11-27-2011 02:06 PM - edited 03-16-2019 08:15 AM
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?
11-27-2011 02:49 PM
dial-peer v 1 voip
destination-pattern 2...$
session-target ipv4:1.1.1.2
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11-27-2011 03:56 PM
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
11-28-2011 05:26 AM
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.
11-28-2011 05:56 AM
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
12-01-2011 09:40 AM
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?
12-01-2011 10:48 AM
Check if you have dial peer in Site A pointing to Site B
12-01-2011 11:14 AM
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
12-01-2011 11:21 AM
Hi ,
If possible ,can you post a sh run conf for the sites.
12-01-2011 02:32 PM
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
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.
12-01-2011 03:25 PM
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?
12-01-2011 07:11 PM
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
12-02-2011 08:45 AM
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
12-03-2011 12:43 AM
Thanks to you all. Its working now.
12-03-2011 09:35 PM
Hi
If you could tell what was the issue...just for informartion
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