we've been getting complaints that DTMF is not working, i checked full log
for a call that they claimed DTMF didnt work, I noticed this:
DTMF begin '7' received
DTMF begin ignored
DTMF end '7' received
DTMF end passthrough '7' why is the "DTMF begin ignored" called?
Reposting this since since i was not a member of the list when i first sent
it I'm trying to configure an asterisk serrver with a NTT T1 line, i get the
error 'no D-Channel found', i searched the net and seems that NTT Japan uses
a special switch type called NTT (
http://en.wikipedia.org/wiki/Primary_Rate_Interface) there was sa patch in
only, use google translate) but is for an older version of
asterisk. Do you have an idea on how this can be fixed? asterisk-220.127.116.11-0
I am unable to call *2*999... because my phone automatically sends the
number after I press *. So my IP-phone calls *2. Now this is a Cisco, but that's not my question. Does anyone know what
setting I need to adjust so my phone (but actually any IP-phone) accepts
an * in the middle of a number ? At the moment, I don't really know what I'm looking for. So if anyone
knows how to do it in a Cisco, Grandstream, Yealink or Snom IP-phone I
can find out myself what settings to…
When using an extension to my android gingerbread nexus one,
calls drop after a n minutes of call due as per the following
[Jun 21 09:34:37] == Begin MixMonitor Recording SIP/nexusone-0a39
[Jun 21 09:34:37] -- Executing [0031xxxxxxxx0@default:4] Dial(" ...
[Jun 21 09:34:37] -- Called 45xx:firstname.lastname@example.org/00x ...
[Jun 21 09:34:37] -- Call accepted by 18.104.22.168 (format al ...
[Jun 21 09:34:37] -- Format for call is alaw
[Jun 21 09:34:38] -- IAX2/4506-8090 is making progress passing
[Jun 21 09:34:42] -- IAX2/4506-8090 answered SIP/nexusone-0a39a .c: Context 'macro-notifymobile' for macro 'notifymobile' lacks 's'
Thank you Alec,
I needed some confirmation that it wasn't something I was doing. I can
live without pickupsound, and the bug is already reported - so it's all
good. Sebastian On 21/06/11 00:29, Alec Davis wrote:
> This has been fixed only last month, see
> https://issues.asterisk.org/view.php?id=18654 and try bug18654.diff.txt
> That will avoid the deadlock, but it's not the proper fix, there are other
> issues that could trip you up,
> mainly to do with race conditions with multiple channels picking up the same
> ringing extensions.
Dear Asterisk Users,
I have a Sipura 2000 device, and since last few days I have been searching
for its latest firmware for upgrade. Googling tells me that Cisco has
stopped the support for this device and I dont have definite idea on where
would I be able to find the firmware to upgrade my device.
Any help in regards to getting the firmware will be helpful.
I still have the same problem trying to configure ITSP failover in
extensions.conf for a connected PRI. Any comments thoughts or direction
would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with
multiple DIDs we will spread them across two or three ITSPs so that all
inbound connectivity will not be lost if one of them has an issue. I have a little experience with using SS7 from when we set up multiple call
centers in Norway for Telenor. Using SS7 we were able to determine incoming