05-21-2015 12:28 AM - edited 03-17-2019 03:04 AM
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
05-21-2015 02:16 AM
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
05-21-2015 02:26 AM
Hello manish,
Yes it is working with mobile
what type of debug required please let me know
05-21-2015 02:30 AM
Please provide the call flow details. Is it SIP / H323 / MGCP?
Manish
05-21-2015 02:35 AM
Hello Manish,
the call flow detail as below
ip phone(sccp) -------> cucm -------> h323 ----> cube --------> SIP-----> ITSP
05-21-2015 02:54 AM
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
05-31-2015 12:28 AM
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
05-31-2015 08:54 AM
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
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide