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CME WITH PSTN to CME without PSTN.

rammany19
Level 1
Level 1

i have 2 sites named A and B

Site A has CME and PSTN.

Site B has CME no PSTN.

What I want is to make CME to CME calls and Site B can call PSTN located @ Site A.

Is it by just pointing to SITE A IP address, I will achieve this?

23 Replies 23

shrvaran
Cisco Employee
Cisco Employee

dial-peer v 1 voip

destination-pattern 2...$

session-target ipv4:1.1.1.2

Sent from Cisco Technical Support iPad App

As suggested by Shrvaran you achive this by using dial peers and pointing to the other CME as session target, you will also need to allow sip or h323 protocol via the following configs:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

HTH,

Chris

rammany19
Level 1
Level 1

Hello,

i want to make use of the PSTN @ Site A from Site B. Site B doesnt have PSTN so if Site B want to make outside call it should route it to Site A CME with PSTN.

please reply ASAP if this is achievable.

Hi

As the shrvaran & chris have shown this CAN be acheived. (+5 )

You need IP connectivity between the sites.

You need to create dial peers and allow the service in voip.

You need to know if you want to use H323 or SIP

So at site "B" say dial a 9 for PSTN access

CME-B

!

dial-peer voice 9 voip

description *** TESTING PSTN H323 DIAL PEER ***

destination-pattern 9T

voice-class codec 1

session target ipv4:172.16.255.11

incoming called-number .

dtmf-relay h245-alphanumeric

ip qos dscp cs3 signaling

no vad

!

CME-A

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

Regards

Alex

Regards, Alex. Please rate useful posts.

Hello

Having d issue here, SiteB can call SiteA but SiteA cannot call SiteB. I have issued "voice service voip" allow-connectionson both routers.

Please what might cause this?

Check if you have dial peer in Site A pointing to Site B

Hello

I have a dial-peer in Site A pointing to Site B, I use 4digit. When you call from Site A 4digit, when you type in all the 4 digit...I get busy tone. Debug showing its initiating the call but showing "Disconnectcause is 10" .

Please need help ASAP.

Thanks

Hi ,

If possible ,can you post a  sh run conf for the sites.

SITE A directory and Dial-peer: 1..,208 and 218 ,3..,4..,5xx,6xx,

Dial-peer voice 1001 voip

Destination-pattern 7T

Session protocol sipv2

Session target sip-server

Dial-peer voice 600 voip

Description calls to SiteB

Destination-pattern 2[1-3]..$

Session target ip:1.1.1.10

Incoming called-number .

Dtmf-relay h245-alphanumeric

SITE B directory number and Dialplan

2[1-3]..$

Dial-peer voice 10 voip

Description calls to SiteA

Destination-pattern ..$

Session target ip:192.168.8.1

Incoming called-number .

Dtmf-relay h245-alphanumeric

Dial-peer voice 20 voip

Description calls to SiteA

Destination-pattern 1..$

Session target ip:198.168.8.1

Incoming called-number .

Dtmf-relay h245-alphanumeric

Dial-peer voice 20 voip

Description calls to SiteA

Destination-pattern 20.$

Session target ip:198.168.8.1

Incoming called-number .

Dtmf-relay h245-alphanumeric

Dial-peer voice 20 voip

Description calls to SiteA

Destination-pattern 2[4-9].$

Session target ip:198.168.8.1

Incoming called-number .

Dtmf-relay h245-alphanumeric

Note; Site B can call site A but not vice versa.

So site A has 3 digit extension's and B has 4 digit..

Is the call from site A reaching the Site B  .Can you see any  q931 logs in B?

rammany19
Level 1
Level 1

hello,

note: Site A have the voice-card connected to the pstn but site B did not have it. site B do connect to site A for pstn access.

site A has 3 digit while site B has 4 digit extension number.

site B can call site A but Site A cannot call back to site B.

we are using FXO/FXS port on site A.

through debug command, calls are showing from site A to Site B but after u dial the last 4 digit you get a busy signal tone.

please, what might have cause this and what debug command shud i turn-on on site B to check maybe calls are hitting site B router?

debug isdn 931 IS NOT WORKING.

HELP

Hi ,

sorry 'debug isdn q931' was not the command.

You can use 'debug voice ccapi inout' to check if the calls are hitting site B.

Also you can check if correct dial-peers are matched during the call flow with 'debug voice dialpeer'.

HTH

Thanks to you all. Its working now.

Hi

If you could tell what was the issue...just for informartion

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