CLI Intermittent issue on CUBE router

[sarwarm123]

Hi Akon as you know, we recently configured CUBE (ip2ip) gateway for the new SIP link but caller-id is not working properly. It works for some outbound calls and does not work for some outbound calls.

I noticed a pattern when it CLI works and and when it doesn't work in the ccsip debug. I have attached ccsip messages debug results. Please help to figure out issue.

[sarwarm123]

Hi Aok,

Shall I send you more logs or the one I posted above are ok?

[Ayodeji Okanlawon]

Muhammad,

I dont know what to say. It works on one CUBE and diesnt work on the other, This cant be config related. It only affects mobile numbers. Please remove the anonymous config and use the config on CUBE-A...dont forget to change the ip address...Tell your provider what you are experiencing...This has to be their issue

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[sarwarm123]

Hi Aok,

I also think its a problem at service provider end. I have logged an incident see whay they reply. Will let you know what they reply. Thanks for your help

[sarwarm123]

Hi Aok,

Yeah I do agree with you, I have logged with service provider, see what they responce back.

[sarwarm123]

Hi Aok,

Just got reply from the service provider regarding this issue.

The discrepancy between the two endpoints is due to the firmware version differing on each on the SBC’s that the endpoints reside. 

Going forward, I can only suggest you send a P-Asserted Identity header with ‘Privacy: Full’, alternatively

So instead of send Remote Party ID header with privacy full they are demanding to send P-Asserted Identity header with Privacy Full

[Ayodeji Okanlawon · Accepted solution]

Ok..

Please disable remote-party id and try this profile

voice class sip-profiles 3

request INVITE sip-header P-Asserted-Identity "<>" "<>;privacy=full"

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[sarwarm123]

Hi Aok,

CLI witheld issue has not been resolved yet, yesterday when I tested P-Asserted sip profile it was actually going out from the CUBE-A router instead of CUBE-B.

I have attached ccsip messages and ccapi inout debug. I can see this error "Reason: Q.850;cause=127" Also in the log file I can also see in log

P-Asserted-Identity: "Muhammad Sarwar" <44043>;privacy=full

Is that service provider issue?

voice class sip-profiles 4

request INVITE sip-header P-Asserted-Identity modify "<>" "<>;privacy=full"

dial-peer voice 2 voip

description *** Outbound LANDLINE calls to ITSP ***

destination-pattern 0[1-37].........

session protocol sipv2

session target sip-server

voice-class codec 1

voice-class sip asserted-id pai

voice-class sip profiles 4

dtmf-relay rtp-nte

ip qos dscp cs4 media

ip qos dscp cs3 signaling

no vad

!

dial-peer voice 3 voip

description *** Outbound 0845-0870 calls to ITSP ***

destination-pattern 0[8][047][0458]......

session protocol sipv2

session target sip-server

voice-class codec 1

voice-class sip asserted-id pai

voice-class sip profiles 4

dtmf-relay rtp-nte

ip qos dscp cs4 media

ip qos dscp cs3 signaling

no vad

[Ayodeji Okanlawon]

Ok,

Looks like the P-Asserted-Identity header is not properly formatted..

Your provider dosent like it...I will look and see how it should be formatted

*Apr  9 09:53:20.112: //3357/97437D000001/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 400 Bad Request

Via: SIP/2.0/UDP 10.60.34.106:5060;branch=z9hG4bKC55C5B

From: "Muhammad Sarwar" <44043>;tag=1797B4B0-E1F

Call-ID:

24D16B65-A03211E2-AAFFB442-D4C147C4@10.60.34.106

CSeq: 101 INVITE

To: <07967999481>;tag=3574490123-862955

Content-Length: 0

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[Ayodeji Okanlawon · Accepted solution]

NB: syntaxt error in the previous post, so I deleted it..Use this one..

I looked into the RFC for P-Assreted-Identity and it looks like you need a seperate Privacy field to use Privacy with PAI..

So lets give this a go..

on CUBE B

Please remove the sip proflie for the P-Assersted -Identity..add the below to the profile..

voice class sip-profiles 4

request INVITE sip-header Privacy add "Privacy: id"

Please check your CLI..if the Privacy header  is in the logs and it still doesnt work then try

voice class sip-profiles 4

request INVITE sip-header Privacy add "Privacy: full"

Please test again and send me debug ccsip messages...

Pls make sure you remove the previous profile..

Leave all the other configs

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[sarwarm123]

Hi Aok,

Fantastic Its working now, I tried the privacy:id in the header and that worked. I can also see in the log Privacy: id

voice class sip-profiles 4

request INVITE sip-header Privacy add "Privacy: id"

Once again bundle of thanks for your help

[sarwarm123]

Hi Aok,

While testing SIP trunk I noticed we cannot dial anymore 8 digit numbers without STD code for example 976XXXXXX. To resolve that I added another dial-peer and also added another translation rule to add STD code.

But I dont know where should I apply this, I mean in which direction incoming or outgoing?

for example user dial a number 976123456 it should strip 9 and add 024 infornt of that

voice translation-rule 25

rule 1 /^9\(........\)/ /024\1/

dial-peer voice 4 voip

description *** Outbound Without STD calls to ITSP ***

destination-pattern 76......

session protocol sipv2

session target sip-server

voice-class codec 1

voice-class sip asserted-id pai

voice-class sip profiles 4

dtmf-relay rtp-nte

ip qos dscp cs4 media

ip qos dscp cs3 signaling

no vad

[Ayodeji Okanlawon]

Here you go..

voice translation-rule 25

rule 1 /^9\(........\)/ /024\1/

voice translation-profile STD

translate called 25

dial-peer voice 4 voip

destination-pattern 976......    (this must match the dialled number before stripping the 9)

translation-profile outgoing STD

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[sarwarm123]

Hi Aok,

I have applied this config but its not working. One thing more we are striping 9 on router level not from the CUCM. When I see the logs I can see 9 is not stripping off and also cannot find Outgoing dial-peer

I have attacehd logs

voice translation-rule 100

rule 1 /^9/ //

voice translation-profile STRIP9

translate called 100

dial-peer voice 101 voip

description *** Inbound Calls from ITSP ***

translation-profile incoming STRIP9

session protocol sipv2

session target sip-server

incoming called-number .

voice-class codec 1

dtmf-relay rtp-nte

[Ayodeji Okanlawon]

Please send me your updated sh run again (attach it here)

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[sarwarm123]

Hi Aok,

Did you find any thing suspicious in a config?