Hello Manish, the call flow

edwincharles · ‎05-21-2015

Hi

Strange issue which when I make call to all No. DTMF working fine just One No. when I dial DTMF not work like press 1 for sales, 2 for english

I tried to know what issue but I could not, and all No. take same Dial-peer for IN\OUT

@anyone can help me How I can troubleshoot this issue because only happen with one No.

Thank you

Manish Gogna · ‎05-21-2015

Hi,

Can you try calling that number from a different phone like your cellphone to check if the DTMF is actually working fine on that number. Once it is verified, you can post the debugs for a working DTMF call and the non-working DTMF call through the same gateway so that it can be checked what is different in the two calls. Please provide details of the exact call flow with protocol info.

HTH

Manish

edwincharles · ‎05-21-2015

Hello manish,

Yes it is working with mobile

what type of debug required please let me know

Manish Gogna · ‎05-21-2015

Please provide the call flow details. Is it SIP / H323 / MGCP?

Manish

edwincharles · ‎05-21-2015

Hello Manish,

the call flow detail as below

ip phone(sccp) -------> cucm -------> h323 ----> cube --------> SIP-----> ITSP

Manish Gogna · ‎05-21-2015

For a working and non-working call, please provide the following debugs alongwith calling and called party details and DTMF digits pressed

debug voip ccapi inout
debug ccsip all
debug voip rtp sess name

Running config

Manish

edwincharles · ‎05-31-2015

Dear Mr. Manish,

Attached logs as requested

@Regarding to this issue only happen with this No. 920001100 (this is what we find not sure for any other number) if I dialed any other No. working fine except this No.

+In attachment there is two file

-TAC Work No. press3:

Calling. 2069888

Called. 920001100

I press 3 for DTMF but not work

-TAC non-work No press3

Calling. 2069888

Called. 920000702

I press 3 for DTMF it is work

Manish Gogna · ‎05-31-2015

Hi,

I had a quick look into the debugs. For the non working DTMF call i do not see rtp-nte being advertised in the 200OK received on the gateway ( check the 'm' field )

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.68.9.178:5060;branch=z9hG4bK27DB5E7
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Call-ID: D6A080A3-12311E5-B990EEC6-7BA243B6@10.68.9.178
From: "Mohammed Munavar"<sip:2069888@10.68.9.178>;tag=4238AFD0-C5A
To: <sip:920001100@10.200.7.157>;tag=sbc080242psutch-CC-45
CSeq: 101 INVITE
Contact: <sip:920001100@10.200.7.157:5060;user=phone>
Content-Length: 137
Content-Type: application/sdp

v=0
o=- 60019922 60019922 IN IP4 10.200.7.157
s=SBC call
c=IN IP4 10.200.7.157
t=0 0
m=audio 28990 RTP/AVP 8
a=rtpmap:8 PCMA/8000

For the working call i do see it getting advertised

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bK2teptedose2uce72duadaupteT39364
From: <sip:533852289@10.68.9.178;user=phone>;tag=sbc0806cehud74c-CC-26
To: <sip:2069888@10.68.9.178;user=phone>;tag=424869E4-2240
Date: Sun, 24 May 2015 08:32:24 GMT
Call-ID: isbcha77s7pbd47o4kb44fkpo4tcohfpcckf@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:2069888@10.68.9.178:5060>
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 229

v=0
o=CiscoSystemsSIP-GW-UserAgent 6395 4087 IN IP4 10.68.9.178
s=SIP Call
c=IN IP4 10.68.9.178
t=0 0
m=audio 17936 RTP/AVP 8 97
c=IN IP4 10.68.9.178
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15

The incoming and outgoing dial-peers are the same for both calls i.e 852 and 851 respectively. You should check this with the service provider.

HTH

Manish

No. Problem ..