11-10-2021 12:32 PM
Hi all,
I have Cisco CME with some SCCP & SIP phones. I have also sip provider to route local and international calls to.
When I configured the router for sip phones, outbound call stopped working. I'm still able to make inbound calls through sip trunk.
Attached is the config and traces.
appreciate if somebody can look at it and see if I missed something.
Thanks,
Solved! Go to Solution.
11-12-2021 11:05 AM
Thanks a lot Roger, I have everything working fine and tested. unfortunately I can't check RTP quality but I will test it again on Monday.
What I had to add under each dial-peer is:
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
as the sip phones are using Gig0/0/1 under global register
11-12-2021 11:25 AM - edited 11-12-2021 11:31 AM
I don’t understand why this would be needed. The bind statements for the dial peers would be done on the tenant. The SIP phones should use the bind statements on the global configuration under voice service voip.
11-12-2021 12:00 PM
11-12-2021 12:52 PM - edited 11-12-2021 11:20 PM
Remove this from voice service voip.
localhost dns:sip.bluetelecoms.com
And attach the tenant to the inbound dial peer as well. On the topic of inbound dial peer I would recommend you to use a more specific match criteria than incoming called number. The recommendation is to use information in the VIA header.
The previous shared document outlines details on this.
Given that your version of IOS is quite old and outdated I would also recommend you to upgrade it. If you want to keep using traditional licenses the last version you could use is 16.9. Otherwise you should be able to go with 17.3.4 that we are currently standardised on at my work. This requires Smart Licensing FYI.
11-12-2021 03:45 PM
I will upgrade IOS next week to 17.3.4 checking with cisco if I have the valid license for this upgrade.
I have removed
localhost dns:sip.bluetelecoms.com
I always was suspicious about it..
as for the inbound dial peer I checked the document you sent does it have to be matching the INVITE header to be honest I don't know what is VIA header. trying to read about it.
Thanks again for you help
11-12-2021 11:34 PM
I’ll post an example later today on how to use VIA header to match inbound dial peer.
Yes it would be in the incoming invite. It’s one of the mandatory SIP headers. Basically you match on the information that you get in the VIA header, usually this is one IP or a network of IPs depending upon what your service provider sends to you.
11-13-2021 06:01 AM
Example of how to match on IP addresses in VIA header.
voice class uri PSTN sip host ipv4:<ITSP SBC IP 1> host ipv4:<ITSP SBC IP 2> ! dial-peer voice 4 voip description Incoming Dial Peer from PSTN incoming uri via PSTN
There are other options that can be used in the voice class uri, those are covered in the before linked to document.
11-13-2021 02:03 PM - edited 11-13-2021 11:29 PM
Thanks Roger, I went through sip structure and rfc and yes I understand now what is via. thanks for the example.
-------------------------------------------------------
the following is the update to inbound call for security measures:
-------------------------------------------------------
voice class uri ITSP sip
host 185.32.xx. -------------- 3 octet this works for me to match the whole subnet instead of matching ipv4 one at a time. I also tested the dial plan from other ip's and it work as it should
dial-peer voice 9000 voip
no incoming called-number .
incoming uri via ITSP
-------------------------------------------------------------
For some reason SIP phone is not dialing out though sip trunk.
When you get a chance is it possible to look at the final config attached.
3546 ORG T88 g711ulaw VOIP P3202 192.168.50.69:29120 D10
3554 ORG T0 g711ulaw VOIP Psip:sip.bluetelecoms.com:5060 0.0.0.0:0 D39
3553 ANS T0 g711ulaw VOIP P3202 192.168.50.69:23500 D39
3557 ORG T0 g711ulaw VOIP Psip:sip.bluetelecoms.com:5060 0.0.0.0:0 D39
3556 ANS T0 g711ulaw VOIP P3202 192.168.50.69:18512 D39
3559 ORG T0 g711ulaw VOIP Psip:sip.bluetelecoms.com:5060 0.0.0.0:0 D39
3558 ANS T0 g711ulaw VOIP P3202 192.168.50.69:18626 D39
3567 ORG T0 g711ulaw VOIP Psip:sip.bluetelecoms.com:5060 0.0.0.0:0 D39
3562 ANS T0 g711ulaw VOIP P3202 192.168.50.69:30810 D39
appreciate your help
Moustafa
11-14-2021 12:17 AM
I think that you might need to create another inbound dial peer that matches on the network address of your phones with the inside network as it’s bind statements for the phone to be able to make the call.
11-11-2021 09:48 AM - edited 11-11-2021 09:48 AM
Hi,
What happens when you remove the SIP phones / unregister them from the CME?
Is the SIP trunk operational again and can you dial out via the SCCP registered phones?
11-12-2021 07:19 AM
I see that the "From" has a caller ID in England (country code 44), but the "To" has a long-code without the 0 in front. A 404 Not Found error means the service provider didn't like the digits you sent. For the UK, this may be either or both of the dialed number or the caller ID.
Has BlueTelecoms confirmed to you that the dialed number (12146043977) and caller ID (442037752050) are in the format they want?
Maren
11-10-2021 05:42 PM
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