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outbound call fail using SIP Trunk on CME

moustafa.idc
Level 1
Level 1

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,

 

26 Replies 26

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

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.



Response Signature


I removed the bind under dial-peer, it worked as well. the only thing is it took a little bit more time for call setup.

So I guess both ways are valid.

 

attached is a copy of the final config

 

Thanks a lot Roger!

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Given that your version of IOS is quite old and outdated I would also recommend you to upgrade it.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.



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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

 

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.



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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.



Response Signature


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

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.



Response Signature


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?

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

moustafa.idc
Level 1
Level 1

Hi Leonardo,

 

please find attached file