Outgoing Calls with SIP Trunk fail

[Solomon Rufus Kavala]

I have a SIP trunk from a TSP terminated on the CUBE. - c3925 gateway connected via H323 & SIP to a CUCM 8.5 cluster. Incoming calls are working fine, but outgoing calls fail with "SIP/2.0 488 Not Acceptable Media" and also with Warning: 304 10.180.174.1 "Media Type(s) Unavailable" from CCSIP Debugs. I have attached Voice configuration, Dial-peers configured in the gateway and also debug of Outgoing call. Any general advise would be much appreciated.

Thanks,

Solomon.

[Solomon Rufus Kavala]

I have modified the Translation rule for Outgoing calls and tested couple of calls.But I still get Request Timed Out from ITSP.Please find the debugs attached.

Thanks

Solomon.

[Ayodeji Okanlawon]

Solomon,

OK everything looks okay on your side. However VzB SBC is not responding to the invite from you. Contact them and find out why they are not responding to your invites

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Solomon Rufus Kavala]

Thank you very much for you help.Will contact VzB and will let you know the outcome.

[Solomon Rufus Kavala]

Hello Sir,

I contacted VzB on this Issue and they said that PAI takes precedence over the diversion, which takes precedence over the From. Then they looked in to their database to find the reason for rejections and corrected their database.

Now when I call out,Calls are getting connected .But I did not hear either ringbacak tone or audio in the call.I have also changed the region settings for SIP Trunk calls from G729r8 to G711.Still no Luck.I have attached the debugs of the connected call.Please help me out,whether any thing I need to correct from my end.

One more information,Verizon Business Tech asked me to add signalling target on the outbound dial-peer(100).

This is what VzB tech sent in his mail:

"You  need to make sure that:

CUBE Address:                                  10.180.174.1                      

                You are signaling to Verizon at:  172.31.134.49:5170

                For now, stop using the P-Asserted, and DH Header until we get calls flowing

                Also, lets remove the leading 1 and just send the 10 digit number.

                Your TN’s are:                                    972-538-0226     Test DID

                                                                                972-538-0227     Screening TN."

I have removed Voice Class SIP Profiles from my configuration,hope that was the PAI which VzB asked me to remove.

Configuration which has been removed from my configuration:

"voice class sip-profiles 100

request INVITE sip-header Remote-Party-ID modify "sip:(.*)@[172.31.134.49]+" "sip:\1@diebold.globalipcom.com" voice class sip-profiles 100
request INVITE sip-header Remote-Party-ID modify "sip:(.*)@[172.31.134.49]+" "sip:\1@diebold.globalipcom.com"

But still no luck.Call is getting connected,but no audio in the connected call.

Thanks,

Solomon.

[Ayodeji Okanlawon]

Solomon,

I have looked at the logs and here is what I see..

1. On the ringback issue: VzB is sending 183 with SDP.

Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.180.174.1:5060;branch=z9hG4bK6B1B73
From: "Solomon Kavala 80226" <9725380226>;tag=537507D0-2137
To: <13304903845>;tag=662308445-1357609661452
Call-ID: A7ACB58F-586B11E2-B99A99C3-5100E663@10.180.174.1
CSeq: 101 INVITE
Timestamp: 1357609417
Supported:
Contact: <13304903845>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 219

v=0
o=BroadWorks 40595214 1 IN IP4 172.31.134.164
s=-
c=IN IP4 172.31.134.164
t=0 0
m=audio 11100 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20

By default, Cisco Unified Communications Manager  will signal the calling phone to play local ringback if SDP is not  received in the 180 or 183 response. If SDP is included in the 180 or  183 response, instead of playing ringback locally, Cisco Unified  Communications Manager will connect media, and the calling phone will  play whatever the called device is sending (such as ringback or busy  signal).

This then suggests that If you do not receive ringback, the device to which you are  connecting may be including SDP in the 180/183 response, but it is not  sending any media before the 200OK response.

If ITSP sends 183 with SDP, ITSP has the responsibility to play the ringback or any audio, for the CUBE or CUCM, there is only one thing to do which is to cut through media.

So on your ringback you still need to go back to Verizon, they are not playing any media to you.

2. On the Media issue:

Verizon are telling you to connecto to RTP/listen to media on thie followign ip address

c=IN IP4 172.31.134.164

Do you have ip connectivity to this ip address?

I also think that your ringback issue is related to this, because this is where VzB is sending you media (both ringback and RTP).

You need to have ip connectivity to this ip address.

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[Solomon Rufus Kavala]

Thank you for your reply. IP 172.31.134.164 is VzB SBC's Media IP.Issue has been resolved,seems VzB has corrected their database from their end.Now I am able to call Local and Long Distance calls with in US,but while dialing International calls,I got an announcement from announciator stating "Your call cannot be completed as dialed,Please check the number and dial again".As you have asked me to confugre an Incoming dial-peer that matches calls to ITSP using SIP Protocol,I have tried to configure but never worked. But we already have a dial-peer configured for International calling (011T),so can you guide me to configure and dial international calls as well. I have posted successful output debug and also CUBE's configuration.

Thanks,

Solomon        

[Ayodeji Okanlawon]

Solomon,

I am glad its working now. Can you check the CSS on the phone. Ensure that it has access to the partiton of the Route pattern for international calls. Send me debugs again. Only debug ccsip messages...

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Solomon Rufus Kavala]

I have alredy checked CSS on softphone and found it had access to dial International calls. Please find the debugs attached,also I have configured dial-peer 4002 as an incoming dialpeer to ITSP using SIP.                 

[Ayodeji Okanlawon]

Your call is matching the dial-peer 8801. So its not going over the sip trunk.

To have it go over the sip trunk, you should configure a dial-peer like this

dial-peer voice 102 voip

preference 1

description OUTBOUND G711 Voice SIP calls to VzB

translation-profile outgoing Outgoing-Translate

destination-pattern 011T

session protocol sipv2

session target ipv4:172.31.134.49:5170

voice-class sip early-offer forced

dtmf-relay rtp-nte

codec g711alaw

no vad

!

dial-peer voice 8801 pots

preference 2----------------------------------ensure you add this to this dial-peer

destination-pattern 011T

progress_ind alert enable 8

progress_ind progress enable 8

progress_ind connect enable 8

port 0/3/1:1

!

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Solomon Rufus Kavala]

     I have added outbound dial-peer 102 for International calls using SIP.Also gave preference 2 to pots dial-peer 8801.Still call got failed.

[Ayodeji Okanlawon · Accepted solution]

First of all,

The call is sent incorrectly to your ITSP..

*Jan  8 11:19:56 UTC: //8578/E14686800000/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:11919959470707@172.31.134.49:5170 SIP/2.0---------------the leading 0 is  missing from the call

Via: SIP/2.0/UDP 10.180.174.1:5060;branch=z9hG4bK961D7F

Remote-Party-ID: "Solomon Kavala 80226" <9725380226>;party=calling;screen=yes;privacy=off

From: "Solomon Kavala 80226" <9725380226>;tag=5584A984-0

To: <11919959470707>

Date: Tue, 08 Jan 2013 11:19:56 GMT

voice translation-rule 3

rule 1 /^80226/ /9725380226/

rule 2 /^7\(...\)/ /4695497\1/

voice translation-profile Outgoing-Translate

translate calling

*Jan  8 11:19:56 UTC: //8578/E14686800000/SIP/Msg/ccsipDisplayMsg:

+++The reason is because of this translation rule+++++++

voice translation-rule 4

rule 1 /^0\(.*\)/ /\1/

!

voice translation-profile Outgoing-Translate

translate calling 3

translate called 4

This rule is applied to the dial-peer 102 and is stripping the "0". I dont know why you are stripping the 0..

If you need to strip the "0" for other calls to work, then you need to configure a new translation profile for international calls like this...

voice translation-profile Outgoing-Intl----------------------------------(NB "I"=capital letter for i)

translate calling 3

Then apply this to the dial-peer 1002 like this..

dial-peer voice 102 voip

preference 1

description OUTBOUND G711 Voice SIP calls to VzB

translation-profile outgoing Outgoing-Intl

The second this is that your dial-peer config is still not correct...Because the call is matching dial-peer 8801 first..

Please send me your config again.

List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=8801

     2: Dial-peer Tag=102

     3: Dial-peer Tag=100

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Solomon Rufus Kavala]

   Thank you very much for all your help.As soon as I have changed the Translation Profile,call started working.But I want to tune dial-peer 4002as a generic dial-peer,which is an incoming dial-peer to ITSP using SIP .Please find the config attached and suggest me for one last time.

[Ayodeji Okanlawon]

Assuming all calls coming from cucm is sent to your sip trunk then you can remove all DP 4000, 4001, 4002 and have only one dp as foolows

dial-peer voice 4000 voip

session protocol sipv2

incoming called-number.

codec g711alaw

dtmf-relay rtp-nte

no vad

[Solomon Rufus Kavala]

As you already know, we are using H323 along with SIP. So there were already couple of dialpeers with incoming called-number.

We already have dialpeer 4444 which is a H323 dialpeer,which routes incoming calls from CUCM to ITSP.

[Ayodeji Okanlawon]

In that case, there is no way to configure a generic dial-peer. You will have to leave it as it is.

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson