Accepted Solutions
Issues with no or one way audio is about 100% related to network issues. I see that you have or at least had specific IP routing statements in your configuration. Add a default route instead and remove these and then test if you have connectivity between your gateway and the phone that is receiving the call.
Not sure if I understand what your comment about the numbers are about. Would you please mind to elaborate?
You don’t add the card in CM when there is a SIP connection with the gateway. The card and the port is controlled by the router and you reference it with normal dial peer configuration. See this document for details on this and how call routing operates in IOS. In Depth Explanation of Cisco IOS and IOS-XE Call Routing
The document that I shared earlier has most of that information. For anything else there are an ample amount of information on cisco.com. If not already done so I advise you to spend a little time on that page and search or browse through the documentation section where I’m sure you’ll find what you need. Apart from this you can find a wealth of information on the community if you search for it. After doing all of this if you still have questions please let us know the specifics and we’ll do our best to help you out. Just please have in mind that the community is not a place to get a complete example of a configuration, especially not when it involves multiple elements as in your question.
On you specific question, you should look at voice translation rules in the gateway to modify the called and calling numbers to fit your needs in your system landscape.
You dont need to add the NIM T1/E1 on CUCM. As sip trunks are already configured from CUCM pointing to Voice gateway, on the router you need some SIP dial-peer and voice translation rules(if required).
The document @Roger Kallberg shared is a good one. You can also find a large number of documents,video on. the internet.
If you can share more informations about your numbering plans may be we can provide you a sample config.
Lets assume that your DID block is 24559XXX. You can strip 2455 by using voice translation rules on gateway or From CUCM trunk configuration page set the significant digits 4. By this way you send the last 4 digits to CUCM.
Configure extensions/Dn's on something in 9XXX range. And you can receive the call..
Yes.
I hope the ISP is not sending 3 if its 9101.
if your dial plan is 9101 instead of destination-pattern ....$ use destination-pattern 9...$
Also incoming called use specific instead of ".".
If you are using a SIP trunk with your service provider I would recommend you to use the information in the VIA header to match the inbound dial peer. But based on your original post I would think that you use a ISDN circuit for the connection with your service provider, so based on that you need to have a POTS dial peer as the inbound and for that matter outbound calls. You should configure at a minimum four dial peers as per the below list.
- Inbound from your SP
- Outbound to your SP
- Inbound from your CM
- Outbound to your CM
Please share the output from debug ccsip message and debug voip ccapi inout running in parallel and capture an inbound call.
In general it’s not a good idea to hardcode the codec on a dial peer, it’s better to use a codec list so that there can be a negotiation of what codec to use.
Looks like you send multiple invites to your CM, but you never get any response back from it and after multiple retry’s the call fails with this.
DISCONNECT pd = 8 callref = 0x0721
Cause i = 0x82E600000000 - Recovery on timer expiry
Make sure that you can communicate between the gateway and the CM and that the SIP trunk that you have configured in CM has the correct IP address of the gateway, otherwise CM will not accept the communication and will drop the traffic as it is coming from an unknown source.
