SRST IP Phone registration issue

[fuhrersk8]

                   Hello guys!

Currently I'm configuring SRST on a Cisco 2921 in conjunction with a CUCM8.6 (BE6000). When testing SRST, the phones show "MODE SRST", some pick up dial tone, some not, and on phones with dialtone phone calls did not progress. On the Cisco 2921 the debug ccapi all did not show nothing when attempting calls during the SRST test. It appears to me like phones were unable to register with the Voice Gateway (Cisco 2921). Below is the partial SRST config of Voice Gateway. My questions are the following:

1. Does it is neccesary to configure manually the ephones and directory numbers in the Cisco 2921 or phones will be allocated by the router automatically?

2. How to determine the SRST versipon currently running on Cisco 2921? Phones are 7940, 7960 and 7912.

3. On dialpeers (both voip and pots) it is necesary to enable the service mgcpapp in order to permit SRST calls to flow through the dialpeer?

4. Please reviw the SRST config below and if I'm missing something let me know!

Thanks in advanced guys, appreciate your help.

Partial SRST configuration:

ccm-manager fallback-mgcp

ccm-manager mgcp

no ccm-manager fax protocol cisco

ccm-manager music-on-hold

ccm-manager config server 10.200.54.3

ccm-manager config

!

mgcp

mgcp call-agent 10.200.54.3 2427 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

mgcp rtp unreachable timeout 1000 action notify

mgcp modem passthrough voip mode nse

mgcp ip qos dscp af31 media

mgcp package-capability rtp-package

mgcp package-capability sst-package

mgcp package-capability pre-package

no mgcp package-capability res-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

no mgcp explicit hookstate

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface GigabitEthernet0/0.54

mgcp bind media source-interface GigabitEthernet0/0.54

!

mgcp profile default

!

gatekeeper

shutdown

!

call-manager-fallback

secondary-dialtone 9

max-conferences 8 gain -6

transfer-system full-consult

ip source-address 10.200.54.1 port 2000

max-ephones 25

max-dn 25

system message primary Operating in mode SRST

keepalive 10

default-destination 4123

moh music-on-hold.au

multicast moh 239.1.1.1 port 16384 route 10.200.54.3 10.32.84.54

!

[Chris Deren]

What is the IOS version?

What type of phones are these? Are these SCCP or SIP phones?

How many phones are you expcting to register, you are set for only 25 max phones?

What are the phones showing on the display, are they showing the extension?

Are calls working for any phones? Outbound? Inbound?

Is SRST reference configured in CUCM?

Chris

[fuhrersk8]

Hello Chris and thanks for your reply!

     Answers are as follows:

What is the IOS version? --> Version 15.1(4)M5

What type of phones are these? Are these SCCP or SIP phones? --> SCCP

How many phones are you expcting to register, you are set for only 25 max phones? --> less than 25

What are the phones showing on the display, are they showing the extension? --> Operating in SRST Mode and yes the extensions are showing.

Are calls working for any phones? Outbound? Inbound? --> Yes, calls are working internally. What is not working is calls from and to PSTN. After pressing 9 for outbound calls, after pressing the next digit I get busy tone. Calls inbound not working. Inbound calls should get to AA.

Is SRST reference configured in CUCM? --> Yes and included in the IP Phones Device Pool

Chris, I guess is now more an issue of matching dialpeers and routing of calls. What should I be looking for in terms of debbuging in Voice Gateway?

Thanks again!

[Chris Deren]

So, looks like you have MGCP? Do you have proper dial-peers for H323 when the GW falls into SRST?  Remmeber MGCP does not work with SRST and you need proper dial peers. Also, if this is a PRI circuit you need MGCP fallback configured.

HTH,

Chris

[fuhrersk8]

Yes, MGCP is configured. I have both, a SIP trunk to Service Provider as the main circuit to PSTN and a FXO port as backup (mgcp gateway).

My goal is that if CUCM goes down, SRST phones still be able to route calls to and from PSTN through SIP trunk. Is this possible?

[Steven Griffin]

Nephtali,

Yes. It is possible. 

To route through the SIP Provider either in normal or SRST mode on your IOS 15.1 router be sure to add their session border controller (SBC) to your permitted list under "voice service voip" http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080b3e123.shtml You'll also need to enable CUBE. 

As Chris mentioned, be sure you've configured your dial-peers to use either the SIP trunk or the POTS lines. 

To answer your questions above:

1. The ephone and ephone-dn will be created automatically

2. If you are running 7940, 7942, and 7912 phones you should be just fine on IOS 15.1

3. You do not need to use the service mgcpapp.  Call Manager will create a special dial-peers for FXO/FXS lines that use that command during MGCP configuration and you should not mess with them.  PRI-ISDN lines do not require it.  Also, for best results, ensure that any dial-peer's you create come *after* those that MGCP creates.

4. Post a more complete config and we can continue to nit-pick it for you.

-Steven

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[fuhrersk8]

Thanks for the reply Steven. Excellent explanation. In my case, phones are registering into the router when in SRST mode and calls are proceeding between those internal extensions. The problem lies for inbound and outbound calls. I get busy tone after pressing the second digit after the 9 for outbound calls. I think is a dialpeer issue.

Below is the dialpeer created automatically by the router for mgcp:

dial-peer voice 999000 pots

service mgcpapp

port 0/0/0

!

I've created the following dialpeer for outbound calls when in SRST mode; is it necessary?

dial-peer voice 20 pots

description Outbound Calls in MHQ

service mgcpapp

destination-pattern 9T

port 0/0/0

!

I will only use the FXO port (voice-port 0/0/0) when in SRST. In terms of dialpeers, does any other dialpeer is necessary for SRST to route calls in this scenario?

Thanks again for your support!

[ajayviswanath]

Nephtali,

Can you configure another pots dial peer without the service mgcp command and check. Probably you can give it a lower preference. Also when u do a show voice port sum..do u see the voice port to be up.

AJ

[Steven Griffin]

Nephtali,

You don't need the service mgcpapp in your dial-peer 20.  You do need 'dial-peer voice 20 pots' for SRST to work.   The important bit is that dial-peer 999000 must exist before dial-peer voice 20 in the configuration.

-Steven

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[fuhrersk8]

Great Steven. Can I delete then dial-peer voice 20 and SRST should route both inbound and outbound calls with only dial-peer voice 999000 pots?

Thanks again!

[Steven Griffin]

No, you need both.  Just remove "service mgcpapp" from dial-peer voice 20.

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[fuhrersk8]

My bad. I misread your last post. I will remove service mgcapp from dial-peer voice 20 and then will proceed to shutdown CUCM for SRST testing. One more question before proceeding, when you remark that "dial-peer 999000

must exist before dial-peer voice 20 in the configuration", you mean that it must be the first match in the process of dialpeer matching,.i.e, the most specific?

[Steven Griffin · Accepted solution]

Nephtali,

The order it appears in the router config is important.  MGCP dial-peers must be first and your SRST dial-peers must come after.

dial-peer voice 999000 pots

service mgcpapp

port 0/0/0

!

dial-peer voice 20 pots

description Outbound Calls in MHQ

destination-pattern 9T

port 0/0/0

!

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[fuhrersk8]

Good to know. Thanks again. Any debug commands you recommend while testing?

[ajayviswanath]

You can do a debug voice dialpeer to check which dial peer the call matches

debug voice ccapi inout will show u the call details

AJ