07-09-2018 01:59 AM - edited 03-01-2019 02:07 PM
Hello.
My configuration mainly includes a Cisco ISR4331 with CUCME (Express) and CUBE.
I also have a SIP trunk provided by Keyyo (https://www.keyyo.com/fr/trunk-sip/).
This SIP trunk is for placing calls for (and receiving calls from) "public" phone numbers.
For example, calling someone in another city or country or receiving a phone call from a customer.
I mean something classical that any normal phone was capable of doing "before" without any major difficulty.
It is something simple I need to do, yet it seems nobody ever did this before and I cannot find any example that fits my case.
Examples I find are always about:
- "lab" configuration (which is not my case),
- CUCM which is not my case (I have a CUCME, Express),
- Call Manager (which is obsolete),
- calling from Head Quarters to Branch Office (which is not my case, I have only one office, one physical location),
- calls impling two CUCMs (which is not my case),
- etc.
I have tried to configure VoIP "dial-peers" and "sip-ua" but I cannot make or take any calls.
Everything fails both incoming calls and outgoing calls.
There are so many "debug" and "show" commands like "debug ccsip messages" or "show dial-peer voice" with tons of output that I cannot see anything in that mess.
Is there anyone that can help me here figure out what's wrong?
"a.alekseev" on this forum wrote "check the routing".
That's all. Short sentence. That's not what I call help.
I have tested so many things that I'm lost.
Anyone here to put me on the right track?
Thank you for your understanding.
07-09-2018 07:30 AM
07-09-2018 07:42 AM
Hello Nipun.
I don't think that my ITSP requires the SIP trunk to be registered.
I am counting on the fact that they don't.
I am sorry but I'm not willing to post my configuration publicly on the Internet.
I could send it to you as a Cisco employee but not on the Internet.
Thank you for your understanding.
07-09-2018 07:51 AM
07-09-2018 08:58 AM
Hello Nipun.
Thank you. I sent you a private message.
Best regards.
07-09-2018 10:51 AM
07-10-2018 12:57 AM
Hello Nipun,
Thank you for your examination.
@R0g22 wrote:
The from header and contact header has your internal 4 digit extension. That needs to be corrected to a DID for that SIP trunk. Use a translation rule to fix that and then test.
You've seen my conf. can you please tell me how you would write this translation rule.
I've really no idea what you are talking about.
Thank you.
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