01-11-2023 07:53 PM
I have configured all the setting related to PT, CSS, Route Pattern, SIP Profile, SIP Trunk Security Profile and SIP Trunk. Then i configure IP Phone for that particular partition and CSS but the call form IP Phone to Mobile Phone number doesn't works. Only the calls between the IP Phones works fine.
So, i am not able to find what is wrong with the configuration.
01-11-2023 10:07 PM
There can be a number of different reasons for why this fails. The first thing I would do is to do a DNA to see what the CM thinks should happen when the call is made. If that comes back as it should route the call look at the gateway that is supposed to handle the call to see if it does get the call. If it does then the next thing to check is what happens to the call in the gateway. Depending upon what type of connection you have with PSTN there would be different debugs that you’d need to activate in parallel. For ISDN use theses two, debug voip ccapi inout and debug isdn q931, for SIP use debug voip ccapi inout and debug ccsip message. Post the output from these if you need help interpreting the information.
01-12-2023 07:05 AM - edited 01-12-2023 07:06 AM
What is the call flow?
on the voice gate/cube do this: show dialplan number <PHONENUMBER>
for example: show dialplan number 2135551234
if you have a dial-peer that matches do this on the voice gateway:
service timestamps debug datetime local msec
service timestamps log datetime local msec
no logging console
no logging monitor
no logging rate-limit
no logging queue-limit
logging buffer 100000000 debug
voice iec syslog
debug ccsip messages
debug voip ccapi inout
You will need to do a: show log
then see what's happening.
01-12-2023 10:12 AM
For what reason do you recommend to use no logging monitor? With this you would not get any output to the terminal session if wanted to see what happens in real time, you’d have to do the test and then check whether something resulted in the log buffer. Although working, it might not be the most practical way of checking.
01-12-2023 07:22 PM - edited 01-12-2023 07:25 PM
Actually we have configure many partition PT, CSS, Route Pattern, SIP Profile, SIP Trunk Security Profile and SIP Trunk. However, all other are working fine but for only one PT, CSS, Route Pattern, SIP Profile, SIP Trunk Security Profile and SIP Trunk, its not working. The problem is only with outgoing call not working.
Incoming Call Works fine.
01-12-2023 09:39 PM
If you want help with this it doesn’t help if you simply repeat what you originally posted. Please follow the advice given and if applicable provide the asked for output. Without this we cannot help you.
01-13-2023 10:18 PM
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