call drops from PSTN to CUCM

[grinch182]

good day everybody,

I'm having a problem with calls fom PSTN to VoIP phones.

There is a cheme PSTN (Nortell) ----> Cisco 2911 (gateway) ----> Cisco C2600-IX-M (Gatekeeper) ----> CUCM 8.6

Calls from VoIP phones to PSTN works just fine. When PSTN user making call to VoIP we got strange result: VoIP user hearing silence and call drops after 10 second, however PSTN user keep hearing ringback dial tones no answer.

I could provide any config sections you need as well as debbug output.

Thanks in advance.

[Nishant Savalia]

Hi Grinch,

I think you have defined as "Gatekeeper controlled trunk" while configuring gateway in CUCM?

Please share the snapshot of gateway configuration in CUCM and running config of router.

Also, please do share detaield trace of RTMT from CUCM for the failure call.

Regards,

Nishant Savalia

[grinch182]

Good day Nishant,

There are a trunk configuration printscreen and GK config as weel in attachment.

Could you please tell how to get datailed trace from RTMT as I not very experienced with that.

Thanks

[Nishant Savalia]

No problem... will tell you in detail but mean time please do share following debugs from gateway.

1). debug h225 asn1 and debug h245 asn1

2). debug voice dial-peer

3). debug voice ccapi inout

Also, please share full running-config of router.

Regards,

Nishant Savalia

[grinch182]

Thanks for your help. I have solved this problem.

To solve this problem I changed Significant Digits value from all to 4 in Call Routing Information - Inbound Calls section.

Best regards,

GRinch

[Nishant Savalia · Accepted solution]

Great that you solve the issue but strange part is that by changing Significant digits, it got resolved.

Because as you said that call get disconnect after 10 seconds and PSTN users hears ringback tone, where the most likely issue is due to TCS compatibility. And for which debug was required to confirm it.

Regards,

Nishant Savalia

[grinch182]

Good day Nishat.

First of all thanks for your help. I thought there is a problem with signaling or transcodding. However  problem had a few pieces. VoIP site have prefix 2075XXX (207 is a techprefix ) all phones got DNs 5XXX. CUCM have pattern 2XXXXXX and destination to GW. So without cutting off tech prefix 207 CUCM sending all calls to GW and gateway sending it back to CUCM following call forwarding rulles. Because of poor design and wrong trunk and dial-peer configuration customer having kind of routing loop. 

All the best,

GRinch

[Nishant Savalia]

Hi,

Thank you for sharing information but still i have a doubt that how come call can be connected if routing loop is happening?

Regards,

Nishant Savalia

[grinch182]

GW did cutting off and translate last four digits back to CUCM so phones started ringing. I belive the problem was to sent signaling back to PSTN and start voice flow.

[Nishant Savalia]

Below is the link to enable and colllect detailed trace from RTMT.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094e89.shtm

Regards,

Nishant Savalia