UC500 no calls through sip trunk

[sam2000]

Hello, I am running a problem with the UC500 platform and I am hoping for some advice.

I'm using a UC540W as a small pbx with a variety of internal extensions. The uc540 is connected on a private LAN with NAT access to the Internet, the NAT router also performs SIP ALG.
Directyly attached fxs and isdn phones as well as SIP phones in LAN do work as internal extensions, they can make calls each other.
I have a sip trunk configured (this trunk is working if directly configured on a sip phone on the LAN), but I can't have incoming or outgoing calls on it through the uc540

The sip trunk configuration is this:

sip-ua
credentials number MY_PUBLIC_NUMBER username MY_VOIP_USERNAME password MY_VOIP_PWD realm voip.fastwebnet.it
authentication username MY_VOIP_USERNAME password MY_VOIP_PWD realm voip.fastwebnet.it
no remote-party-id
retry invite 4
timers expires 900000
timers register 100
registrar 2 dns:voip.fastwebnet.it expires 3600 auth-realm voip.fastwebnet.it
connection-reuse

on this I configured two dial-peers, one for incoming calls and one for outgoing calls:

dial-peer voice 11 voip
description *** SIP Trunk towards fastweb ***
translation-profile outgoing FW_OUTBOUND_CID
destination-pattern 33T
session protocol sipv2
session target sip-server
session transport udp
voice-class codec 1
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
ip qos dscp cs5 media
ip qos dscp cs5 signaling
!
dial-peer voice 10 voip
description *** SIP Trunk Fastweb incoming ***
translation-profile incoming FW_INBOUND_CID
session protocol sipv2
session target sip-server
session transport udp
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad

translations rules and translation profiles are as follow (incoming calls from sip trunk should ring extension 622):

voice translation-rule 10001
rule 1 /^.*/ /MY_PUBLIC_NUMBER/
!
voice translation-rule 10002
rule 1 /^.*/ /622/
!
!
voice translation-profile FW_INBOUND_CID
translate calling 10002
!
voice translation-profile FW_OUTBOUND_CID
translate calling 10001

when i run the command
sh sip-ua register status

I can see the connection with the sip proxy is registered:

--------------------- Registrar-Index 2 ---------------------

Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
MY_PUBLIC_NUMBER -1 399 yes

but I can't get any call through this trunk, neither incoming nor outgoing.
I have enabled debug ccsip message, if I call my SIP number from the cellphone I get these debug messages:

000979: Jan 2 14:12:18.644: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:MY_PUBLIC_NUMBER@192.168.1.88:5060 SIP/2.0
Via: SIP/2.0/UDP 85.18.217.100:5060;branch=z9hG4bKmuvim300d865281h6bn0.1
Call-ID: 12dad1a06c2ea01-000f-00ba-0000-0000@10.2.22.199
CSeq: 1 INVITE
To: <sip:MY_PUBLIC_NUMBER@10.247.53.6;lr>
Content-Type: application/sdp
Max-Forwards: 61
Supported: 100rel
Allow: UPDATE,OPTIONS,INFO,REFER,ACK,NOTIFY,INVITE,CANCEL,SUBSCRIBE,MESSAGE,PRACK,BYE
P-Charging-Vector: icid-value="oXcbX51DqfpZ0088";icid-generated-at=10.247.5.40;orig-ioi=10.247.5.40
Accept: application/sdp,application/isup,application/xml
Contact: <sip:MY_CELLPHONE_NUMBER@85.18.217.100:5060;transport=udp>
From: <sip:MY_CELLPHONE_NUMBER0@telecomitalia.it>;tag=582e2469
Content-Length: 271

v=0
o=HPE-AS 47819 1 IN IP4 85.18.217.108
s=IMSS
c=IN IP4 85.18.217.108
t=0 0
m=audio 10156 RTP/AVP 8 18 101
b=AS:80
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn:0
a=cdsc:1 image udptl t38
a=sendrecv
a=maxptime:20
a=ptime:20

000980: Jan 2 14:12:18.656: //101/9E2E82A4807B/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.18.217.100:5060;branch=z9hG4bKmuvim300d865281h6bn0.1
From: <sip:MY_CELLPHONE_NUMBER@telecomitalia.it>;tag=582e2469
To: <sip:MY_PUBLIC_NUMBER@10.247.53.6;lr>
Date: Mon, 02 Jan 2006 14:12:18 GMT
Call-ID: 12dad1a06c2ea01-000f-00ba-0000-0000@10.2.22.199
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 85.18.217.100:5060;branch=z9hG4bKmuvim300d865281h6bn0.1
From: <sip:MY_CELLPHONE_NUMBER@telecomitalia.it>;tag=582e2469
To: <sip:MY_PUBLIC_NUMBER@10.247.53.6;lr>;tag=7969E8-960
Date: Mon, 02 Jan 2006 14:12:18 GMT
Call-ID: 12dad1a06c2ea01-000f-00ba-0000-0000@10.2.22.199
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=21
Content-Length: 0

000982: Jan 2 14:12:18.668: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:MY_PUBLIC_NUMBER@192.168.1.88:5060 SIP/2.0
Via: SIP/2.0/UDP 85.18.217.100:5060;branch=z9hG4bKmuvim300d865281h6bn0.1
CSeq: 1 ACK
Call-ID: 12dad1a06c2ea01-000f-00ba-0000-0000@10.2.22.199
To: <sip:MY_PUBLIC_NUMBER@10.247.53.6;lr>;tag=7969E8-960
Max-Forwards: 61
From: <sip:MY_CELLPHONE_NUMBER@telecomitalia.it>;tag=582e2469
Content-Length: 0

I am obviously either missing something or made some misconfiguration, I am at loss here and I don't know what it could be.

Any suggestion is warmly welcome, thanks.

Paolo

[sam2000]

Hi Roger, I admit I didn't think at the tenant option, I suppose it could help because my final configuration would have two sip trunks. But unfortunately the config command voice class tenant is not supported on the UC500. Nice idea though.

Thanks

Paolo

[Carlo Poggiarelli]

Hi ,

I'm not talking about ALG ';)' but what Fastweb accept as calling number or IP.

So please attach 2 separated files with a debug ccsip message (please post the entire log) one  from a working phone and another from a non working one.

Please let us know

Regards

Carlo

[sam2000]

Hi Carlo,

here is the current config, the debug log from the UC540 trying to call my cellphone and the debug log from the IP phone making a successful call to the same phone. Mind that the IP phone doesn't run with IOS so the debug log is a bit different from the UC500.

thanks

Paolo

[Carlo Poggiarelli]

Hi Paolo,

Please send a complete debug ccsip of an outgung call. also attach a debug  voip dialpeer inout. 

Thanks 

Carlo

[sam2000]

Hi Carlo,

here is a log with debug ccsip all and debug voip dialpeer inout

thanks

Paolo

[Carlo Poggiarelli]

Hi Paolo,

I see that you are sending the Station Name (on FXS Port) to fastweb as calling name.

On Voice-port 0/0/0 modify the "station-id name" parameter with your public number and try again.

Please let me know

Regards

Carlo

[sam2000]

Hi Carlo,

changing station-id name didn't work: I tried but it was no use. I kind of already figured the provider wanted the public number and not the extension's, that's why I created the translation-rule 10001 and translation-profile FW_OUTBOUND_CID, whereas by changing the station-id name in the voice port calling other extension send the public number instead of the internal one. Besides, in the future I should have two SIP trunks (one with fastweb and another with asterisk and gsm dongle) so binding the extensions to only one public number won't be my goal

Searching around the web, I found a list of parameters suggested for VoIP connections to fastweb, but I am unsure where to configure some of them, here they are:

• SIP voip.fastwebnet.it:5060 UDP
• PRACK RFC 3262
• OPTIONS support
• REGISTER TIME 3600s
• INVITE Expires 430s or more
• Codec G729, G729A, G711 A-law
• VAD (Voice Activity Detection) disabled
• Packetization 20ms
• DTMF in band RFC2833
• Fax T38
• QoS SIP and RTP: DSCP CS5
• QoS data: DSCP 0

If you know fastweb you probably know how scarce is their support related to this issue

Thanks

Paolo

[Carlo Poggiarelli · Accepted solution]

Hi ,

With Paolo we found the solution.

In the invite message the provider requires his domain in the invite message.

In the outgoing dialpeer we shoul apply the following sip profile

voice class sip-profiles 105
request INVITE sip-header From modify "<sip:(PUBLICNUMBER)@(.*)>" "<sip:\1@voip.fastwebnet.it>"

Also in global configuration, being the UC500 behind a NAT, to make audio working in both directions, the following missing commands should be applied :

voice service voip

sip

asymmetric payload full
early-offer forced
midcall-signaling passthru

session refresh

HTH

Regards

Carlo

[sam2000]

Carlo you are a genius, that did the trick!

Paolo

[Maren Mahoney]

Paolo,

I am SO GLAD you found a solution to this. It was a tricky one. Props to @Carlo Poggiarelli for the final bits!

Maren

[Carlo Poggiarelli]

Thank you Paolo for your kind words.

Glad that helped and thanks for nice rating

Cheers

Carlo