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captainacronym Newbie
Joined: 10 Nov 2005 Posts: 5 Location: Buffalo, NY
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Posted: Thu Nov 10, 2005 6:01 pm Post subject: Navigating inside Touch Tone PTSN Phone systems |
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Hi!
We started out using Gizmo for our office, but recently we decided that actual phones are still better in a multi-tasking work environment where you need to use both phone and computer at the same time.
So, we've installed 3 Call-In-One adapters, each with a Uniden TRU 8865 hooked up to it, on a 4.5Mb down / .85Mb up cable connection.
We LOVE it! [except for this]
When we press keys to try to navigate through a phone system [i.e. voicemail, banking, or virtually anything else powered by some sort of automated system], the system almost always hears multiple key-presses for each press on the phone. Obviously, this renders the phones useless on those circumstances.
Is there some setting we need to adjust in order to make this work? Or is there some other factor we should know about?
Thanks in advance for your help!
C. _________________ -------------------------------
Craig C. Chapman
1-747-612-2464
Last edited by captainacronym on Sun Nov 13, 2005 12:42 am; edited 1 time in total |
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captainacronym Newbie
Joined: 10 Nov 2005 Posts: 5 Location: Buffalo, NY
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Posted: Sun Nov 13, 2005 12:41 am Post subject: Fixed it! |
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Hi folks,
Well the guys at support didn't exacly fix my issue, but they did point me in the right direction.
It all resides in the OOB Settings. Now contrary to what is posted on the support site here, you actually want it set this way:
• Send Out-Of-Band: SIP INFO
• Payload Value: 101
• Check BOTH: "Supress voice packets..." AND "Squelch inband DMTF audio" options
• ABCD event signalling mode: Transition
Don't ask me exactly why it works...but it seems to limit the length of the tone, so that the effect is that it only generates a single short tone vs. multiple short tones. It also doesn't matter how long you hold the keys - you'll still get a single short tone.
I've configured all our Call-In-Ones this way, and successfully entered a 9-digit conference code in our conferencing service nearly every time - in fact out of 6 tries, it only failed once.
Likewise with our VM system, which only requires a 4-digit extension number and 4-digit passcode.
Then I successfully tested a 16-digit bank card number and 5-digit security code.
In all, this seems to have done the trick, and now we have a fully-scaleable, state-of-the-art office phone system, all tied together with our PBX.
Happy SIP-ing! _________________ -------------------------------
Craig C. Chapman
1-747-612-2464 |
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k0fcc Enthusiast

Joined: 03 Jun 2004 Posts: 26
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Posted: Wed Apr 19, 2006 7:18 pm Post subject: |
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I had this problem with CIO calling conferencing services. Your settings didn't work for me so I played around and this is what I was able to get t work:
OOB - RFC2833
Payload - 100
Supress - yes
Squelch - no
mode - transition |
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JustaSIP Forum Freak
Joined: 25 Aug 2005 Posts: 272
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Posted: Wed Apr 19, 2006 7:56 pm Post subject: |
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| Thanks for sharing, k0fcc. Sometimes it takes trial-and-error to find the best combination for a given system |
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captainacronym Newbie
Joined: 10 Nov 2005 Posts: 5 Location: Buffalo, NY
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Posted: Wed Apr 19, 2006 11:58 pm Post subject: WOW! |
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Hey! That works even better! Thanks! _________________ -------------------------------
Craig C. Chapman
1-747-612-2464 |
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k0fcc Enthusiast

Joined: 03 Jun 2004 Posts: 26
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Posted: Thu Apr 20, 2006 5:48 am Post subject: |
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My pleasure. I'm the same 'k0fcc' that wrote the FWD FAQs so I have a bit of experience in playing with settings.
My theory as to the payload of 100 is actually based on something that happened with Nortel phones last fall. Using 101 caused the Nortel system to think that it was getting more than 1 tone at time. More and more newer systems are causing the full payload to get thru instead of some...what I'll call...attentuation. This is relatively very much prevalent on full IP systems where no line loss is incurred as compared to a traditional phone system.
Anyway, something for the SIPphone FAQs.
Joey |
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