Easy, use Read() to capture the incoming DTMF from Server-B
Server-A Server-B
Initiate-Call - AnswerCall()
SendDTMF(5)-- Read()
Read()-SendDTMF(4)
SendDTMF(3)-- Read()
2011/12/29, Mindaugas Jasiulis mindaugas.jasiu...@mediafon.lt:
AEL2 ifTime use it too. AEL2 ifTime become the same GoToIfTime in the
dialplan :)
I feel a bit shameful but I must say that it's now working on my system !
I don't understand why it didn't yesterday.
Anyway, thank you very much for
In server B if I use SendDTMF then it means I am changing programming at
server B. Actually I don't have right or permission to change programming
in server B.
otherwise your suggestion is best for channel base communication.
On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind govoi...@gmail.com
Jeroen Eeuwes писал 29.12.2011 07:29:
Probably my
understanding is limited, but it seems to me that they
have already
'access' to your Asterisk for them to be able to try to
make outgoing
calls. Wouldn't it be better to make sure they get the
usual errors
like Registration from failed -
o in that case you need to observer the call flow in Server-B, i.e what is
the length of sound file playing. what DTMF it requires etc etc and once
you detect the call flow for a successful IVR traversal then mimic the
behaviour of the call from Server-A.
Thats all you can do.
Think of it exactly
Hi all,
I am using to Xlite to save video voice mail.
when i retreive it, then only video show , no voice is here out.
Plz tell me where ,i am wrong , and how i can able to see video plus here audio
in voice mail box.
I did following configuration
In Sip.conf
videosupport=yes
On 12/28/2011 03:57 PM, Jerry Geis wrote:
I ran into a rare situation today.
A really short message is being played over the ALSA or console
channel from one asterisk box to another. Both running 1.4.30.
the incoming context on the ALSA or Console port box first runs an AGI
before connecting
On Thu, Dec 29, 2011 at 12:05 AM, Bruce B bruceb...@gmail.com wrote:
I have been running 1.8.7 with a few fixes back ported from the 1.8.8
release candidate for the last 2.5 months. The system processes around
4,000 calls per day over PRIs for 250 Polycom phones.
Previously I was running
Hi,
I Have added this line for asterisk 1.8 (i have allowguest=yes and
context=default in sip.conf):
NOTICE.* .*: Call from '.*' (HOST) to extension '.*' rejected because
extension not found in context 'default'.
Em 29-12-2011 13:03, Patrick Lists escreveu:
Hi,
In the thread Interesting
Hello
Asterisk only says that the iax2 channel don't work maybe you look the
iax.conf. you trunk. Is iax I think
Regards
On Dec 29, 2011 6:49 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Hello all,
I attempted to make a couple of outbound calls this morning and always got
the busy tone. I
Log are being filled with g729 transcoding error in 1.8.7x now :-(
I don't dare to test 1.8.8x as it might have something else broken.
Unfortunately I can no longer trust the release candidates. Thanks for the
input.
On Thu, Dec 29, 2011 at 8:29 AM, Ryan Wagoner rswago...@gmail.com wrote:
On
I have a couple of performance/memory related questions:
Is there any downside to using long URIs as far as memory or database (mysql)
performance is concerned, e.g.
sip:1234567890_1234567...@abc.com? Or is this negligible?
Also is there a performance hit if no pattern matching is used?
We are currently using an older version of Eyebeam on our deployment and
keep having an issue with the disappearance of SIP accounts, and after
research found it is a bug on the version we currently have.
I am looking for a new softphone solution and I was wondering what
everyone was using
Hi,
In the thread Interesting attack tonight fail2ban them Bruce B
mentioned it would be nice to have input from the Community to come up
with the best set of fail2ban filters. That's a great idea. So let's
start with Bruce's filters (thanks!) and take it from there. Anyone have
any
1. I checked the log and I don't see any registration attempt, so I *assume*
they simply send an invite, and so they are in the external/outside context of
my dialplan. So they are trying to reach extensions which don't exist. If
they succesfully registered they would be on the internal
On Wed, Dec 28, 2011 at 4:10 PM, Danny Nicholas da...@debsinc.com wrote:
Can somebody point me to an explanation from Kevin or Tzafir or someone else
up the food chain explaining the differences/benefits of 1.6/1.8 vs
1.4/10.0?
What's the difference between a car released in 2006 versus a car
Maybe your logger is not setup properly?! You should get the IP in logs. I
can't think of when you won't get the IP in your logs unless the SIP
packets are manipulated. That IP is from Voxel.net. You don't have a VPS or
service from them do you?
2011/12/29 Michelle Dupuis mdup...@ocg.ca
1. I
Hello all,
I attempted to make a couple of outbound calls this morning and always got the
busy tone. I checked the Asterisk console and was greeted with:
[Dec 29 11:29:22] WARNING[12039]: app_dial.c:2218 dial_exec_full: Unable to
create channel of type 'IAX2' (cause 20 - Unknown)
==
Hi,
I Have added this line for asterisk 1.8 (i have allowguest=yes and
context=default in sip.conf):
NOTICE.* .*: Call from '.*' (HOST) to extension '.*' rejected because
extension not found in context 'default'.
Em 29-12-2011 13:03, Patrick Lists escreveu:
Hi,
In the thread
Hello,
I'm trying to interface Asterisk with a third party voicemail system. This
voicemail system registers itself as extension 199. This voicemail system
gets the DID number (mailbox) from the SIP To: header field.
My problem is creating the SIP INVITE with a To: field that's different
Trying to set up a simple sip - sip connection over tcp. Home : 10.0.0 -
Office: 1.8.8.0
Home sip.conf:
register =
tcp://office-going-to-home:password@office-ipaddr/home-coming-from-office
[home-coming-from-office] ; receives calls
type=friend
transport=tcp
dtmfmode=rfc2833
On Thu, Dec 29, 2011 at 12:18 PM, Bruce B bruceb...@gmail.com wrote:
Log are being filled with g729 transcoding error in 1.8.7x now :-(
I don't dare to test 1.8.8x as it might have something else broken.
Unfortunately I can no longer trust the release candidates. Thanks for the
input.
What
Dears,
1-I have a GSM gateway (GOIP) with 8 ports, I used to let every port
register to VoIPSwitch in order to know how many minutes does this GSM
card, ASR ,ACD on each card.
It's too simple on VoIPSwitch to add the registrar client to dial plan ,but
in asterisk only I can find trunks
How
Greetings,
Is there a way to block a specific inbound number? I've found code online for
blocking all nocallerid and all 800, etc. but nothing for a specific number. My
company is wanting me to block a specific number. Is this possible in Asterisk
1.4 and 1.6 or do I need to go through my
Take a look at Blacklist
I love that command and love to send nice intercept messages to the other
side J
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Thursday, December 29, 2011 8:40 PM
To:
Check out the X Boy/Girl friend feature.
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf
Around the middle of the page.
Stu
-Original Message-
From: Kevin Oravits korav...@rcolegal.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
Hi There,
I'm fairly new to asterisk and I'm trying to play a video file during a
video call without success for a couple of days now.
I've posted a question at stack-overflow describing my problem -
http://stackoverflow.com/questions/8675713/asterisk-video-playback-mp4-3gp
*
In short, what
Hi Michelle,
1. I checked the log and I don't see any registration attempt, so I *assume*
they simply send an invite, and so they are in the external/outside context
of my dialplan. So they are trying to reach extensions which don't exist.
If they succesfully registered they would be on the
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