Inbound voip calls not working

[Chris Hobbs]

Needs some help trying to troubleshoot why I can't receive calls. I have a 2621xm behind a pfsense box. The router can register to the sip provider and I can make outbound calls but I'm not able to get incoming calls to work when I call my did number. I've searched all over for a solution but I'm stuck this time. My config is attached but for some reason this site won't let me attach the debug file. Thanks in advance for any help. The debug is ccsip all.

[b.winter]

For me, I cannot access the share link, without a google account.

You can also ZIP all the files and attach it. Normally this works.

[b.winter]

Just by checking your config, you have no incoming dial-peer for the SIP provider.

You only have an outbound dial-peer and 2 other dial-peers for the pots side

Also, I strongly advise you to use a tenant configuration when working with SIP trunks on routers, since there are some feature compatibilities, where you must use tenants and it is the current "standard" to configure SIP trunks.

Just search the forum, you will find a some from the last 1 or 2 month.

[Nithin Eluvathingal]

Does the incoming  calls land on your gateway ?

what this dial-peer used for ?

dial-peer voice 100 pots
description ***Incoming from ADTRAN***
destination-pattern 279594
incoming called-number 1..........
direct-inward-dial
port 1/0:23

2.12.0.0

[Chris Hobbs]

I have an adtran 550 that I use as a sudo pbx. The 2621xm is attached to it using a voice card and I have my phones plugged into the adtran along with my landline. I have the dail plan setup in the adtran to forward all the local calls out the land line and all the long distance calls go out the isdn line between the adtran and the cisco router which end up going out my sip trunk. So anytime I dial 1-XXX-XXX-XXXX the adtran sends the call to the router. If I don't put in that destination pattern the sip trunk does not register. And yes the incoming call does hit my router but I keep getting this error in the messages.

SIP/2.0 400 Bad Request - 'Invalid IP Address'

[Scott Leport]

Hi,

You need a voip dial-peer specifying the incoming called-number command. If you are between PSTN services (PRI & SIP) then specify the test number your provider have given you to test incoming calls on your SIP service as your incoming called number. 

The voice class tenant config is good in theory, but not applicable to your hardware or IOS version. 

Make the suggested dial-peer addiction and test again. If it doesn’t work, post the debugs here as a text file, instead of a google drive link. 

[Chris Hobbs]

So I added this dial-peer for the incoming voip.

dial-peer voice 201 voip
description ***Inbound SIP***
session protocol sipv2
session target sip-server
incoming called-number 2709170147
dtmf-relay rtp-nte
codec g711ulaw
no vad

See the debug.txt.zip above for what happens when a call comes in to my DID number.

[b.winter]

You don't need the "session target sip-server", since it's a routing option (where to send the call to) for outgoing dial-peers.

Do you use NAT between the Cisco Router and the provider / internet? Because the provider sends the INVITE to the IP 173.25.110.118, but your Fa0/0 has an IP 192.168.5.11. That's why the router replies with 400 Bad Request - 'Invalid IP Address'.

Probably you would need to turn on SIP ALG on the FW, so that it also translates the IP's in the SIP headers.

[Chris Hobbs]

Yes, I do use nat, the router sits behind a pfsense firewall. Thanks for the help.

[Roger Kallberg]

If you cannot get the firewall to translate the address you’ll need to create a SIP profile on your SBC to handle this. If you have a look at the documentation for how to setup Direct Routing for Microsoft Teams you’ll get an idea on what would be needed for this.

[Scott Leport]

Hi Roger, 

I don't believe he can do SIP profiles as he's using a 2621xm router. I could be wrong though.

Other than replacing the hardware, I think only choice he's got is to enable SIP Inspection / ALG on his Firewall. 

[Roger Kallberg]

Your absolutely correct. Totally missed that part. ':-|'

[Chris Hobbs]

And that is the problem. Pfsense does not do ALG. It does have the ability to install a program called Siproxd which is a proxy/masquerading daemon for the SIP protocol. I just haven't had time to work on it yet.

[Roger Kallberg]

As @Scott Leport suggested it would be advisable to use something more recent for the hardware that you run the SBC on. Suggest a ISR 4K or at a minimum ISR 2, aka 29xx/39xx.