Good afternoon,
I'm trying to configure my asterisk to work with DTMF signaling (I
live in Brazil) , i've put these lines in my chan_dahdi.conf
usecallerid=yes
callerid=asreceived
cidsignaling=dtmf
cidstart=polarity
my dahdi system.conf
loadzone=br
defaultzone=br
last time i had this issue with teliax, they recommended to upgrade to 1.4
On Fri, Aug 29, 2008 at 3:44 AM, Chris Mason <[EMAIL PROTECTED]> wrote:
> I tried DTMFmode=auto and it did not help. Any further ideas?
>
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I tried DTMFmode=auto and it did not help. Any further ideas?
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As
Have you set dtmf mode rfc2833 or avt in your phone?
On Thu, Aug 28, 2008 at 4:29 PM, Chris Mason (Lists) <[EMAIL PROTECTED]>wrote:
> I have a client with 30 extensions, all Polycom 501 phones, an Asterisk
> 1.2.30.2 installation, and trunking over SIP to TelIAX. Everything works
> fine except wh
I have a client with 30 extensions, all Polycom 501 phones, an Asterisk
1.2.30.2 installation, and trunking over SIP to TelIAX. Everything works
fine except where they need to use DTMF to navigate IVRs such as
Dell.com. The tones are not recognized at all.
My sip.conf lists for each extension:
OK, looks like I made a mistake in the rev number -- its 67457 not
6745 from early Sunday.
The bug was fixed a while ago, but has reared its ugly head again.
on Sunday 07/01/2007 Russell Bryant([EMAIL PROTECTED]) wrote
> John covici wrote:
> > OK, using zaptel 1.4 and asterisk 1.4 rev 6745, i
John covici wrote:
> OK, using zaptel 1.4 and asterisk 1.4 rev 6745, if someone types an
> asterisk from the other end of a call, I here it forever till the call
> hangs up. Looks like it starts the vldtmf, but never ends it from the
> logs.
>
> I am using Digium 400P rev I with one fxs and one f
OK, using zaptel 1.4 and asterisk 1.4 rev 6745, if someone types an
asterisk from the other end of a call, I here it forever till the call
hangs up. Looks like it starts the vldtmf, but never ends it from the
logs.
I am using Digium 400P rev I with one fxs and one fxo module.
Any way around thi
Hello:
I have a problem when a person call a queue and the agent answer the call.
If the caller press any key during the dialog the agent hear a continue
sound like dtmf.
Any ideas?
Thanks.
Patricio.
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In my expirience the Broadvoice has a lot of "audio issues". A number of times DTMF did not work. You pretty much get what you pay for.Will Glass-Husain <[EMAIL PROTECTED]> wrote: Hi,I'm struggling a bit with DTMF. It seems to work fine on my internal network, but when I call outside lines with
Brian Capouch wrote:
Martin Joseph wrote:
On Mar 22, 2006, at 2:49 PM, Avi Miller wrote:
Will Glass-Husain wrote:
My local phone is a Grandstream GXP-200
dtmfmode=info
For the GXP2000's, you want to change this to:
dtmfmode=rfc2833
They don't really handle INFO mode well, in my exper
he cellphone is in a poor reception area, it might not
send DTMF tones correctly.
-- Bjorn
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Will
Glass-HusainSent: Wednesday, March 22, 2006 5:47 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] problems
with DTMF
Hi,
On Mar 22, 2006, at 2:49 PM, Avi Miller wrote:
Will Glass-Husain wrote:
My local phone is a Grandstream GXP-200
dtmfmode=info
For the GXP2000's, you want to change this to:
dtmfmode=rfc2833
They don't really handle INFO mode well, in my experience.
That's what I was thinking also. In a
Will Glass-Husain wrote:
My local phone is a Grandstream GXP-200
dtmfmode=info
For the GXP2000's, you want to change this to:
dtmfmode=rfc2833
They don't really handle INFO mode well, in my experience.
--
National Manager - Special Projects
< Sydney / Melbourne / Canberra / Hobart / London
Hi,I'm struggling a bit with DTMF. It seems to work fine on my internal network, but when I call outside lines with telephone trees, some systems understand the DTMF and some ignore it. Anyone have tips on solving this? Thanks in advance.
My local phone is a Grandstream GXP-200mailbox=89username
hi doug,
I do have rfc2833 and i also tried with inband ... the actual scenario is like this...
I have my polycomms working for POTS which i connect using through Cisco As5300 but when i try to send my call to a VOIP provider using a T1 line through Cisco 3620.. i have problems with DTMF.. My
At 15:36 11/7/2005, Krishna Sumanth Chava wrote:
hi,
Would like to have help in fixing the DTMF problem i am facing on Polycomm
Soundpoint IP Phones
I am having the following network setup..
I have my Asterisk PBX server connected to the Cisco 3620 Router with an
ethernet cable which intur
hi,
Would like to have help in fixing the DTMF problem i am facing on Polycomm Soundpoint IP Phones
I am having the following network setup..
I have my Asterisk PBX server connected to the Cisco 3620 Router with an ethernet cable which inturn is connected with a T1 circuit to my SIP Provider
I've got a problem with DTMF, again.
My asterisk box is connected with the outside world (PSTN) via a sip
proxy. The problem is that for some reason, I need to use rfc2833 for
signaling digits to the gateway and inband to accept digits from outside
(eg. when someone dials one of our DIDs). It's pos
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