CUBE)/Gateways and Media Gateway Control Protocol (MGCP) Gateways with Cisco expert Edson Pineiro. Learn about the different types of  various aspects of call signaling methods, requests, response SIP exchange and behaviors and why secure device provisioning (SDP) is important. You'll also learn about common issues when troubleshooting call manager MGCP registration, digital signal processors, foreign exchange subscriber interface (FXS), foreign exchange office (FXO), Integrated Services Digital Network (ISDN) Primary Rate Interface (PRI), integration with service providers and Q signaling (QSIG) Private Branch Exchanges (PBX).

Edson Pineiro is a senior customer support engineer in the Cisco Technical Assistance Center in Sydney. His current role includes configuring, troubleshooting, and designing gateways, gatekeepers, Cisco Unified Border Element Enterprise Edition, and Cisco Unified Call Manager using his deep knowledge of signaling protocols such as SIP, H.323, MGCP, SKINNY, and others. He has been involved in several bug fixes, escalations, and critical account cases from around the globe. He has over seven years of experience in the IP voice industry.

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

Edson might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the Collaboration Voice, & Video IP Telephony subcommunity shortly after the event.This event lasts through Friday April 19, 2013. Visit this forum often to view responses to your questions and the questions of other Cisco Support Community members.

[Edson Pineiro]

Hello Muhammad,

   Thank you for your question.

   With regards to IP Routing and IP addressing, your questions is outside the scope of this particular forum. Can you please post your question in the Network Infrastructure Community forum to get a final answer.

Network Infrastructure Community:

https://supportforums.cisco.com/community/netpro/network-infrastructure

   I hope you get your question answered soon.

PS: have you tried using a sub-interface or a loopback? It might be worth a quick search…

Thank you

&

Regards

Edson Pineiro

[timothy.bailey]

Edson

What would the correct configuration be to get the CUBE to respond to another SBC when it sends the CUBE an OPTIONS-PING message?

At present it is rejecting the message saying it is not recognized?

[Edson Pineiro]

Hello Tim Baily,

   Thank you for your question. With regards to Options Ping, from my understanding the CUBE should response with a SIP 200-ok when receiving the Options Ping. According to the following document the feature was introduced in IOS 15.0(1)M, so please ensure you have the correct version. With regards to the configuration to enable options ping on a dial-peer and monitor the device behind the session target the command is "voice-class sip options-keepalive". However the CUBE should respond to the Options with a 200-ok given you have configured a dial-peer with session protocol sip.

Cisco UBE Support for generating Out-Of-dialog SIP Options Ming Message to Monitor SIP:

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb_9321.html

Voice Command Reference

voice-class sip options-keepalive:

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_v2.html#wp1272954

Thank you

&

Regards,

Edson Pineiro

CISCO

[Rob Huffman]

Hi Edson,

I just wanted to take a moment to say thanks for all the great support you've

shown in this "Ask the Expert" event (+5) These wonderfully thoughtful and thorough

answers really show your expertise in this field! Great stuff my friend!

Cheers!

Rob

"Talk about a dream
Try to make it real" 

- Springsteen

[goldnetps]

Hi Edson,

I wonder if you can make the configuration of SIP-UA with 2 carriers SIP, if so what setup procedure I do.

Tks.

[Edson Pineiro]

Hello Goldnetps:

    Thank you for your questions. According to the following document it is possible to support multiple sip registrars with multiple outbound authentications. You can register to up to 6 different registrars servers with different credentials. Each credential used to register with each server will be identified by different realms. For example if the credentials for user1 password password1 had a realm configured with cisco1, when the router sends an outgoing register message to all 6 registrar servers, each server will respond with different realms. User1 would attempt to register with the registrar server that returned the realm cisco1.

   The above registration example can also be applied to the authentication request. In this case you can configure multiple authentication usernames and passwords. When the outbound call is made and the authentication server returned a 401 authentication required it will include a realm. The realm included in the 401 challenge will need to match the authenticated user name and password. After the IOS matches the realm to a specified username and password a RE-INVITE is sent back to the SIP endpoint including the username, password and the realm.

   Basically to both register and authenticate a number or a call, the username or credentials need to match the carriers realm. For incoming calls you need to register and for outgoing call you need to authenticate.

   Please find the following authentication, registrars and credential examples:

!

!

dial-peer voice 1 pots

authentication username potsuser1 password potspassword1 realm pots1

authentication username potsuser2 password potspassword2 realm pots2

!

!

sip-ua

credentials number 2222 username user1 password cisco1 realm cisco1

credentials number 2222 username user2 password cisco2 realm cisco2

credentials number 2222 username user3 password cisco3 realm cisco3

authentication username user1 password cisco1 realm cisco1

authentication username user2 password cisco2 realm cisco2

authentication username user3 password cisco3 realm cisco3

registrar 1 dns:cisco1.com expires 300

registrar 2 dns:cisco2.com expires 300

registrar 3 dns:cisco3.com expires 300

!

!

   For further details regarding the above examples, please review the following document:

Configuring Multiple Registrars on SIP Trunks:

http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-multi-registrars.html

   If you have any further questions please let me know.

Thank you

&

Regards,

Edson Pineiro

CISCO

[jsteinberg]

How I can prevent digit manipulation in SIP/IOS from being passed back from the gateway to CUCM?   I have a standard configuration, with CUCM/IP phones and IOS/SIP gateway to PSTN.  I want to maintain 10 digits on the IP phone screens at all times, to ensure the Jabber software can correctly lookup the 10 digit phone number in the directory.

In some cases, we will need to send 7 digits to PSTN and I will configure this manipulation in the IOS gateway, as shown below:

dial-peer voice 1000

destination-pattern +1..........

forward-digits 7

port 0/0/0:23

With the above configuration, on my IP phone screen I will initially see the called number display in full e164 format.  However once the IOS processes the digit maniuplation, the phone screen updates and I see the 7 digit number when I am using SIP between CUCM and the VG.    If I switch to H323, I can use the command below and it prevents the IOS/H323 from sending the updated 7 digit number to CUCM.

voice service voip

no supplementary-service h224-notify cid-update

However, I can not find a similar command with SIP.

I have seen some examples of using sip-profiles, but I can't figure out how to use a sip-profile since I don't think I can store the 3 digit area code that I am stripping off with the 'forward-digits' command.

[Edson Pineiro]

Hello Jsteinberg,

   Thank you for your question. With regards to the called party number update, from my recollection this is performed using a SIP Update request on an ISDN SIP Gateway.

   The h323 command 'no h225-notify cid-update' is used to remove caller id updates by disabling the h225 notify message on call forwards and transferred calls. This feature is similar to the SIP command "no update caller-id" under voice sevice voip, sip or the removal of the remote party id using "no remote-party-id" configured under the sip-ua for transferred calls. However the command chosen will depend on the type of call your making when the issue occurs. If either or both these methods fail to remove the called party number update on Jabber, then we can simply apply the following sip profile to remove the advertised UPDATE header from the supported field during both request and response sip exchange.

   For your convenience please see the following command references:

Disable update caller-id:

!

voice service voip

  sip

   no update caller-id

!

!

Remove the remote party id header:

!

sip-ua

  no remote-party-id

!

!

   If the above two options fail, and it may depending on your call flow topology please apply the following sip profile:

Remove UPDATE support from both the request and response events:

!

!

voice class sip-profiles 1

  request ANY sip-header Allow-Header modify "UPDATE, " ""

  response ANY sip-header Allow-Header modify "UPDATE, " ""

!

voice service voip

  sip

    sip-profile 1

!

!

   For further details regarding the above configuration changes please review the following documents:

Unified Border Element (CUBE) Session Initiation Protocol (SIP) Normalisation with SIP Profiles Configuration Example:

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a0080982499.shtml

supplementary-service h225-notify cid-update (voice service voip):

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_s12.html#wp1148789

remote-party-id:    

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_r1.html#wp1548923

   Please keep me updated with your progress and if the issue continues please provide the following debugs and output:

!

debug isdn q931

debug ccsip messages

debug voip ccapi

!

&

!

show run

!

   If you have any further questions or concerns please let me know.

Thank you

&

Regards,

Edson Pineiro

CISCO

[Ayodeji Okanlawon]

Jsteinberg,

In addition to Edson's answer, the sip proflie below is anothe r option you can use. The advantage of this is that you wont have to remove your remote party id completely....

voice class sip-profiles 1

  response 183 sip-header Remote-Party-ID modify "<>" "<>"

response 200 sip-header Remote-Party-ID modify "<>" "<>"

You need to replace the 177.1.254.1 with the ip address of your gateway...

With this profile the full 11 digit will be maintained at all times

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[goldnetps]

ok, thanks for the explanation!

But in the example that was set, the credentials are the same, what changes is only username and password. '

I need to make configuration of 2 SIP accounts with authentication credentials are different.

This process SIP autenthication is possible to make the CUCM? SIP Proxy ?

Tks!!

[Edson Pineiro]

Hello Goldnetps,

   Thank you for the follow up questions.

   Please be advised that my examples were for sample purposes only. A SIP Gateway or CUBE can authenticate and register with different usernames and passwords. It is possible to have two or more separate accounts, one for authentication and the others for registration.

   My understanding of your second question is, can a CUCM process a SIP authentication request? To answer your question, a CUCM can authenticate with other SIP endpoints. For example when the SIP trunks initiates an outbound INVITE and the terminating sip end point returns a 401 to authenticate, the CUCM will re-send the INVITE including a username password to authenticate. I also previously answered the same question within this forum.

   For further details regarding SIP authentication on CUCM please visit the following document:

Cisco Unified Communications Manager Security Guide

Configuring Digest Authentication for the SIP Trunk:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/security/8_0_1/secugd/secrealm.html#wp1028144

   With regards to the 3rd questions, is CUCM a SIP Proxy? Please be advised that CUCM is not a SIP proxy. SIP proxy functionality however can be found using either a Cisco Unified SIP Proxy (CUSP) or a Cisco Unified Presence Server (CUPS). Both these devices have built-in SIP Proxy engines.

   For further details regarding CUSP and CUPS please reference the following documents:

Cisco Unified SIP Proxy:

http://www.cisco.com/en/US/prod/collateral/modules/ps2797/data_sheet_c78-521390_ps2797_Products_Data_Sheet.html

GUI Administration Guide for Cisco Unified SIP Proxy Release 8.5:

http://www.cisco.com/en/US/docs/voice_ip_comm/cusp/rel8_5/OLH/gui_config_olh.html

CLI Configuration Guide for Cisco Unified SIP Proxy Release 8.5:

http://www.cisco.com/en/US/docs/voice_ip_comm/cusp/rel8_5/cli_config/cusp_cli_config.html

Cisco Unified Presence (Data Sheet):

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps6837/data_sheet_c78-646724.html

Configuring Proxy Server Settings in Cisco Unified Presence Administration:

http://www.cisco.com/en/US/docs/voice_ip_comm/cups/7_0/english/config_maintenance/administration/guide/b03pcs.html

   If you have any further questions please let me know.

Thank you

&

Regards,

Edson Pineiro

CISCO

[baker.frank]

I have a 7962 phone that can not access VOICEMAIL. it keep asking for a 5 digit pin number. How can I troubleshoot this.

baker.frank@yahoo.com

[Edson Pineiro]

Hello Frank,

   Thank you for your question. Please be advised that voicemail is a topic outside of the scope of this forum. However to troubleshoot your voicemail issue please post your question in the Collaboration, Voice and Video community section IP Telephony.

   Please find the community on the following link:

Collaboration, Voice and Video

IP Telephony: 

https://supportforums.cisco.com/community/netpro/collaboration-voice-video

   If you have any further questions please let me know.

Thank you

&

Regards,

Edson Pineiro

CISCO