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UC560 translation rule help

localhosta
Level 1
Level 1

Hi guys,

Our UC560 is working fine with incoming calls, but when I add forward to number got this error:

*Apr 13 09:57:32.089: //-1/7140017D81A4/RXRULE/sed_subst: Successful substitution; pattern=02xxxxx717 matchPattern=02xxxxx7 replacePattern=7 replaced pattern=717
*Apr 13 09:57:32.089: //-1/7140017D81A4/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
*Apr 13 09:57:32.089: //-1/7140017D81A4/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
*Apr 13 09:57:32.089: //-1/7140017D81A4/RXRULE/regxrule_profile_translate_internal: xlt_number=717 xlt_type=unknown xlt_plan=unknown
*Apr 13 09:57:32.093: //-1/7140017D81A4/RXRULE/regxrule_profile_translate_internal: number=004xxxxxxx type=unknown plan=unknown numbertype=redirect-target
*Apr 13 09:57:32.093: //-1/7140017D81A4/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
*Apr 13 09:57:32.093: //-1/7140017D81A4/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-target number not found
*Apr 13 09:57:32.093: //-1/7140017D81A4/RXRULE/regxrule_profile_translate_internal: No match: number=004xxxxxxx type=unknown plan=unknown

I am calling 02xxxx717 number and should be redirected to 004xxxxxxx but the call just dropped.

Am I missing something here?

Cheers,

Ivan

 

13 Replies 13

b.winter
VIP
VIP

What is your protocoll you are using? SIP, ISDN, H.323?
Take a debug of a full call and post it together with the config.

The debug you posted only shows the output for translation, you won't see anything in there why the call is not working.

localhosta
Level 1
Level 1

Is this enough for debbugging or should add more?

UC560#show debugging
DIALPEER:
debug voip dialpeer error call is ON (filter is OFF)
debug voip dialpeer error software is ON
debug voip dialpeer inout is ON (filter is OFF)
voip translation-rule
voip translation debugging is on (Filter is OFF)

CCSIP SPI: SIP Call Message tracing is enabled (filter is OFF)
CCSIP SPI: SIP media debug tracing is enabled (filter is OFF)

 

 

If you have SIP:
debug ccsip messages
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74
debug voice translation

should be enough for the first logs.

localhosta
Level 1
Level 1

Attaching sanitized config and debug.

I've replaced numbers:

MY_MOBILE - calling from this number

SIP_IP_ADDR - our VOIP provider IP

REDIRECT_PHONE - where call is redirected to

0211111700 - our changed number.
Basically when there is no redirection call is accepted (incoming call). Once I add redirect - nothing.

 

So, you are calling from external this number 0211111700.
But what are you trying to achieve? What do you mean with "redirect"?
Do you want to forward the call back to PSTN directly on the router?

And is your rule 1 in the translation-rule 10 working correctly?
Do you want to translate 0211111700 to 700?

Which router are you using? UC560 is only the hostname, and not a router type.
And maybe you should think about upgrading the firmware of your router too.

localhosta
Level 1
Level 1

This is the model:

Cisco UC560-T1E1-K9 (MPC8378) processor (revision 0x100) with 497664K/26624K bytes of memory

I am calling this number 0211111700, translated to internal phone 700.

Internal phone 700 is forwarded to external but this does not work.
Normal incoming calls are accepted.

Maybe then you have to talk to your provider:

Here, you are telling the provider, that the number is redirect to the 0REDIRECT_PHONE number.

*Apr 13 11:28:45.893: //105/2FE2ED7081C2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP SIP_IP_ADDR;rport;branch=z9hG4bKtQy1BKNcD4a4g
From: "0MY_MOBILE" <sip:0MY_MOBILE@SIP_IP_ADDR>;tag=U4F5QSDFFSy7e
To: <sip:0211111700@OUR_IP_ADDR>;tag=5DB6F10-2C
Date: Thu, 13 Apr 2023 11:28:45 GMT
Call-ID: 67d8bee3-5493-123c-2291-3a14ce3d8aae
CSeq: 66139345 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Diversion: <sip:700@OUR_IP_ADDR>;reason=unconditional;counter=1
Contact: <sip:0REDIRECT_PHONE@SIP_IP_ADDR>
Content-Length: 0

And then the provider accepts this redirect:

*Apr 13 11:28:45.897: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0211111700@OUR_IP_ADDR SIP/2.0
Via: SIP/2.0/UDP SIP_IP_ADDR;rport;branch=z9hG4bKtQy1BKNcD4a4g
Max-Forwards: 34
From: "0MY_MOBILE" <sip:0MY_MOBILE@SIP_IP_ADDR>;tag=U4F5QSDFFSy7e
To: <sip:0211111700@OUR_IP_ADDR>;tag=5DB6F10-2C
Call-ID: 67d8bee3-5493-123c-2291-3a14ce3d8aae
CSeq: 66139345 ACK
Content-Length: 0

I have never worked with a UC560 router/uc server. So I don't know how call forwards are working there.
Normally in CUCM, when a phone has set a forward, there is a 2nd call going out to the forwarded number.
E.g.
1st call: Phone A --> Phone B
2nd call: Phone B forwards --> Phone C

And normally on a standalone SIP-router (CUBE), I disable the feature "sip move-temporarily" and "sip refer"
voice service voip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer

But I don't know if that's also applicable to a UC560 system.

localhosta
Level 1
Level 1

Tried those configs but no changes.

Thanks for that hints. I will speak with them tomorrow as well and will update you.

 

localhosta
Level 1
Level 1

With those two settings in the voip service saw extra things in the logs and fixed it

*Apr 13 13:25:13.868: //-1/750DD00D81D6/RXRULE/sed_subst: Successful substitution; pattern=0MY_MOBILE matchPattern=.* replacePattern=0211111& replaced pattern=02111110MY_MOBILE

Sent:
INVITE sip:0REDIRECT_PHONE@SIP_IP_ADDR:5060 SIP/2.0
Via: SIP/2.0/UDP 10.28.3.10:5060;branch=z9hG4bK10E46
From: "MY_MOBILE" <sip:02111110MY_MOBILE@10.28.3.10>;tag=6460FDC-1E3
To: <sip:0REDIRECT_PHONE@SIP_IP_ADDR>

As you can see in the logs translation rules 71 was applied and that gave an errors.

Basically this rule is to strip 0 for incoming and for outgoing to add company number (0211111).

If you call from internal number 700 people will see 0211111700.
But here full mobile is added at the end and number sent to provider is: 021111110MY_MOBILE

Removed this translation rule and worked. Will test more tomorrow as here is too late for this.

Thanks a lot for your help.

 

 

As long as a call is going out, you just have to make the correct number translations.

localhosta
Level 1
Level 1

Now the call is made, but nothing is heard. Awaiting more information from our provider, but for incoming calls it's ok.

Not sure is this relevant to the problem:
*Apr 14 04:05:17.469: //124/5D68C0058215/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:873968F0
*Apr 14 04:05:17.469: //125/5D6A947D821A/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8A7B2320

 

Probably a know problem with calls from external, which are forwarded back to external.

See here for example:
https://community.cisco.com/t5/unified-communications-infrastructure/cube-config-sip-trunk-to-deutsche-telekom/td-p/4447706

But your question now has nothing to with your original question.
You now have a media problem, your orig. question was about signalling. 2 different kind of things. So maybe it would be better to open a new topic, if you still have problems after you talked to the provider.

localhosta
Level 1
Level 1

Now I've added those settings in the config:

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
ADDED redirect ip2ip
sip
ADDED session transport tcp
  registrar server
  no update-callerid

 

 

And voila ... call was ok. Tested from local numbers and was nice and clear.

From international number got error:

see TRYING:

SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.28.3.10:5060;branch=z9hG4bK505C7;rport=36237

....

Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.28.3.10:5060;branch=z9hG4bK505C7;rport=36237
Max-Forwards: 33

 

Perhaps will have to work more on the translation rules...