Solved: UC500 no calls through sip trunk - Page 2

sam2000 · ‎02-01-2024

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 · ‎02-12-2024

Hi Carlo, of course, my subscription is "Fastweb casa full", the connection is FTTC with third party router

Thanks

Paolo

Carlo Poggiarelli · ‎02-05-2024

Hi Paolo,

Please add

ip address trusted list
ipv4 85.18.217.0/24

Please let us know

Regards

Carlo

Please rate all helpful posts "The more you help the more you learn"

Roger Kallberg · ‎02-05-2024

@Carlo Poggiarelli your missing the top part of the configuration. The whole thing for reference to the OP.

voice service voip
 ip address trusted list
  ipv4 85.18.217.0/24

Carlo Poggiarelli · ‎02-05-2024

Hi @Roger Kallberg

Yes I didn’t mention all the config but just the piece to modify ‘:)’

Thanks

Carlo

Please rate all helpful posts "The more you help the more you learn"

sam2000 · ‎02-09-2024

Hi Carlo, I added the address of the proxy server as per your suggestion. It still doesn't work, but it seems that the problem is not the same. Here is the debug log (ccsip messages and ccsip error)

000122: Jan 2 12:14:55.134: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:03xxxxxxx7_MyPublicVoipNumber@192.168.1.88:5060 SIP/2.0
Via: SIP/2.0/UDP 85.18.217.100:5060;branch=z9hG4bKvt804e1018v8u9ce7ep0.1
Call-ID: 00545d3e00545d2-001e-0117-0000-0000@10.2.22.207
CSeq: 1 INVITE
To: <sip:03xxxxxxx7_MyPublicVoipNumber@10.247.53.6;lr>
Content-Type: application/sdp
Max-Forwards: 61
Supported: 100rel
Allow: UPDATE,INVITE,BYE,PRACK,MESSAGE,NOTIFY,INFO,ACK,SUBSCRIBE,OPTIONS,CANCEL,REFER
P-Charging-Vector: icid-value="q8ztNSd1sQ6y0086";icid-generated-at=10.247.5.40;orig-ioi=10.247.5.40
Accept: application/sdp,application/isup,application/xml
Contact: <sip:33xxxxxxx0_cellphone@85.18.217.100:5060;transport=udp>
From: <sip:33xxxxxxx0_cellphone@telecomitalia.it>;tag=f11ada5f
Content-Length: 270

v=0
o=HPE-AS 7553 1 IN IP4 85.18.217.108
s=IMSS
c=IN IP4 85.18.217.108
t=0 0
m=audio 10166 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

SIP: (12) Attribute mid, level 1 instance 1 not found.
000123: Jan 2 12:14:55.138: //12/37EB97B8800E/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
000124: Jan 2 12:14:55.138: //12/37EB97B8800E/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
000125: Jan 2 12:14:55.138: //12/37EB97B8800E/SIP/Error/sipSPI_ipip_update_call_entry:
failed to update call entry
000126: Jan 2 12:14:55.138: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 12
000127: Jan 2 12:14:55.138: //12/37EB97B8800E/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.
000128: Jan 2 12:14:55.146: //12/37EB97B8800E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.18.217.100:5060;branch=z9hG4bKvt804e1018v8u9ce7ep0.1
From: <sip:33xxxxxxx0_cellphone@telecomitalia.it>;tag=f11ada5f
To: <sip:03xxxxxxx7_MyPublicVoipNumber@10.247.53.6;lr>
Date: Mon, 02 Jan 2006 12:14:55 GMT
Call-ID: 00545d3e00545d2-001e-0117-0000-0000@10.2.22.207
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

000129: Jan 2 12:14:55.150: //12/37EB97B8800E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.18.217.100:5060;branch=z9hG4bKvt804e1018v8u9ce7ep0.1
From: <sip:33xxxxxxx0_cellphone@telecomitalia.it>;tag=f11ada5f
To: <sip:03xxxxxxx7_MyPublicVoipNumber@10.247.53.6;lr>;tag=DF0F4-183F
Date: Mon, 02 Jan 2006 12:14:55 GMT
Call-ID: 00545d3e00545d2-001e-0117-0000-0000@10.2.22.207
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=1
Content-Length: 0

000130: Jan 2 12:14:55.158: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:03xxxxxxx7_MyPublicVoipNumber@192.168.1.88:5060 SIP/2.0
Via: SIP/2.0/UDP 85.18.217.100:5060;branch=z9hG4bKvt804e1018v8u9ce7ep0.1
CSeq: 1 ACK
Call-ID: 00545d3e00545d2-001e-0117-0000-0000@10.2.22.207
To: <sip:03xxxxxxx7_MyPublicVoipNumber@10.247.53.6;lr>;tag=DF0F4-183F
Max-Forwards: 61
From: <sip:33xxxxxxx0_cellphone@telecomitalia.it>;tag=f11ada5f
Content-Length: 0

looks like there's some issue with CHANNEL_COUNT but I can't figure it out

Thanks

Paolo

Maren Mahoney · ‎02-09-2024

The channel count part may also need looking at, but I'm more concerned with the 404 Not Found error that your system sent back to the PSTN.

The number in the inbound INVITE is "03xxxxxxx7_MyPublicVoipNumber", but the numbers on the voice ports are in 3-digit format. I don't see (in the config you posted earlier) where you are taking the PSTN-form of your number and truncating it to three digits.

I think you may have intended that with your voice translation-profile FW_INBOUND_CID, but instead of translating the called party (from anything to 622) you are translating the caller ID. This means your router is trying to match the 03xxxxxxx7 to something internal and isn't finding a match.

May I suggest that you edit the voice translation profile for inbound calling and try the call again?

voice translation-profile FW_INBOUND_CID
  translate called 10002

Maren

sam2000 · ‎02-09-2024

OMG Thanks Maren, that was it! I wrote "calling" instead of "called".

Now the incoming calls do work. Still no luck on outbound though

Thanks

Paolo

sam2000 · ‎02-09-2024

Now the incoming calls go through, but I still can't place outgoing calls.

This is the debug (ccsip messages and ccsip error) when I try to place an outbound call:

000171: Jan 2 12:40:08.297: //45/B20F498F800F/SIP/Error/sipSPI_ipip_set_history_info_header: ccb->src_addr_str is NULL
SIP: (45) Group (a= group line) attribute, level 65535 instance 1 not found.
000172: Jan 2 12:40:08.305: //45/B20F498F800F/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:33xxxxxxx0_PSTN_CalledNumber@voip.fastwebnet.it:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.88:5060;branch=z9hG4bKB103B
From: <sip:03xxxxxxx7_MyPublicVoipNumber@192.168.1.88>;tag=250788-4F6
To: <sip:33xxxxxxx0_PSTN_CalledNumber@voip.fastwebnet.it>
Date: Mon, 02 Jan 2006 12:40:08 GMT
Call-ID: BDD5E91C-7AC311DA-8066C52A-75F680A@192.168.1.88
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2987346319-2059604442-2148492288-4049517024
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1136205608
Contact: <sip:03xxxxxxx7_MyPublicVoipNumber@192.168.1.88:5060>
Expires: 900
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 8682 7927 IN IP4 192.168.1.88
s=SIP Call
c=IN IP4 192.168.1.88
t=0 0
m=audio 19420 RTP/AVP 8 18 19
c=IN IP4 192.168.1.88
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:19 CN/8000

000173: Jan 2 12:40:08.321: //45/B20F498F800F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.88:5060;branch=z9hG4bKB103B
From: <sip:03xxxxxxx7_MyPublicVoipNumber@192.168.1.88:5060>;tag=250788-4F6
To: <sip:33xxxxxxx0_PSTN_CalledNumber@voip.fastwebnet.it>
Call-ID: BDD5E91C-7AC311DA-8066C52A-75F680A@192.168.1.88
CSeq: 101 INVITE
Content-Length: 0

000174: Jan 2 12:40:08.321: //45/B20F498F800F/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.88:5060;branch=z9hG4bKB103B
From: <sip:03xxxxxxx7_MyPublicVoipNumber@192.168.1.88:5060>;tag=250788-4F6
To: <sip:33xxxxxxx0_PSTN_CalledNumber@voip.fastwebnet.it>;tag=aprqngfrt-u5usei30000a6
Call-ID: BDD5E91C-7AC311DA-8066C52A-75F680A@192.168.1.88
CSeq: 101 INVITE
Content-Length: 0

000175: Jan 2 12:40:08.325: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:33xxxxxxx0_PSTN_CalledNumber@voip.fastwebnet.it:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.88:5060;branch=z9hG4bKB103B
From: <sip:03xxxxxxx7_MyPublicVoipNumber@192.168.1.88>;tag=250788-4F6
To: <sip:33xxxxxxx0_PSTN_CalledNumber@voip.fastwebnet.it>;tag=aprqngfrt-u5usei30000a6
Date: Mon, 02 Jan 2006 12:40:08 GMT
Call-ID: BDD5E91C-7AC311DA-8066C52A-75F680A@192.168.1.88
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

Roger Kallberg · ‎02-09-2024

Not related to your issue, but on dial peer 10 you do not need to have this command “session target sip-server”. It is used on outgoing dial peers and as your intent with dial peer 10 is inbound it’s not needed. Apart from that when you post output from your system please attach that as text file(s) instead of posting it directly in the post as that makes it much easier for anyone to read and work with your output.

sam2000 · ‎02-10-2024

Thanks, i removed that part. And sorry for not attaching files, for some weird reasons yesterday Cisco's website didn't let me.

Carlo Poggiarelli · ‎02-10-2024

Hi,

Can you please post you entire config?

Fastweb provider usually authenticate the user also with the IP address so you need to present the call with the right IP address and with the authorized calling party number (don't remeber if they need in a E.164 format or local one)

Let us know

Regards

Carlo

Thanks

Please rate all helpful posts "The more you help the more you learn"

sam2000 · ‎02-10-2024

Hi Carlo, I attached the config. Consider that in the same LAN I have an IP phone (cisco CP6851) that works with Fastweb, the ALG is performed by the router but I suppose I can check some parameter in the phone config, if this helps?

thanks

Paolo

Roger Kallberg · ‎02-11-2024

From what I can see you still do not have anything that defines what this refers to on your outbound dial peer.

session target sip-server

Please correct this and also look into putting the registration configuration that you currently have under sip-ua into a tenant configuration as you’ll need to have the router acting as a registrar server for your SIP phones to be operational with your router. This tenant is then used on both your dial peers that is facing your service provider. The link that I provided earlier has the needed information for you to do this. Also I see that you still have not changed how you match the dial peer for inbound calls, please do so as previously advised. Again the provided link to the document has the necessary details on this. Last thing, under your global voice service section you have two lines that has H.323 in them, you can remove them as none of them are needed.

sam2000 · ‎02-11-2024

you are right Rogert it's my fault, I tried following your advice about setting up the lan properly but ended up mixing things. I am posting now the correct config with working inbound calls.

Paolo

Roger Kallberg · ‎02-11-2024

It’s really hard to help you as you’re not following the advice given. If you want help with this please follow the advice you get, otherwise we can’t help you.