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DTMF problems & Sangoma Card


Walter

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Hello, recently installed a Snom one plus with 2 sangoma cards on it.

 

I can receive and make calls fine.

 

Some of the DTMF work, like when pressing 1 2 and so, (just one digit. but if they dial an extension it wont go anywhere.

 

Here is the Log file of 2 different calls.

 

Log 1 INFO:

- Called into Attendant, and dialed extension 212. Nothing happened, I hanged up.

- 401 is main Attendant

- 212 is an extension.

-Attached image is the settings on 401 Attendant

-------------------------------- LOG 1 ------------------------------------------

[5] 2012/01/27 16:12:43: Identify trunk (IP address and DID match) 1

[7] 2012/01/27 16:12:43: Set packet length to 20

[6] 2012/01/27 16:12:43: Sending RTP for 08cd5ede-1dd2-11b2-b7ac-97770de55725@S1-XXXXX to 127.0.0.1:14182, codec not set yet

[5] 2012/01/27 16:12:43: Domain trunk Netborder Express@pbx.XXXXX.com sends call to 401 in domain pbx.XXXXX.com

[7] 2012/01/27 16:12:43: set_codecs: for 08cd5ede-1dd2-11b2-b7ac-97770de55725@S1-Sliquid codecs "0", codec_preference count 2

[7] 2012/01/27 16:12:43: Set packet length to 20

[6] 2012/01/27 16:12:43: Codec pcmu/8000 is chosen for call id 08cd5ede-1dd2-11b2-b7ac-97770de55725@S1-XXXXX

[7] 2012/01/27 16:12:54: Received RFC4733 DTMF on codec 101

[6] 2012/01/27 16:12:54: Received DTMF 2

[6] 2012/01/27 16:12:55: Received DTMF 1

[6] 2012/01/27 16:12:55: Received DTMF 2

--------------------------------END Log 1 ---------------------------------------

 

Log 2 INFO: (Just 1 digit option works fine)

- Called into attendant, selected Option 2 (secondary Attendant), then SELECTED option 3 (Agent Group) and it took me to a voicemail since it is the normal when no agents are logged on.

 

------------------------------ LOG 2 -------------------------------------------

[5] 2012/01/27 16:30:32: Identify trunk (IP address and DID match) 1

[7] 2012/01/27 16:30:32: Set packet length to 20

[6] 2012/01/27 16:30:32: Sending RTP for 862e1dd0-1dd2-11b2-af4c-fe9c9b9c9ee8@S1-Sliquid to 127.0.0.1:14188, codec not set yet

[5] 2012/01/27 16:30:32: Domain trunk Netborder Express@pbx.xxxxx.com sends call to 401 in domain pbx.xxxxx.com

[7] 2012/01/27 16:30:32: set_codecs: for 862e1dd0-1dd2-11b2-af4c-fe9c9b9c9ee8@S1-Sliquid codecs "0", codec_preference count 2

[7] 2012/01/27 16:30:32: Set packet length to 20

[6] 2012/01/27 16:30:32: Codec pcmu/8000 is chosen for call id 862e1dd0-1dd2-11b2-af4c-fe9c9b9c9ee8@S1-xxxxx

[7] 2012/01/27 16:30:44: Received RFC4733 DTMF on codec 101

[6] 2012/01/27 16:30:44: Received DTMF 2

[6] 2012/01/27 16:31:07: Received DTMF 3

[6] 2012/01/27 16:31:07: ACD 601: Set priority to 1

[7] 2012/01/27 16:31:07: set_codecs: for 2c8d6786@pbx codecs "", codec_preference count 7

[7] 2012/01/27 16:31:08: Call 2c8d6786@pbx: Clear last request

[7] 2012/01/27 16:31:27: Last message repeated 2 times

[7] 2012/01/27 16:31:27: Call 2c8d6786@pbx: Clear last INVITE

[5] 2012/01/27 16:31:27: INVITE Response 487 Request Terminated: Terminate 2c8d6786@pbx

[6] 2012/01/27 16:31:35: Reg 156, Sent MWI notification to user 200@pbx.xxxxx.com

-----------------------------END Log 2 -------------------------------------------

 

Please HELP, thank you.

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Can you send us the invite to both scenarios?

 

Set the following.

Log level 9

Log general events:Yes

Log SIP events:Yes

Log other messages:Yes

Log media events:Yes

 

everything else you can set to no then save.

 

Clear the log-file

 

1. make a inbound call to the AA, dial the extension

2. Call internal from an extension to the AA and dial an extension.

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OK I'm remote so I can't get to the Office phones right now.

But here is the inbound call log.

 

I'm also attaching the Trunk setup. Note that I already tried with the Out of band DTMF

 

[8] 2012/01/28 08:55:49: Last message repeated 34 times

[9] 2012/01/28 08:55:49: UDP: Opening socket on 0.0.0.0:64064

[9] 2012/01/28 08:55:49: UDP: Opening socket on 0.0.0.0:64065

[9] 2012/01/28 08:55:49: UDP: Opening socket on [::]:64064

[9] 2012/01/28 08:55:49: UDP: Opening socket on [::]:64065

[9] 2012/01/28 08:55:49: Resolve 19925: aaaa udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19925: a udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19925: udp 127.0.0.1 5066

[7] 2012/01/28 08:55:49: Set packet length to 20

[6] 2012/01/28 08:55:49: Sending RTP for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX to 127.0.0.1:14006, codec not set yet

[8] 2012/01/28 08:55:49: Incoming call: Request URI sip:2148281088@localhost:5060;transport=udp;user=phone, To is "2148281088" <sip:2148281088@localhost:5060>

[8] 2012/01/28 08:55:49: Trunk Netborder Express@pbx.MYPBX.com has country code not set, area code not set

[9] 2012/01/28 08:55:49: Incoming: formatted From is = "GRANDPRARI TX" <sip:12144486474@127.0.0.1:5066;user=phone>

[9] 2012/01/28 08:55:49: Incoming: formatted To is = "2148281088" <sip:2148281088@localhost:5060;user=phone>

[9] 2012/01/28 08:55:49: Incoming: formatted URI is = sip:2148281088@pbx.MYPBX.com:5060;transport=udp;user=phone

[8] 2012/01/28 08:55:49: Set the To domain based on To user 401@pbx.MYPBX.com

[7] 2012/01/28 08:55:49: set_codecs: for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX codecs "0", codec_preference count 2

[8] 2012/01/28 08:55:49: Play audio_moh/noise.wav

[8] 2012/01/28 08:55:49: call port 30: state code from 0 to 183

[7] 2012/01/28 08:55:49: Set packet length to 20

[9] 2012/01/28 08:55:49: update_codecs for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX: adding codec pcmu/8000 to available list

[9] 2012/01/28 08:55:49: update_codecs for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX: codec_preference size 2, available codecs size 2

[6] 2012/01/28 08:55:49: Codec pcmu/8000 is chosen for call id 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX

[9] 2012/01/28 08:55:49: Resolve 19926: aaaa udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19926: a udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19926: udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19927: aaaa udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19927: a udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19927: udp 127.0.0.1 5066

[8] 2012/01/28 08:55:49: Packet authenticated by transport layer

[8] 2012/01/28 08:55:50: Play recordings/att110.wav space20

[8] 2012/01/28 08:55:50: call port 30: state code from 183 to 200

[9] 2012/01/28 08:55:50: Resolve 19929: aaaa udp 127.0.0.1 5066

[9] 2012/01/28 08:55:50: Resolve 19929: a udp 127.0.0.1 5066

[9] 2012/01/28 08:55:50: Resolve 19929: udp 127.0.0.1 5066

[8] 2012/01/28 08:55:55: Packet authenticated by transport layer

[8] 2012/01/28 08:56:00: Last message repeated 4 times

[7] 2012/01/28 08:56:00: Received RFC4733 DTMF on codec 101

[6] 2012/01/28 08:56:00: Received DTMF 2

[8] 2012/01/28 08:56:00: call port 30: state code from 200 to 200

[8] 2012/01/28 08:56:00: Play audio_moh/noise.wav

[6] 2012/01/28 08:56:01: Received DTMF 1

[8] 2012/01/28 08:56:01: Packet authenticated by transport layer

[6] 2012/01/28 08:56:01: Received DTMF 2

[8] 2012/01/28 08:56:03: Packet authenticated by transport layer

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looks like the pbx detected the DTMF from the sangoma card.

[7] 2012/01/28 08:56:00: Received RFC4733 DTMF on codec 101

[6] 2012/01/28 08:56:00: Received DTMF 2

[8] 2012/01/28 08:56:00: call port 30: state code from 200 to 200

[8] 2012/01/28 08:56:00: Play audio_moh/noise.wav

[6] 2012/01/28 08:56:01: Received DTMF 1

[8] 2012/01/28 08:56:01: Packet authenticated by transport layer

[6] 2012/01/28 08:56:01: Received DTMF 2

[8] 2012/01/28 08:56:03: Packet authenticated by transport layer

Can you add 212 to the Direct Destinations options and dial 1.

 

Lets see if the extension can be reached.

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looks like the pbx detected the DTMF from the sangoma card.

[7] 2012/01/28 08:56:00: Received RFC4733 DTMF on codec 101

[6] 2012/01/28 08:56:00: Received DTMF 2

[8] 2012/01/28 08:56:00: call port 30: state code from 200 to 200

[8] 2012/01/28 08:56:00: Play audio_moh/noise.wav

[6] 2012/01/28 08:56:01: Received DTMF 1

[8] 2012/01/28 08:56:01: Packet authenticated by transport layer

[6] 2012/01/28 08:56:01: Received DTMF 2

[8] 2012/01/28 08:56:03: Packet authenticated by transport layer

Can you add 212 to the Direct Destinations options and dial 1.

 

Lets see if the extension can be reached.

 

No it doesn't that is the problem, it seams to get the dtmf but it doesn't go any where.

I added the trunk to my prior post probably after you replied to it

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Here is the second Auto Attendant. First one is basically pres 1 press 2 .

 

It will go anywhere in the list but not if I dial an Extension while the recording is going.

 

If you want send me a provate and i can give you IP and access if you want to take a look at the PBX

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By placing a pound sign after the direct destination (e.g., “1#”), the system will wait 3 seconds before dialing the direct destination. This is useful if you have extension numbers in the 200 range (201, 200, etc.). The 3-second delay ensures that the caller’s complete input (e.g., 212) will be processed rather than just the first digit.

 

This is a classic scenario.

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