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5 posts

Wannabe Geek


Topic # 18001 21-Dec-2007 16:20
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Finished adding a VFX line to an existing Asterisk box, and must say many thanks to all for the new guides, it only took about 20 minutes to get up and running.  Call quality is great, better than the old POTS lines.

Have encountered a strange problem with VFX/Asterisk and DTMF not being sent to the far end.

When I call a service that requires DTMF i.e Bank/Telecom/IVR Menus on another PABX, etc, the other end does not respond or times out

i.e.  if I call 0800 000 000 (telecom information) via VFX/Asterisk I can't selected any menus, works fine from using a POTS/Asterisk line


If I call the VFX voice portal from the VFX/Asterisk line, all the menus there work fine.


I called the VFX help desk, they have checked their end, and all seems fine, the guy at the other end asked me to dial some numbers, and could not hear anything at his end.


If I call from VFX to POTS and keys are pressed on the VFX phone nothing is heard at the POTS end, keys pressed at POTS end give a short burst of DTMF on the VFX phone when the key is released.

Calls from VFX to VFX customers, no DTMF tones are heard at the opposite end when keys are pressed.

Calls from POTS to VFX seem to be OK, when keys are pressed and released on either phone, a burst of the corresponding DTMF is heard at the other end.


Software etc...
Trixbox 1.0.0.30
FreePBX 2.3.1.3
Asterisk 1.2.24

VFX over TelstrClear ADSL

Anyone with ideas, information or comments?

Cheers,
  Craig

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  Reply # 101268 21-Dec-2007 16:46
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What do you have the DTMF setting set to in your extension settings & in the phone?

The safest option is to use dtmfmode=rfc2833 if you're using dtmfmode=inband you'll strike problems unless you're using ulaw/alaw since the tones won't work very well with g729.



5 posts

Wannabe Geek


  Reply # 101269 21-Dec-2007 16:52
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Hi, just using the standard settings from the guide,

You haven't seen this issue?  Thought it may be me, but have a friend with a similar asterisk setup that is experanceing the same problems.


PEER DETAILS

allow=ulaw&alaw
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
fromdomain=as.wxcnz.net
fromuser=37xxxxxx
host=as.wxcnz.net
insecure=invite,port
nat=yes
secret=............
type=peer
username=............

Cheers,
  Craig

 
 
 
 


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  Reply # 101271 21-Dec-2007 16:58
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What type of phone are you using and what is the DTMF setting set to in that? What do you have the dtmfmode= setting set to in your extension configuration?

The only DTMF I've ever had was getting it to work correctly with Asterisk using a Budgetone phone however that was related to the poor quality of the phone!



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Wannabe Geek


Reply # 101272 21-Dec-2007 17:15
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Checked all asterisk config files and phones all dtmfmode statements set are rfc2833.

All Snom 320 or Linksys SPA922, should be good phones...  My friend was using X-Lite soft-phone or similar

Strange thing is that it seems to work fine with the voice portal, just not other any outside service.


Have always had the problem since installing VFX.  But did try setting the VFX dtmfmode to "rfc2833,inband" which sort of worked, but seemed to be flaky, and I think that the VFX voice portal didn't work with it, so swapped it back to rfc2833 and contacted the help desk.

Cheers,
  Craig

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WorldxChange

  Reply # 101277 21-Dec-2007 18:21
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DTMF through the network is rfc2833 and is pretty standard across all devices, what you need to do is look at the invite for this line

m=audio 6068 RTP/AVP 0 8 18 101 ----This tells us you are sending rfc2833, 

most likely this may be missing first guess without seeing a trace , as the inbound calls work fine you say and I know this
paramater is passed so this would be why, if you send me the details of a bad call I will look for you and see if we
can track it down




Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well Wink

             

https://www.facebook.com/wxccommunications

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  Reply # 101281 21-Dec-2007 18:52
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Argh my fine Asterisk friend, it was as I thought, you are missing the rfc2833 from your invite, I logged on and found your fault ticket so I now have your number


This is one of your invites and is the same for all invites you send, btw the voice portal works a little differently as it supports different types so thats why it works even though your not sending rfc2833 Wink
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 21 Dec 2007 04:01:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 3526 3526 IN IP4 192.168.0.252
s=session
c=IN IP4 203.167.xxx.xxx
t=0 0
m=audio 16246 RTP/AVP 0 8 96 <---NOTE THE MISSING 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off -


this is a config issue on your box I'm afraid






Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well Wink

             

https://www.facebook.com/wxccommunications



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Wannabe Geek


  Reply # 101283 21-Dec-2007 19:00
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OK, I'll have a closer look at the system and config files, and try a few things, must be there somewhere...

Won't get a chance to look at it now until the morning, but shouldn't take too long now I know what I'm looking for, I'm starting to get used to all the Asterisk config file lingo. 

The red herring for me was that it is working with the voice portal, made me think it was something further up the system, but that makes sense now.

Thanks for the help, I'll let you know what I find...

Cheers,
  Craig

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  Reply # 101284 21-Dec-2007 19:03
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No problem, glad to be of service ...I must be in a festive mood though as we are going to leave Asterisk support to the Asterisk community Wink




Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well Wink

             

https://www.facebook.com/wxccommunications



5 posts

Wannabe Geek


Reply # 101922 27-Dec-2007 18:07
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Many thanks to Maverick and sbiddle for help solving the DTMF issues, and as maverick suggested it was a problem with Asterisk at my end.

It seems that a bug has been introduced into Asterisk 1.2.24, which sets the payload type to 96 instead of 101.  One way of getting around this bug is to roll the Asterisk install back to an earlier version.

(rollback info courtesy of this trixbox forum article on NVFax issues)
http://www.trixbox.org/forums/trixbox-forums/open-discussion/what-versions-what-will-give-you-working-fax-it-sits-right-now

To  go back to an earlier release of Asterisk, try the following.

Files required for rollback of Asterisk.

 asterisk-1.2.23-1.i386.rpm
 asterisk-addons-1.2.7_1.2.23-4.i386.rpm

1) Find these files on your box.  They should be already in /var/trixbox_load/rpms/
2) cd /var/trixbox_load/rpms/
3) yum remove asterisk
4) rpm -ivh asterisk-1.2.23-1.i386.rpm
5) rpm -ivh asterisk-addons-1.2.7_1.2.23-4.i386.rpm

Restart or Reboot Asterisk.  All done.

Cheers
  Craig


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