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DTMF Issue between Yeastar TG100 and Cisco CUCM

georgeotj
Level 1
Level 1

‎07-06-2018 03:59 AM - edited ‎03-17-2019 01:09 PM

Hi,

 

I have a working SIP Trunk between Yeastar TG100 GSM Gateway and Cisco CUCM as follow :

 

TG100 ip address = 172.68.9.41

Cisco CUCM ip address = 172.68.9.31

 

TG100 -- {sip trunk} -- Cisco CUCM -- Cisco VoIP Phones (extension 10xxx)

 

I can successfully make a call from Cisco VOIP Phones to outside office via TG100, and

I can successfully make a call from outside office to Cisco VoIP Phones via TG100 if i directly forward call to extension number 10xxx

 

Problem is that, when i make call from outside to TG100, i can get the Cisco CUCM IVR (extension 41000) when i forward the call to it but cannot pass the DTMF, so it did not response when i press anything from IVR Menu

 

Attached is the DTMF config from TG100 and SIP Trunk at Cisco CUCM. and below is the log from TG100

 

Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK6acb369e
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.68.9.41>;tag=as3af18067
To: <sip:172.68.9.31>
Contact: <sip:Unknown@172.68.9.41>
Call-ID: 5ac4415144d75c0862f8938278d7c4b3@172.68.9.41
CSeq: 102 OPTIONS
User-Agent: TG100
Date: Fri, 06 Jul 2018 10:21:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:172.68.9.31:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK6acb369e
From: "Unknown" <sip:Unknown@172.68.9.41>;tag=as3af18067
To: <sip:172.68.9.31>;tag=946573330
Date: Fri, 06 Jul 2018 10:21:41 GMT
Call-ID: 5ac4415144d75c0862f8938278d7c4b3@172.68.9.41
Server: Cisco-CUCM12.0
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK4f73705f
Max-Forwards: 70
From: "+62xxxxxxxxxx" <sip:+62xxxxxxxxxx@172.68.9.41>;tag=as62de9927
To: <sip:41000@172.68.9.31:5060>
Contact: <sip:+62xxxxxxxxxx@172.68.9.41>
Call-ID: 247536582a6d41b93fe7b31a57561221@172.68.9.41
CSeq: 102 INVITE
User-Agent: TG100
Date: Fri, 06 Jul 2018 10:21:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 334

v=0
o=root 632790171 632790171 IN IP4 172.68.9.41
s=Asterisk PBX 1.6.2.6
c=IN IP4 172.68.9.41
t=0 0
m=audio 10692 RTP/AVP 0 8 3 9 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:172.68.9.31:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK4f73705f
From: "+62xxxxxxxxxx" <sip:+62xxxxxxxxxx@172.68.9.41>;tag=as62de9927
To: <sip:41000@172.68.9.31:5060>
Date: Fri, 06 Jul 2018 10:21:54 GMT
Call-ID: 247536582a6d41b93fe7b31a57561221@172.68.9.41
CSeq: 102 INVITE
Allow-Events: presence
Content-Length: 0


Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK4f73705f
From: "+62xxxxxxxxxx" <sip:+62xxxxxxxxxx@172.68.9.41>;tag=as62de9927
To: <sip:41000@172.68.9.31:5060>;tag=10193674~b160b51f-a124-41da-88f6-b2410e9f8a63-27998731
Date: Fri, 06 Jul 2018 10:21:54 GMT
Call-ID: 247536582a6d41b93fe7b31a57561221@172.68.9.41
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Server: Cisco-CUCM12.0
Call-Info: <sip:172.68.9.31:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-ID: 2e8788b50fa1c2e03f3380ab10193675;remote=312f7925e058f3858fb6d5ab10193674
P-Preferred-Identity: <sip:40000@172.68.9.31>
Remote-Party-ID: <sip:40000@172.68.9.31>;party=called;screen=no;privacy=off
Contact: <sip:41000@172.68.9.31:5060>
Content-Length: 0


Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK4f73705f
From: "+62xxxxxxxxxx" <sip:+62xxxxxxxxxx@172.68.9.41>;tag=as62de9927
To: <sip:41000@172.68.9.31:5060>;tag=10193674~b160b51f-a124-41da-88f6-b2410e9f8a63-27998731
Date: Fri, 06 Jul 2018 10:21:54 GMT
Call-ID: 247536582a6d41b93fe7b31a57561221@172.68.9.41
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence, kpml
Supported: replaces
Server: Cisco-CUCM12.0
Call-Info: <sip:172.68.9.31:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uas
Require: timer
Session-ID: 2e8788b50fa1c2e03f3380ab10193675;remote=312f7925e058f3858fb6d5ab10193674
P-Preferred-Identity: <sip:40000@172.68.9.31>
Remote-Party-ID: <sip:40000@172.68.9.31>;party=called;screen=no;privacy=off
Contact: <sip:41000@172.68.9.31:5060>;automata
Content-Type: application/sdp
Content-Length: 225

v=0
o=CiscoSystemsCCM-SIP 10193674 1 IN IP4 172.68.9.31
s=SIP Call
c=IN IP4 172.68.9.33
b=TIAS:64000
b=AS:64
t=0 0
m=audio 18346 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK7759cb23
Max-Forwards: 70
From: "+62xxxxxxxxxx" <sip:+62xxxxxxxxxx@172.68.9.41>;tag=as62de9927
To: <sip:41000@172.68.9.31:5060>;tag=10193674~b160b51f-a124-41da-88f6-b2410e9f8a63-27998731
Contact: <sip:+62xxxxxxxxxx@172.68.9.41>
Call-ID: 247536582a6d41b93fe7b31a57561221@172.68.9.41
CSeq: 102 ACK
User-Agent: TG100
Content-Length: 0

 

[2018-07-06 17:21:55] DEBUG[2763] rtp.c: Ooh, format changed from unknown to ulaw

[2018-07-06 17:21:55] DEBUG[2763] rtp.c: Created smoother: format: 4 ms: 20 len: 160

[2018-07-06 17:21:55] NOTICE[2763] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 172.68.9.33
BYE sip:41000@172.68.9.31:5060 SIP/2.0
Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK0bc175ee
Max-Forwards: 70
From: "+62xxxxxxxxxx" <sip:+62xxxxxxxxxx@172.68.9.41>;tag=as62de9927
To: <sip:41000@172.68.9.31:5060>;tag=10193674~b160b51f-a124-41da-88f6-b2410e9f8a63-27998731
Call-ID: 247536582a6d41b93fe7b31a57561221@172.68.9.41
CSeq: 103 BYE
User-Agent: TG100
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:172.68.9.31:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK0bc175ee
From: "+62xxxxxxxxxx" <sip:+62xxxxxxxxxx@172.68.9.41>;tag=as62de9927
To: <sip:41000@172.68.9.31:5060>;tag=10193674~b160b51f-a124-41da-88f6-b2410e9f8a63-27998731
Date: Fri, 06 Jul 2018 10:22:15 GMT
Call-ID: 247536582a6d41b93fe7b31a57561221@172.68.9.41
Server: Cisco-CUCM12.0
CSeq: 103 BYE
Content-Length: 0

 

 

Could anyone please help give me a hint where to check this issue.

 

Thanks.

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R0g22
Cisco Employee
Cisco Employee

‎07-06-2018 08:37 AM

Keep the DTMF on CUCM to "No Preference".
Also, when you say outside, how are the outside calls reaching CUCM ?
From the logs, I see RFC 2833 but I also see CUCM highlighting KPML as part of "Allow Events" header. This is in the INVITE with CSeq 102. Not sure why is this a second INVITE since this is not a reINVITE.

georgeotj
Level 1
Level 1
In response to R0g22

‎07-06-2018 09:02 AM

Ok, i have config the DTMF on CUCM to "No Preference" and do reset, but still same result.

 

Outsite calls can reach CUCM because i create route for incoming calls to be forwarded to CUCM via the SIP Trunk, also there is a parameter at TG100 where i can set specificly which extension to be the destination (dumpscreen attached)

 

extension 41000 is the CUCM IVR

 

Thanks Nipun.

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