Calls going to wrong CSS

[mohammed asif suleman]

Hi,

we recently configured branch Gateway for the PSTN calls originating from the brach should routed through PSTN line connected on FXO ports

we have call manager cluster at the Head Office, The internal calls to Headoffice have no problem.

but when the brach office dials any PSTN number its asking for FAC and CMC codes,eventhough i didn't configure the route pattern for FAC and CMC(they can call without codes).

The route Pattern is -----9.!

The partition   BR_PT_INTERNATIONAL

css    BR_CSS_INTERNATIONAL

gatway/Route List   LOCAL ROUTE LIST

Route Group   JED_RG

Gateway ip  192.168.21.5  which is H.323 gateway connecting to call manager at Head Office throug HO gateway 192.168.12.5 over MPLS

I thought there is a problem with Route pattern...but after all R&D also i cannot solve this

Please can any one help

[Harmit Singh · Accepted solution]

Hi Mohammed,

Looking at the DNA output:

-

  " LineCSS="BR_CSS_INTERNATIONAL" AARGroup=" " AARCSS=" " />

  90542014824

  RouteThisPattern

  0542014824

  No

  0

  No

So it looks like the right CSS and partition is being matched. When you say, the call again got disconnected, did it prompt you for the FAC/CMC code? If yes, are you sure it is coming from the CUCM and not from the Telco? Can you connect an analog phone to the FXO lines and make an international call successfully?

--
Regards,
Harmit.

[mohammed asif suleman]

Hi Harmeet,

The Phone is at remote site and the persons working there cannot connect the anlog phone to the line.

international is not enabled for the anlog line but they can dial mobile and local, but when dialing to mobile the call getting disconnected.

the call didn't prompt for the FAC or CMC now as for international css, there is no FAC and CMC required.

[Harmit Singh · Accepted solution]

Hi Mohammed,

That's good to know that you dont hear the prompt for FAC/CMC anymore. Based on what you mentioned, only the mobile calls are getting disconnected. Have mobile calls worked successfully on these FXOs before?

Can you run the following debugs for one test call:

debug mgcp packets

debug voip ccapi inout

debug vpm signal

debug voip vtsp default

debug voip vtsp session

Once the call disconnects, capture the following output:

Show voice trace

This would help us to see if the call is hitting the FXO port or terminating before hand.

--
Regards,
Harmit.

[mohammed asif suleman]

Hi All,

Thanks a lot for your priceless support.

Harmit you really Rocks.

The problem was the prefix digit '9' getting stripped at 2 places one is at route pattern level and another in pots dial-peer on the gateway.

so i change at route pattern discard digit predot to no digits and it started working.

once again thakyou so much to all.

[Harmit Singh · Accepted solution]

Hi Mohamed,

Thanks for sharing the resolution (+5), glad we could help! :-)

--
Regards,
Harmit.

[mohammed asif suleman]

Hi Harmit,

sorry if i am making this discussion too long.

i have one more query...

At this branch we don't have IVR system. if any one calls the call will directly lands on 2022, the receptionist has to pick the call and transfer to the person the call is for.

how can i implement IVR system for the branch, shell i install unity on any server on VM and point the calls to the unity.

or shaell i go for unity module for router it self. or can i use the unity available at Head Office.

if i am using the existing Unity at Head Office then call will go to Head office first, after IVR finishes again it will come to Branch, this will eatup all the bandwidth available or i have to restict the calls.

which is suitable for our needs , Please suggest........

[Harmit Singh · Accepted solution]

Hi Mohammed,

No worries :-)

You've correctly identified a few possiblities for implementing an Auto Attendant:

1>     Install a Unity Express at the branch site, where the calls can be sent for the Autoattendant script. This would avoid bandwidth consumption over the WAN.

2>     Utilize the existing Unity / Unity Connection system at the Head Office for calls to be sent to an Autoattendant Greeting. Bandwidth utilization can be reduced by using G729 codec. Once the call is transferred to an IP Phone at the branch site, the bandwidth utilized over the WAN would be released.

3>     I believe you should be able to create a TCL script on the remote gateway itself which would let you route the call to the branch phones.

Please review the design guide for Unity Connection (assuming you have Unity Connection) and SRND for CUCM:

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/8x/design/guide/8xcucdg050.html#wp1085205

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/vmessage.html#wp1086561

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/vmessage.html

HTH.

--
Regards,
Harmit.

[mohammed asif suleman]

Hi Harmeet,

thankyou for the preciuos suggestion,

i will implement the Auto-Attendent to use the existing cisco unity connections at head office and test the bandwidth consumption.

if it is ok then we can continue that other wise i have to purchase the unity module for router.

[techguy · Accepted solution]

Use unity express (cue) make cti route point and ports. Your calls will goto cucm and greeting message will be heard from cue


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