SIP to H323 to SCCP Call are not connecting after IVR configured on Cisco Voice Gateway 3825

[ahmed-mujeeb]

Hi Everyone,

I have set a SIP trunk from a ITSP provider on Cisco 3825 router, It sas successfully registered with ITSP.

1) we can successfully made a call form internal network to PSTN.

2) But when trying to call from PSTN to internal network, IVR played but when we dialed the desired extension the ring continuous to go, hence the IP Phone rings but when anyone pick up there was no voice and that time the caller is continuously listening the ringing tone. IVR is ios based on Cisco router.

My call flow is as follows,

PSTN ==> SIP(ITSP) ==> H323(VG to CUCM) ==> SCCP(IP Phones)

Kindly Help me,

[Ayodeji Okanlawon]

OK.

pleade do this..

voice service voip

 sip
 no  outbound-proxy dns:multinet.pri:5083
  no listen-port non-secure  5083

dial-peer voice 100 voip
description **** Outgoing Call to Multinet SIP Trunk ****
translation-profile outgoing remove_prefix
destination-pattern 0.T
session protocol sipv2
session target sip-server
session transport udp
voice-class codec 1 
dtmf-relay rtp-nte sip-notify
no vad
!

voice service voip

sip

sip-ua

sip-server dns:multinet.pri:5083

Test again and send me the ff from your gateway

debug ccsip mess

debug voip ccapi inout

[ahmed-mujeeb]

debugs are attached.. but i am not able to call any outgoing number and also not able to call to ITSP

[Ayodeji Okanlawon]

Ok. i dont see any sip messages in this call...

What is the calling and called number..

Can you attach your current configuration please.?

[ahmed-mujeeb]

called number 703122401592

calling number 4199

running config is attaced

[Ayodeji Okanlawon]

OK. This call didnt make it to the logs. I dont see anything about the call in those logs..

What we need to do now is this.. (ensure you do this when the gateway is not so busy)

service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit

Then..

<Enable debugs, then test again.>

debug ccsip all

<Enable session capture to txt file in terminal program.> (such as Putty)

then do the ff:

terminal length 0
show logging

Next..

dial-peer voice 110 voip

no shut

dial-peer voice 111 voip

no shut

do another test call and send the logs. include call details

[Ayodeji Okanlawon]

Also please add this command back..

voice service voip

sip

 localhost dns:multinet.pri:5083

[ahmed-mujeeb]

Hi Ayodeji,

after adding local host command back, sip trunk is in working now, i have shifted some route patterns on sip trunk for testing, hope i will not hear any issue. further i will update you tomorrow.... thank you soo much for your help.. really really apreciated...

[Ayodeji Okanlawon]

Glad to help. Dont forget to rate useful posts

[ahmed-mujeeb]

OK Ayodeji... when i call from my internal network to any mobile number... mobile rings.. but if i cancel the call from mobile... call doesn't disconnect from the ip phone and it continuous ringing and then the call automatically take another path i.e PSTN to again ring the mobile phone and this continue untill all possible options finished..

[Ayodeji Okanlawon]

OK.

Again please send me a debug ccsip message. include calling and called number. Is this the only issue you have now, Does call work okay through the sip trunk?

[ahmed-mujeeb]

Hello, Actually I am facing one more issue, we have multiple branches all over the country, so the issue is when any branch try to transfer or make conference the call drops, the issue was that all branches are using g.729 to complete the call and the sip trunk to itsp is using g.711 and I don't have enough resources to configure transcoder so for now i changed the branhes codec to g711..

attached are the ccsip mesages for the call I have one ITSP sip trunk and two pri, so first call take itsp when call canceled from mobile phone it automatically takes another path and then third path..

callled number : 03122401592

calling number : 4199 (cipc)

[Ayodeji Okanlawon]

Ahmed,

You will have to speak to your ITSP.

Two things you need to tell them..

1. Your ITSP is sending both 180 ringing and 183 session progress for ringing.

They need to send only one..

2. When you cancel the call, your ITSP needs to send "487 cancel" but they are sending 480 temporarily unavailable..

BURJ-VGW#
290888: Oct 31 00:42:11.362: //509749/4CE6AD800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.99.3.102:5060;branch=z9hG4bK4852E22BA
From: <sip:multinet.pri>;tag=9700EE70-1A35
To: <sip:03122401592@multinet.pri>;tag=J*9kWWNVOV
Call-ID: 23D62F72-7E7511E5-B5549F63-3301D68D@multinet.pri:5083
CSeq: 102 INVITE
Timestamp: 1446234128
Supported: timer
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:03122401592@125.209.93.196:5083;transport=udp>
Content-Length: 0
User-Agent: TELES.C5/5.0.7.4
Allow-Events: talk

290889: Oct 31 00:42:11.366: //509748/4CE6AD800000/SIP/Msg/ccsipDisplayMsg:
Sent:

BURJ-VGW#SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.20.11.11:5060;branch=z9hG4bK45686b35738e
From: <sip:4199@172.20.11.11>;tag=13878~18d2dfe6-ba54-48c4-b16e-0053a74e5282-43901612
To: <sip:903122401592@172.20.11.50>;tag=9700FABC-987
Date: Fri, 30 Oct 2015 19:42:08 GMT
Call-ID: 4ce6ad80-6331c810-2350-b0b14ac@172.20.11.11
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: kpml, telephone-event
Remote-Party-ID: <sip:03122401592@multinet.pri:5083>;party=called;screen=no;privacy=off
Contact: <sip:903122401592@172.20.11.50:5060>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

BURJ-VGW#
290890: Oct 31 00:42:14.922: //509749/4CE6AD800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.99.3.102:5060;branch=z9hG4bK4852E22BA
From: <sip:multinet.pri>;tag=9700EE70-1A35
To: <sip:03122401592@multinet.pri>;tag=J*9kWWNVOV
Call-ID: 23D62F72-7E7511E5-B5549F63-3301D68D@multinet.pri:5083
CSeq: 102 INVITE
Timestamp: 1446234128
Supported: timer
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:03122401592@125.209.93.196:5083;transport=udp>
Content-Length: 0
User-Agent: TELES.C5/5.0.7.4
Allow-Events: talk

290891: Oct 31 00:42:14.926: //509748/4CE6AD800000/SIP/Msg/ccsipDisplayMsg:
Sent:

BURJ-VGW#SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.20.11.11:5060;branch=z9hG4bK45686b35738e
From: <sip:4199@172.20.11.11>;tag=13878~18d2dfe6-ba54-48c4-b16e-0053a74e5282-43901612
To: <sip:903122401592@172.20.11.50>;tag=9700FABC-987
Date: Fri, 30 Oct 2015 19:42:08 GMT
Call-ID: 4ce6ad80-6331c810-2350-b0b14ac@172.20.11.11
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: kpml, telephone-event
Remote-Party-ID: <sip:03122401592@multinet.pri:5083>;party=called;screen=no;privacy=off
Contact: <sip:903122401592@172.20.11.50:5060>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

BURJ-VGW#
290892: Oct 31 00:42:19.646: //509749/4CE6AD800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.99.3.102:5060;branch=z9hG4bK4852E22BA
From: <sip:multinet.pri>;tag=9700EE70-1A35
To: <sip:03122401592@multinet.pri>;tag=J*9kWWNVOV
Call-ID: 23D62F72-7E7511E5-B5549F63-3301D68D@multinet.pri:5083
CSeq: 102 INVITE
Timestamp: 1446234128
Supported: timer
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:125.209.122.194:5083>
Content-Length: 0
User-Agent: TELES.C5/5.0.7.4
Allow-Events: talk
Reason: Q.850;cause=31;text="Normal, unspecified"

This is the cause of the issue you see. So please escalate to your ITSP. They have not implemented their SIP solution correctly

[ahmed-mujeeb]

But I cannot understand that this is only happening when I configured a sip trunk between CUCM and Voice Gateway before that when I was using H.323 gateway this was not happening. further I don't understand how ITSP can control my gateway as when the caller disconnect the call how ITSP send to take another PRI which is POTS to reach the destination. I think the issue is with our configuration.

these all are my opinions.

[Ayodeji Okanlawon]

Well....

SIP and H323 are totally different protocols and they operate differently. SIP to SIP and SIP to H323 is a different game entirely.

My opinion is what I have already mentioned to you. Its up to you how you want to proceed with it. I could be wrong, but I can only tell you what I see in the logs.

Good luck with it.

[ahmed-mujeeb]

Hi Ayodeji,

I am not an expert though I am only telling you my opinion, I have one more setup with BE6K, the same issue is coming there also all the config are up to the mark and that ITSP is different than this, I am only asking if there is any option in CUCM so from where we can tune the SIP trunk and It can be resolved, one thing I am not getting is the call go through E1 line and the third one is another E! PRI, now I am only asking if this is possible that ITSP message 183 can cause rerouting of call through another path? Thank you for all your precious time in solving my issue..