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3rd party SIP Phone - can receive, but cant make calls

josepaulopetry
Level 1
Level 1

Hello,

I just registered a 3rd Party SIP Phone on CUCM 9.1, i can receive calls but im having problems to dial from those phones.

Attached is a trace from CUCM..

Thanks

15 Replies 15

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

Jose,

The call is failing with 404 not found. Does extension 454 exist? If it does does the CSS on phone 450 has access to the partition of 454?

Outgoing SIP TCP message to 192.168.242.106 on port 1025 index 10020

[595451,NET]

SIP/2.0 404 Not Found

Via: SIP/2.0/TCP 192.168.242.106;branch=z9hG4bKc_18e63-38262923513a85fa_I454

From: sip:454@teltec.net;tag=124f15bd5138f75e513a8ffe_F454192.168.242.106

To: sip:450@teltec.net;tag=246115~c7880263-3c24-4ff5-a1a9-0183cf090450-24206735

Date: Thu, 07 Mar 2013 20:23:58 GMT

Call-ID:

b_18e5f-6efde915513a85c0_I@192.168.242.106

CSeq: 12 INVITE

Allow-Events: presence

Reason: Q.850;cause=1

Content-Length: 0

Outgoing SIP TCP message to 192.168.242.106 on port 1025 index 10020

[595451,NET]

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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yes, 454 exists, and 450 has a CSS that access the PT of 454...

Jose,

Did you get a chance to implement my suggestions? is this resolved now?

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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Ayodeji Okanlawon
VIP Alumni
VIP Alumni

Jose,

So looking closely into your traces here is what I see...

Once CUCM receives the Invite for a call setup, it does digit analysis using tyhe process (DaReq)

|1,100,13,10395.50^192.168.242.106^*      |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0]  CI=24206735 Fqdn=ti=1nd=454pi=0si1 Cgpn=tn=0npi=0ti=1nd=454User=Host=teltec.netPort=0PassWord=Madder=Transport=4mDisplayName=RawUrl=pi=1si0 DialedNum=tn=0npi=1ti=1nd=450User=450Host=teltec.netPort=5060PassWord=Madder=Transport=4mDisplayName=RawUrl=sip:450@teltec.netpi=0si1 requestID=0 DigitAnalysisComplexity=0 CallingUser= IgnoreIntercept=0

CUCM has determined that the FQDN of the called number is this...

user=454

host=teltec.net

The first question is this..Can CUCM resolve "teltec.net" CUCM needs to resolve the host portion of the uri so it can know where to send the call to.

We see immediately after the DaReq (digit analysis request) We get a response back from DaRes:

07458748.000 |17:23:58.841 |SdlSig   |DaRes                                  |setup_da                       |Cdcc(1,100,212,12252)            |Da(1,100,204,1)                  |1,100,13,10395.50^192.168.242.106^*      |[R:N-H:0,N:0,L:0,V:0,Z:0,D:0] CI=24206735 Block NoPotentialMatchesExist OnNetpatternUsage =2requestID =0

And then CUCM disconnects the call:

07458752.000 |17:23:58.841 |SdlSig   |CcDisconnReq

Here are the pointers:

!. who is testec.net

2. Can CUCM resolve testec.net to an ip address

NB: I dont even see CUCM doing a sipdns to resolve the host portion of the address...

So check your DNS config and ensure CUCM can resolve the host portion of the URI.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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hello aokanlawon

thanks for you notes... ill only be able to test on thursday

just a question:

1. who is teltec.net

this is my internal domain... cucm.teltec.net for example is the FQDN of CUCM

2. Can CUCM resolve teltec.net to an ip address

So the question is, what entry should I have on DNS for teltec.net??

So you need to make some adjustments...teltec.net is the main domain ..you need to route to a host on that domain...

e.g 454@cucm.teltec.net

then configure A records for cucm in the teltec domain on your DNS..ensure cucm has DNS server configured

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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ok, will test this on thursday then i update here

tks

hello Jose

can you share the process of registering. avaya 96xx series phone regiatration process with CUCM.? 

I have cucm 9.1 and trying to register it as basic sip 3rd party device.

need to understand that where in avaya phone we can put in digest credentials for authentication with cucm for regiatration.

michel.perez
Level 1
Level 1

The problem was only the Subscribe Calling Search Space, on the past versions it was not necessary at my point of view.

Subscribe calling search space..That is used for presence....How does that affect call routing? Dont get it?

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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Yes, because we are using Avaya phones, and everything is different for Avaya, so my tests wasn´t working, so i registered one X-Lite Softphone with the same credentials and changing some things, the Sip X-Lite worked, so i made the same configuration at Avaya phones.

Michel,

Do you work with Jose? Can you send me the traces for a working call. I am surprised about subscribe css

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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yes, Michel works with me...

Firstly he changed SIP domain configuration (in the Avaya phone), from name to IP addres of CUCM... after this change we tested again and still could not make a call

Then we changed some codec configuration in the configuration file of the phone, tried again and still the same problem

Then applied a CSS to Subscribe CSS and now we have an Avaya phone registered as 3rd party SIP device on CUCM 9.1, making and receiving calls, and also transfering and creating conference. We are now testing anothers features..

Attached are the requested debug of a working call from Avaya phone (DN 454) to a Cisco phone (DN 494)...

Hi Jose,

as per your post, we have the Dallas delta sip phone that trying to register with cucm 11.5..but incoming working and outgoing fails when dail..local, internal or outside..
please share what change you did in phone or in cucm to make it working...
i have applied subscribe css but still outgoing not working...
thanks
Warm Regard's
Amit Sahrma