Displaying 20 results from an estimated 11000 matches similar to: "Polycom DTMF"
2005 Jul 09
3
polycom soundpoint 300 sip phone and hold music
I have an extension setup in my extensions.conf for hold music. ext. 600.
If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600
I hear the hold music playing. If I call another extension and pick it
up and put the call on hold with the hold button on the phone I hear
nothing at all. Does anyone have any experience with these phones and
getting the hold button to work?
2004 Sep 03
2
Wall-mounting UIP 200 and SoundPoint IP600 keeps coming off hook
I am looking for a large number (probably about 100 or so) low-cost
phones that I can hang on the wall. I need the phones to use PoE. Do
the Uniden phones support wall-mounting? These phones are not going to
be high-usage; they simply need to be there in case of an emergency.
Another question, along the same kind of lines, has anyone figured out
how to keep the SoundPoint IP 600 receiver
2005 Feb 09
5
polycom soundpoint ip 300
hello,
I try to set up two lines per ip 300 phone,
registration is ok but i get Failure to authenticate
407 for subscribe.
Anybody could help me to configure Asterisk in order
to set instant message and presence ?
I've tried with Ondo sip server it's ok !
Regards
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2005 Mar 23
2
*-1.0.7 DTFM => Not working
My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it
works in version 1.0.5 (was working with 1.0.3).
I'm using SPA-3000 and dtmfmode=inband
--
#Joseph
2007 Jun 22
10
inband DTMF for g729
Does anybody know why Asterisk does not support inband DTMF for G.729?
Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system.
Any suggestion to solve this problem?
Gary
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2012 Sep 28
1
ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
Hi list!
ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF
gets transmitted throughout the conference. I've tried Asterisk 10.7.1
from the official RPMs and 10.8.0 compiled from source.
I've confirmed that it's disabled via the CLI "confbridge show profile
user <profilename>".
It's an all-SIP scenario with RFC2833 as the DTMF protocol.
2005 Mar 18
1
XML config files for Polycom SoundPoint IP 300?
I bought a couple Polycom Soundpoint 300's, and have them working nicely
with SIP... but I'd like to be able to do automatic config via FTP, but it
requires some XML config files. The docs discuss them in detail, but I
can't seem to d/l them from Polycom. [No, it doesn't appear to be on the
CD that comes with the phone.] I've created an account at Polycom's tech
support
2004 Apr 23
6
Polycom registration
I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone
registered on an asterisk box but am having no luck. I get the
following errors 192.168.22.196 being the phone and 22.254 being the
asterisk box..
Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request:
Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for
2004 Sep 01
5
dtmf problem
Hello!
I have asterisk updated from CVS on 31/8/2004 with
sample configuration. I have just changed the
sip.conf to register asterisk with sip proxy in out
intranet.
Then I can successfully make call to asterisk and go
to demo IVR, but no response to dtmfs.
I try to make call from several sip phones: Cisco7960,
Ata186, Snom200. All of them send telephone-event in
INVITE, but asterisk answers
2005 Jul 20
2
Last two digits getting cut off?
We've just setup our A@H server, with our quad port card. Everything works
well so far.
One thing I notice is that when I leave the handset on the hook and dial a
#, all is well. If I pick up the phone and dial, it cuts off at 10 digits,
which is a problem if I need to dial 1+area+phone # (12 digits).
The phones are Poly Soundpoint IP 600's. I'm wondering if I've missed a
2004 Jul 12
3
dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames
Hi can anyone help me on this error msg??
dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames
thnx
St
2003 Mar 09
6
DTMF detection on SIP provider ?
Hi..
I just wondering why DTMF are not recognized by aterisk on incoming calls
from my SIP provider ...
ANy suggesteions ?`
/Mike
2007 May 03
1
Double DTMF digits
When dtmfmode is set to inband for SIP, and i originate a call from sip
out to the PSTN, I can hear the DTMF digit twice in the audio stream.
Once very briefly and once for normal duration.
Our Theory: While Asterisk is parsing the DTMF, for a fraction of a
second, while the end user generated DTMF is being detected, the DTMF is
passed inband. Once the DTMF is detected Asterisk silences it
2007 Jul 25
5
IAX2 INBAND DTMF?
Is it possible to make Asterisk do inband DTMF over IAX?
2009 Sep 25
3
disable dtmf on SIP peer
Hello,
I have one problem and I need to disable dtmf (disable rfc2833, info and
inband) on one (other peers must support dtmf) SIP peer . Is it possible?
Workaround would be use g729 codec with dtmfmode=inband.
Maybe there is better solution?
Thanks for help.
--
Pagarbiai / Best Regards,
Giedrius Augys
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2009 Apr 09
2
DTMF
[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted:
Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but
URGENT[image:
Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>
2005 Jun 01
3
DTMF not working
Im trying to configure voicemail, but asterisk doesnt respond to dtmf codes.
I uses kphone with g711u codec (I've tryed the others one) and in sip.conf I
configure dtmfmode=rfc2833 (I've tryied inband and info).
Asterisk seems not to "see" the tones. Could somebody help me? Thanks
2004 Jun 11
11
Broadvoice and DTMF
I understand there has been some issues sending DTMF tone through
Broadvoice. Can some provide me with symptoms?
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2018 May 01
2
DTMF tones in MixMonitor recording
Hello list,
Hope you are all doing fine!
I have stumbled over some piece of dialplan code in which apparently they
were trying to avoid recording the DTMF tones in the wav file. It is really
messy and I am not sure if this really works. So after a bit of research I
found this comment (
https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is
said:
*"Asterisk strips the
2005 Jun 06
5
Polycom 500...
I am having a strange problem with a couple of Polycom IP 500 phones. I
know this is not related to Asterisk, but maybe someone here had the same problem.
I configured my phones following the documentation at voip-info.org and
they are working very well. The only problem I have is that when I dial an
extension like 1100 the phone changes that to 0110 and obviously the call
fails. I have