SIP Trunk, Cisco 3825 and CUCM 6.1

[Alexis Sulbaran]

I have a SIP trunk from a provider terminated on my 3825 gateway which is connected via h323 to my CUCM 6.1 cluster.

I am not able to place outbound calls.

Any help would be much appreciated.

Thanks,-

Alexis                  

[Alexis Sulbaran]

CX-REN-VGW01#sh ver

Cisco IOS Software, 3800 Software (C3825-IPVOICEK9-M), Version 12.4(20)T1, RELEASE SOFTWARE (fc3)

Technical Support:

http://www.cisco.com/techsupport

[Carlo Poggiarelli]

Ok.
That's why you don't have that command ( introduced on 15.1)
Anyway try to check again with your provider why they are rejecting your call.

Let me know

Regards

Carlo

Sent from Cisco Technical Support iPhone App

[Ayodeji Okanlawon]

Looking at the disconnect cause code of 57..This implies bearer capability not authorized. I suggest tha you offer your provider more than one codec...You are offering then only G711ulaw..I though you had voice class codec on your dial-peer. It looks like you have removed it and hard coded g711ulaw. I suggest you hard code g711alaw to your dial-peer and try again and also ensure that you include g711alaw in your voice class codec..

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

[Alexis Sulbaran]

I did not remove the voice class codec. I added the codec g711alaw to the voice class as sugggested but still unable to place a call. See config below.

voice class codec 1

codec preference 1 g711ulaw bytes 160

codec preference 2 g711alaw bytes 160

codec preference 3 g729br8 bytes 20

codec preference 4 g729r8 bytes 20

!

dial-peer voice 901 voip

voice-class codec 1

incoming called-number 2T

dtmf-relay h245-signal h245-alphanumeric

no vad

!

dial-peer voice 900 voip

destination-pattern 2T

voice-class codec 1

voice-class sip profiles 1

session protocol sipv2

session target ipv4:66.249.145.144

incoming called-number 7248...

[Ayodeji Okanlawon]

Ok, try this..

voice service voip

sip

early-offer forced

Then send me only debug ccsip messages

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

[Alexis Sulbaran]

Ok, made the requested change. Find attached the debug ccsip messages.

[Ayodeji Okanlawon]

OK, something is not quite right..The CUBE is not offering all the codecs to the ITSP...Please renove the voice class codec on the dial-peer to ITSP ( I believe its 900) and hard code codec g711alaw..

eg..

dial-peer voice 900 voip

no voice-class codec 1

codec g711alaw...

send me debug cccsip messages again

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

[Alexis Sulbaran]

Changed the config as requested, I am unable to call and not getting any ccsip messages. I think my client is set to g729.

Also tried changing my client to g711 but still no ccsip messages.It looks like my client (softphone) is only talking g711ulaw.

[Ayodeji Okanlawon]

You should check the region seeting between your softphone and the gateway..ensure it is set to 64Kbps...

Ok..please send the full logs now..debug h225 asn1 and debug h245 asn1 as well as debug voip ccapi inout

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

[Alexis Sulbaran]

Here you go, dial-peer config below. Debug messages attached but not getting ccsip messages.

dial-peer voice 900 voip

destination-pattern 2T

voice-class sip profiles 1

session protocol sipv2

session target ipv4:66.249.145.144

incoming called-number 7248...

codec g711alaw

[Ayodeji Okanlawon]

Can you check on CUCM if you have fast start configured? What codec is fast start set to use? Can you change this to G711alaw?

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

[Alexis Sulbaran]

Yes, it was set to fas start with g711ulaw, changed it g711alaw. Still unable to call but I am now getting ccsip messages (see attached).

[Ayodeji Okanlawon · Accepted solution]

Alexis,

Now you need to speak to ITSP. We have tried a few things and they are still rejecting your call.

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

[Alexis Sulbaran]

Its working, it was an issue with the SIP account. The ITSP corrected the issue with the account.

Now, is it possible to configure multiple SIP accounts to the same ITSP as we want all our companies to dial out using their own SIP account.