06-21-2012 06:31 AM - edited 03-16-2019 11:46 AM
Hello,
We are having a strange issue that we can't seem to track down and I was hoping someone could help point us in the right direction. Previously we were using a 3725 ISR integrated with our Hicom 300 and calls between the two were working fine, but lacked caller ID. This is a feature we wanted to give to our users moving forward with our migration (slow) to Cisco VoIP. We replaced our 3725 that had failing hardware with a new 2951 and had to convert the T1 integration to a PRI (QSIG). So, we copied what we had in our previous config and made the necessary adjustments for the PRI. At one point we thought that we had everything working including caller ID from PBX to VoIP extensions and VoIP to PBX extensions.
As it stands now, we cannot make 4-digit calls from PBX to VoIP extensions. 4-digit dialing from VoIP to PBX works perfectly fine. The strange thing is that if we call from a cell phone or dial out of the PBX by using 8+7-digit DID number from a PBX extension, calls get delivered to the correct VoIP phone. It seems that when we call the 4-digit VoIP extension that it just wants to come right back over the PRI to the PBX again and eventually after a long period of time, it times out. We having been looking heavily at the dial peers, but this config used to work on the old ISR, so I don't know that this is the issue.
Here is our current setup and integration:
PSTN <<==>> Hicom 300 9006.6 (PBX) <<== (PRI-QSIG) ==>> 2951 ISR (Gateway) <<== (H.323 Gateway) ==>> CUCM 7.1.3
If someone could point us in the right direction, as far as troubleshooting is concerned, that would be fantastic. If you need additional information and I'm sure you will, please let me know. Thanks for your help.
06-21-2012 06:36 AM
Jooeph,
Can you provide the config from the 2951 (show run)
Can you also provide a Q931 debug of a failed call (debug isdn q931)
Along with this tell us what numbers you are calling and from what numbers.
Regards,
Alex.
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06-21-2012 06:49 AM
I'm gathering that information now. I should also mention that when SRST mode is in operation, it seems to work fine. It almost seems like that when it isn't working it still picks the correct dial peers, but call manager isn't accepting the 4-digit calls?
06-21-2012 06:55 AM
Joseph,
Can you also look at the H323 gateway page on CUCM.
Check that inbound calls are using the correct CSS and that the significant digits is set to allow the call in.
May be post the screenshots if you are stuck.
Regards,
Alex.
Please rate useful posts.
06-21-2012 07:05 AM
06-21-2012 07:21 AM
Joseph,
Looking at your config and the debug.
You are calling to EXTN 5910
This matches dial peer 110 and sends the call back to the HICOM.
!
dial-peer voice 110 pots
destination-pattern [2-7]...
progress_ind setup enable 3
progress_ind alert enable 8
port 0/0/0:23
forward-digits 4
!
Can you add a test dial peer for VOIP.
!
!
dial-peer voice 5900 voip
description 59 RANGE TEST to VOIP
destination-pattern 59..
session target ipv4:10.0.7.4
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
Try and test
Regards,
Alex.
Please rate useful posts.
06-21-2012 07:27 AM
Alex,
We have a dial-peer 121 specifically for 5910. Wouldn't this be doing the same thing?
06-21-2012 07:59 AM
Joseph,
You are right.Sorry my mistake.
Looking again at the trace
Jun 21 14:00:34.769: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x13A6
15 seconds later the call is sent back to the HICOM
Jun 21 14:00:49.778: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x00C5
Almost immediately the call is returned from the HICOM
Jun 21 14:00:49.890: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x13A7
Again
15 seconds later the call is sent back to the HICOM
Jun 21 14:01:04.896: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x00C6
And so on
So what is happening in this 15 second gap
Can you confirm that there is IP connectivity between the CUCM server and the H323 Address
10.0.7.3
Regards,
Alex.
Please rate useful posts.
06-21-2012 08:07 AM
Alex,
We've confirmed connectivity and some calls are going through the PRi and on to the VoIP extension. For example: If a user calls from a cell phone (or dials out of the PBX) to one of the VoIP DIDs the call gets there. I've attached a debug of this for comparison. Also, we looked at the CSS and believe it to be correct. Also, inbound significant digits is set to "All".
06-23-2012 08:23 PM
After some further testing, we have noticed that other PBX extensions are getting through to these same VoIP extensions. We opened a case with tac and provided them with debugs and this info. It seems as though they are hitting the same dial peers whether it works or not but call manager either never receives it or rejects it. Thoughts?
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