Digital Reception Menu not working from outside calls

[Cody Fontaine]

In most cases, DTMF problems are the result of incorrect DTMF Method settings (using gateways), or the use of low bit-rate codecs when attempting to use audio DTMF.

Not using gateways.

 

[Cody Fontaine]

Here is the sip info from a call that doesn't work:
08/31/2017 4:47:17 PM - L:211.2[Ivr:HOL] got Connected.UAC Recv 200/INVITE from 127.0.0.1:5483 tid=405f19792a1fa270 Call-ID=zW-HgDtB3lE0ZfAR4MvZSQ..: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-524287-1---405f19792a1fa270;rport=5060 Contact: <sip:[email protected]:5483;rinstance=174c0483b6d80430> To: <sip:[email protected]>;tag=4b65ed55 From: "WESTBOROUGH MA"<sip:[email protected]:5060;nf=e>;tag=e8315340 Call-ID: zW-HgDtB3lE0ZfAR4MvZSQ.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CX IVR Content-Length: 170 v=0 o=3cxVCE 46540650 312106290 IN IP4 127.0.0.1 s=3cxVCE Audio Call c=IN IP4 127.0.0.1 t=0 0 m=audio 12002 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000

 

[Cody Fontaine]

Here is the sip info from one that does:

08/31/2017 5:27:53 PM - L:220.2[Ivr:HOL] got Connected.UAC Recv 200/INVITE from 127.0.0.1:5483 tid=a25c6423b96eaa69 Call-ID=S2kOS1cEaydfH-1Wqj5NTQ..: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-524287-1---a25c6423b96eaa69;rport=5060 Contact: <sip:[email protected]:5483;rinstance=174c0483b6d80430> To: <sip:[email protected]>;tag=aa29a719 From: "WESTBOROUGH MA"<sip:[email protected]:5060;nf=e>;tag=b077f91b Call-ID: S2kOS1cEaydfH-1Wqj5NTQ.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CX IVR Content-Length: 303 v=0 o=3cxVCE 219975555 83143575 IN IP4 127.0.0.1 s=3cxVCE Audio Call c=IN IP4 127.0.0.1 t=0 0 m=audio 12012 RTP/AVP 0 8 9 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000

 

[Cody Fontaine]

SIP info from call with DTMF not working:

08/31/2017 5:28:19 PM - L:221.2[Ivr:HOL] got Connected.UAC Recv 200/INVITE from 127.0.0.1:5483 tid=9f51832ecb4f5410 Call-ID=bF9rEYCL_IQlIVqSDwDegg..: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-524287-1---9f51832ecb4f5410;rport=5060 Contact: <sip:[email protected]:5483;rinstance=174c0483b6d80430> To: <sip:[email protected]>;tag=10485677 From: "WESTBOROUGH MA"<sip:[email protected]:5060;nf=e>;tag=d2119b7d Call-ID: bF9rEYCL_IQlIVqSDwDegg.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CX IVR Content-Length: 304 v=0 o=3cxVCE 228160290 343894665 IN IP4 127.0.0.1 s=3cxVCE Audio Call c=IN IP4 127.0.0.1 t=0 0 m=audio 12014 RTP/AVP 0 8 9 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000

Sip info from call with DTMF working:

08/31/2017 5:27:53 PM - L:220.2[Ivr:HOL] got Connected.UAC Recv 200/INVITE from 127.0.0.1:5483 tid=a25c6423b96eaa69 Call-ID=S2kOS1cEaydfH-1Wqj5NTQ..: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-524287-1---a25c6423b96eaa69;rport=5060 Contact: <sip:[email protected]:5483;rinstance=174c0483b6d80430> To: <sip:[email protected]>;tag=aa29a719 From: "WESTBOROUGH MA"<sip:[email protected]:5060;nf=e>;tag=b077f91b Call-ID: S2kOS1cEaydfH-1Wqj5NTQ.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CX IVR Content-Length: 303 v=0 o=3cxVCE 219975555 83143575 IN IP4 127.0.0.1 s=3cxVCE Audio Call c=IN IP4 127.0.0.1 t=0 0 m=audio 12012 RTP/AVP 0 8 9 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000

 

[nb]

The fixes are all in there. We will need to check deeper.
And the extract you provided will show nothing about why it's failing.

Just out of curiosity, Set this to 0 and restart all services and try again. Does it work or not?
Then set it back to one and start making captures of the verizon and the comcast calls. So we can see what is happening.

 

[Cody Fontaine]

The fixes are all in there. We will need to check deeper.
And the extract you provided will show nothing about why it's failing.

Just out of curiosity, Set this to 0 and restart all services and try again. Does it work or not?
Then set it back to one and start making captures of the verizon and the comcast calls. So we can see what is happening.

Set it to 0 and restarted, still does not work. how do you want me to upload the captures? The whole pcap or copy and paste?