VG224 SIP configuration, Outbound calls OK, Inbound calls NOK

[Cr2803]

Hi everyone,

The issue I have is already explained in the subject,

We have recently setup a VG224 for SIP use, with an external SIP provider.

We could make it work for outgoing calls (from analog phone to for instance my mobile phone 069826....), please find attached below the config file.

We failed to make it work for any inbound traffic (please see the log file attached below)

Anyone can help?

Thank you in advance!

Regards

[b.winter]

Are you really calling this number "A2844FD755" as the called party number?
I don't believe so. It's probably just the username of the SIP account to authenticate against your provider.

cc_api_call_setup_ind_common:
cisco-username=069826xxxx
----- ccCallInfo IE subfields -----
cisco-ani=sip:069826xxxx@15.xx.xx.xxx
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=sip:A2844FD755@10.70.0.5:57012
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0

[Cr2803]

Hi b.winter,

First of all, thank you for having replied,

also my colleague and I, we are not SIP specialists and we are discovering this functionality on the voice gateway device.

According to you we shoudln't use this 'username' as the called party number,

so my question now, is: how should we set up the called party number on the VG224 exactly ?

Can you please provide an example? I couldn't find anything on the Cisco.com pages (ie. https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-multi-trunks.html)

==============================

So far we have set it up like this:

dial-peer voice 244 pots
description telephone analogique test.vg244
authentication username A2844FD755 password 7 XXXXXXXXXXX
no digit-strip
port 2/0
!
dial-peer voice 1 voip
description Outgoing 10 number calls
destination-pattern ..........
session protocol sipv2
session target sip-server
session transport udp
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 3 voip
description All incoming calls
session target sip-server
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
!
sip-ua
credentials username A2844FD755 password 7 XXXXXXXXXXX realm sip.preprod.xxx.com
authentication username A2844FD755 password 7 XXXXXXXXXXX
registrar 1 dns:sip.preprod.xxx.com expires 360
sip-server dns:sip.preprod.xxx.com
host-registrar

also...

voice-port 2/0
ring frequency 50
compand-type a-law
cptone FR
timeouts interdigit 120
timeouts call-disconnect 5
timeouts ringing infinity
timing hookflash-in 1550 150
description telephone analogique
bearer-cap Speech
station-id name A2844FD755
station-id number A2844FD755
caller-id enable

Usename and password have been given by our provider.

Can you please point me in the right direction? that would be great, many thanks

[b.winter · Accepted solution]

You cannot do anything on the VG for the incoming call, because you are receiving the SIP message from the provider. So the provider needs to correct this.

Maybe the VG is registering incorrectly with the provider, but then again, the provider needs to tell you, how the registration looks like. This info cannot be found in the internet or in this forum. Only the provider can give you that info.

Please also attach a full output of a call with the following debugs enabled:
debug ccsip messages
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74

[Cr2803]

I got error attaching the file, couldn't do so! You'll find it pasted right below.

Also please find attached debug with an outgoing call, that works fine

For debug of an inbound call, where we get issues, you'll see SIP/2.0 503 Service Unavailable

but I can't tell if it's unavailable by our end

=========================================

Here's the log:

=========================================

//-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:A2844FD755@10.70.0.5:57012 SIP/2.0
Record-Route: <sip:52.xxx.xxx.xxx;lr;ftag=as119c492b;nat=yes>
Via: SIP/2.0/UDP 52.xxx.xxx.xxx:5060;branch=z9hG4bK4e59.3037d899cec6eb237cd2eeb9dfe1efe1.0
Via: SIP/2.0/UDP 172.16.0.207;received=172.16.0.207;branch=z9hG4bK6UK5oaEr;rport=5060
From: <sip:069826xxxx@15.236.xxx.xxx>;tag=as119c492b
To: <sip:A2844FD755@52.xxx.xxx.xxx>
CSeq: 102 INVITE
Call-ID: xxx
Max-Forwards: 69
User-Agent: [provider_name_hidden]
Date: Thu, 14 Sep 2023 09:35:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Contact: <sip:61B65343-6502D3D800031736-1E4E4700@172.16.0.207:5060;transport=udp>
Content-Length: 337

v=0
o=- 143918324 143918324 IN IP4 35.181.xxx.xxx
s=-
c=IN IP4 35.181.xxx.xxx
t=0 0
m=audio 9080 RTP/AVP 107 8 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 maxaveragebitrate=35000;useinbandfec=1;usedtx=1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
a=maxptime:20
a=direction:both

//-1/DB015DB18EDA/CCAPI/cc_api_call_setup_ind_common:
Interface=0x65208520, Call Info(
Calling Number=sip:069826xxxx@15.236.xxx.xxx,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=sip:xxx(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=3, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=4221
//4221/DB015DB18EDA/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 52.xxx.xxx.xxx:5060;branch=z9hG4bK4e59.3037d899cec6eb237cd2eeb9dfe1efe1.0,SIP/2.0/UDP 172.16.0.207;received=172.16.0.207;branch=z9hG4bK6UK5oaEr;rport=5060
From: <sip:069826xxxx@15.236.xxx.xxx>;tag=as119c492b
To: <sip:A2844FD755@52.xxx.xxx.xxx>
Date: Thu, 14 Sep 2023 09:35:20 GMT
Call-ID: xxx
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

//4221/DB015DB18EDA/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x65208520, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=sip:069826xxxx@15.236.xxx.xxx,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=xxx(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=1, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
//4222/DB015DB18EDA/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:A2844FD755@sip.preprod.xxx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.70.0.5:5060;branch=z9hG4bK4AA2AF
Remote-Party-ID: <sip:069826xxxx@10.70.0.5>;party=calling;screen=no;privacy=off
From: <sip:069826xxxx@10.70.0.5>;tag=9A8B458-FD2
To: <sip:A2844FD755@sip.preprod.xxx.com>
Date: Thu, 14 Sep 2023 09:35:20 GMT
Call-ID: xxx
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3674299825-1377309166-2396685637-0955683865
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1694684120
Contact: <sip:069826xxxx@10.70.0.5:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 287

v=0
o=CiscoSystemsSIP-GW-UserAgent 5894 8648 IN IP4 10.70.0.5
s=SIP Call
c=IN IP4 10.70.0.5
t=0 0
m=audio 16748 RTP/AVP 8 100 101
c=IN IP4 10.70.0.5
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

//4222/DB015DB18EDA/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.70.0.5:5060;branch=z9hG4bK4AA2AF;rport=57012
From: <sip:069826xxxx@10.70.0.5>;tag=9A8B458-FD2
To: <sip:A2844FD755@sip.preprod.xxx.com>;tag=358ba19e3e96a89f558cdbe342a00794.6d273483
Call-ID: xxx
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.preprod.xxx.com", nonce="ZLVBGU9iItEiH5MfkvkF3RNb17gEtAO1Fexv8PGA/vISFFe2A"
Content-Length: 0

//4222/DB015DB18EDA/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:A2844FD755@sip.preprod.xxx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.70.0.5:5060;branch=z9hG4bK4AA2AF
From: <sip:069826xxxx@10.70.0.5>;tag=9A8B458-FD2
To: <sip:A2844FD755@sip.preprod.xxx.com>;tag=358ba19e3e96a89f558cdbe342a00794.6d273483
Date: Thu, 14 Sep 2023 09:35:20 GMT
Call-ID: xxx
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

//4222/DB015DB18EDA/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:A2844FD755@sip.preprod.xxx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.70.0.5:5060;branch=z9hG4bK4AB6A9
Remote-Party-ID: <sip:069826xxxx@10.70.0.5>;party=calling;screen=no;privacy=off
From: <sip:069826xxxx@10.70.0.5>;tag=9A8B458-FD2
To: <sip:A2844FD755@sip.preprod.xxx.com>
Date: Thu, 14 Sep 2023 09:35:20 GMT
Call-ID: xxx
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3674299825-1377309166-2396685637-0955683865
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1694684120
Contact: <sip:069826xxxx@10.70.0.5:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="xxx",realm="sip.preprod.xxx.com",uri="sip:A2844FD755@sip.preprod.xxx.com:5060",response="1ea1dc6a885ac433be9c0ac11626f761",nonce="xxx",algorithm=md5
Max-Forwards: 68
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 287

v=0
o=CiscoSystemsSIP-GW-UserAgent 5894 8648 IN IP4 10.70.0.5
s=SIP Call
c=IN IP4 10.70.0.5
t=0 0
m=audio 16748 RTP/AVP 8 100 101
c=IN IP4 10.70.0.5
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

//4222/DB015DB18EDA/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.70.0.5:5060;branch=z9hG4bK4AB6A9;rport=57012
From: <sip:069826xxxx@10.70.0.5>;tag=9A8B458-FD2
To: <sip:A2844FD755@sip.preprod.xxx.com>;tag=358ba19e3e96a89f558cdbe342a00794.a0fb3483
Call-ID: xxx
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.preprod.xxx.com", nonce="xxx"
Content-Length: 0

//-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:A2844FD755@sip.preprod.xxx.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.70.0.5:5060;branch=z9hG4bK4AB6A9
From: <sip:069826xxxx@10.70.0.5>;tag=9A8B458-FD2
To: <sip:A2844FD755@sip.preprod.xxx.com>;tag=358ba19e3e96a89f558cdbe342a00794.a0fb3483
Date: Thu, 14 Sep 2023 09:35:20 GMT
Call-ID: xxx
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0

//4221/DB015DB18EDA/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 52.xxx.xxx.xxx:5060;branch=z9hG4bK4e59.3037d899cec6eb237cd2eeb9dfe1efe1.0,SIP/2.0/UDP 172.16.0.207;received=172.16.0.207;branch=z9hG4bK6UK5oaEr;rport=5060
From: <sip:069826xxxx@15.236.xxx.xxx>;tag=as119c492b
To: <sip:A2844FD755@52.xxx.xxx.xxx>;tag=9A8B488-1801
Date: Thu, 14 Sep 2023 09:35:20 GMT
Call-ID: xxx
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=47
Content-Length: 0

//-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:A2844FD755@10.70.0.5:57012 SIP/2.0
Via: SIP/2.0/UDP 52.xxx.xxx.xxx:5060;branch=z9hG4bK4e59.3037d899cec6eb237cd2eeb9dfe1efe1.0
From: <sip:069826xxxx@15.236.xxx.xxx>;tag=as119c492b
To: <sip:A2844FD755@52.xxx.xxx.xxx>;tag=9A8B488-1801
CSeq: 102 ACK
Call-ID: xxx
Max-Forwards: 69
Content-Length: 0

[b.winter]

If you wouldn't mask out the important information in the debugs, anybody could be able to at least read the debugs ...

But in my opinion, the incoming call is sent out immediately to the provider again.

But you still have to talk to the provider, why he is sending the username and not the number. There is nothing to do currently, to help you.

[Cr2803]

Alright, I'll talk to provider!

Initially we haven't done that because provider told us it was all OK his end,

That's why we thought issue was our conf.

Thank you so much for your help anyway!

Regards

[Cr2803]

I'll post here again in case any improvement within a few days

Cheers.

[Cr2803 · Accepted solution]

Hi,

Finally cracked it!

I started from scratch, looking at the picture from this website: https://frgtech.wordpress.com/2013/11/20/configure-voice-gateway-vg224-in-sip-with-cucm/

I had to slightly modify the configuration,

what was missing in the end was the destination-pattern 85904342 in dial-peer voice 85904442 pots (I had to rename it also, but I guess it's just a name)

voice-port 2/0
cptone FR
description analog phone
bearer-cap Speech
station-id name 85904342
station-id number 85904342
caller-id enable

[...]

dial-peer voice 1 voip
description Outgoing 10 number digits
destination-pattern ..........
session protocol sipv2
session target sip-server
session transport udp
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 3 voip
description All incoming calls
session target sip-server
incoming called-number .
voice-class codec 1
no vad
!
dial-peer voice 85904442 pots
description analog phone for SIP tests
destination-pattern 85904342
no digit-strip
port 2/0
!
!
sip-ua
credentials username 85904342 password 7 08428321C022520A0252 realm sips0.airphone.fr
registrar 2 dns:sips0.airphone.fr expires 360
sip-server dns:sips0.airphone.fr
host-registrar

Many thanks again to b.winter for willing to help and pointing us in the right direction! cheers