11-06-2024 12:08 PM
Hello,
I have a problem with my CUCM. everything works on internal numbers. calls are made to international numbers from outside, there is a conversation. but there is a problem that when calling from the inside to the outside, it drops after the first 10 seconds. During these 10 seconds, the voice of the caller from the inside does not go away, but the voice from the outside is heard. Has anyone faced this problem. Please help me if you have any information.
(register is a separate vlan provided by the ISP. Everything is configured correctly on the Cisco Router. The problem is said to be in the CUCM)
11-06-2024 12:15 PM
If you share your running configuration and output from these debugs all run simultaneously, debug ccsip messages and debug voip ccapi input, we can see if we can identify your problem. Please collect the configuration and debug output in text files and post it in your reply.
11-06-2024 12:23 PM
Check formatting of numbers which is implemented in CUCM is it in E.164 format!!!
Since internally they are assigned extension ids but for internation call it should be terminated in E.164 format
11-06-2024 01:02 PM
@sakshamsharma04 What has that got to do with one way audio?
11-06-2024 01:13 PM
Since only International call is facing issue. So we can rule out RTP, RTCP protocol causing this issue. As he mentioned calls are working fine.
1) No need to inspect wireshark traffic!!
2) CUCM mostly causes this issue because of formatting and extensions which are provided for internal numbers that are in format of "9876" which needs to translated to E.164 for international call since we need to terminate call.
3) Also there could be issue of SIP ALG configured on Firewall which could be causing issue of "Double NAT"
We can go ahead and disable same or check it via SIP ALG tool.
11-06-2024 01:53 PM
In all the years I’ve worked with CM I never encountered one way audio caused by number format. IMHO that is not the reason for the OP issue.
11-06-2024 02:01 PM
I will wait for your solution then
11-06-2024 11:53 PM - edited 11-07-2024 04:02 AM
That's not what I meant. Just wanted to clarify that the format of how you send called or calling numbers has no correlation with one way audio.
11-07-2024 02:36 AM
Before I answer the original post, I thought I would address this.
You need to perform a packet capture to ensure that RTP is flowing. I highly recommend this to always identify who is dropping the packets.
If the number format is incorrect, the call will not be processed by the ISP, and they will simply drop the call. Have you ever had a call connect when you dialed the wrong number?
SIP inspections come into play if the firewall is between the call. As far as I know, it’s disabled globally on the firewalls. Moreover, the firewall doesn’t wake up only for international calls and drop the packets.
11-07-2024 03:23 AM
I really appreciate your input. I would be mindful in future for these issues!!
11-07-2024 02:44 AM
First of all, you haven’t provided us with the details of your connection with the ISP. Is it SIP or PRI? From your post, I assume it’s SIP. It would be great if you could confirm this.
As @Roger Kallberg highlighted, the running configuration of the gateway/CUBE would help us in troubleshooting.
You need to identify why the call got dropped and who dropped the call—was it from your end or the ISP? If it’s the ISP, you need to inform them about this.
Since all other calls, including landline and GSM, are working fine and this issue is only with international calls, most probably the issue could be with the ISP.
If this issue was due to your setup, it would affect all calls, not just international ones, if they use the same path for sending calls to the ISP as landline and GSM calls.
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