Displaying 20 results from an estimated 4000 matches similar to: "DAHDI: How to know since when it is used? How to shutdown after max time?"
2013 Feb 03
2
RTP timeout if the asterisk box behind NAT
Dears;
I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides).
My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the
2009 May 21
2
MeetMe not working with GSM codec?
Hi,
I am not sure if I am doing something wrong, but I can't get MeetMe to
work with GSM codec (Asterisk 1.6.1 SVN r190371).
My config files below:
---- sip.conf: ----
[general]
context=common
canreinvite=no
bindport=5060
bindaddr=78.105.1.127
disallow=all
allow=alaw
allow=gsm
rtptimeout=600
rtpholdtimeout=3600
rtpkeepalive=30
nat=no
jbenable=yes
tcpenable=no
realm=dev-sip.wima.co.uk
2007 May 01
10
Digital Phones
Hi List;
Asterisk does not have any kind of cards that can work
with it to be used with Digital Phones (digital phones
differ than analoge phone and differ than IP Phones).
Anyone can advise about this as I did not find this on
Diguim
Regards
Bilal Ghayad
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2008 May 21
1
speex, ilbc and g729 codecs, GSM with IAX
Dears;
I do not know if any had experience in using speex or
ilbc with IAX and got good results, because I am
facing a problem with GSM.
I am facing a noise problem when I am using GSM with
IAX trunk as following:
IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk
using GSM codec ---> Remote Asterisk Box ---> Digium
Card (FXO) to terminate the call to the destination
While no
2008 Apr 08
3
RTCP not being sent when on hold
Hello,
When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I
place the call on hold, the call is dropped after 30 seconds.
It looks like there is no RTCP/RTP sent to the client from Asterisk while on
hold (music on hold playing to caller) thus client disconnects the call.
During this time, I get the following messages in the CLI:
NOTICE[24194] rtp.c: Unknown RTP codec 126
2012 Sep 11
2
asterisk boxes looses registration
I have a couple asterisk boxes, running sip between both boxes. 1.4.43
on both.
both are installed from source,
both have default settings,
My config for one box is:
[devgeis]
type=friend
defaultname=devgeis
username=devgeis
secret=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
rtptimeout=60
rtpholdtimeout=60
rtpkeepalive=60
host=192.168.1.8
context=panel
The other box is the same.
There
2007 Jan 18
2
Asterisk not hanging up
I have a problem with calls not hanging up if for some reason the
physical phone dies or gets unplugged
I can demonstrate this in practice by making a call from a handset, then
unplugging the handset from the power. The call remains active and
asterisk never seems to disconnect it.
More annoyingly when power is re-applied the handset comes back to life,
won't receive incoming calls
2020 Aug 06
1
asterisk 13.33 and polycom
I am using asterisk 13.33.0 and POlycom phone with the latest firmware.
The polycom phone is behind a firewall, the server is in the cloud.
If the polycom has just booted - it receives a call, after some time
(couple minutes) it no longer receives a ring. I see no errors in the CLI -
looks just like the previous call as far as I can tell.
Then reboot the phone and as soon as its ready call it
2011 Mar 29
1
E1 PRI configuration: DAHDI and LIBPRI
Hi All;
I have an E1 card with two ports for ISDN PRI.
Do I need to install DAHDI in addition to LIBPRI?
For placing outside calls (outgoing) via the PRI, then in the extension.exe file, I will use the Dial function? But how can I determine that I need to use the PRI channels and not the analoge channels?
Last point: how can I know that asterisk is containing libpri? In other words, how can I
2011 Jun 14
3
sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway!
Dear;
Thanks a lot for guiding me.
Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch?
Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5):
libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch
It gave me that patched detected as shown below (example of one file, and I got same for other files):
patching file
2011 Jun 14
1
sig_pri.c:985 pri_find_dchan: Span 1: D-chanannel anyway!
Dears;
To patch libpri: I just place the patch file in the libpri source directory and then I run make and make install?
Or I need to compile the dahdi and asterisk also?
If the problem stayed, do I have to go for previous libpri version? Or for previous dahdi version and asterisk version?
Regards
Bilal
-----------
> bilal ghayyad wrote:
> > But I am afraid it is a bug because I
2007 Sep 25
5
Do I need to run #modprobe zaptel for Digium
Hi List;
If I am configuring Diguim Analoge card, then I need
to run #modprobe wctdm, but the question why I need to
run #modprobe zaptel also?
What #modprobe zaptel does a things that #modprobe
wctdm does not do?
Any help?
Regards
Bilal
____________________________________________________________________________________
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2008 Oct 27
11
Fring: Open VPN client to be installed on the mobile, which mobile?
Hi All;
I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile:
Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server?
Regards
Bilal
2009 Jan 25
2
Zaptel transfer using any button or code, but not flash hook
Hi List;
I need to do a call transfer using analoge phone connected to fxs, but I do not need this to be done using flash hook, let it to be using the # or * or any code, but how I can configure that this code is for transfer? Also, I do not need the flast hook to be used for trasfer as it cause usually a confusion to distinguish between the hangup and the call transfer.
Any advise?
Regards
2009 Jul 05
1
SIP IP-Trunk to be authenticated based on username and password, not IP address
Hi List;
How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed
2007 Jul 24
10
What is the best softphone work with Asterisk
Hi List;
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
iax2?
Regards
Bilal
____________________________________________________________________________________
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2013 Mar 09
7
Sending SMS from asterisk
Hi;
If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How?
Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW?
2008 Dec 21
6
Asterisk and Dabatase
Hi All;
Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)?
Any advise?
Regards
Bilal
2011 Mar 05
3
Prepaid Billing other than A2Billing
Hi All;
Any one advise for open source prepaid billing other than A2Billing that can work with Asterisk and it is rich by features (for large business)?
Regards
Bilal
2009 Jun 18
3
asterisk-gui: read/write in the conf files or db
Hi Danny;
Really I did not understand how I can determine if the IO will be DB or conf files? Is it from the Asterisk manager?
Regards
Bilal
-------------------------
It depends on how you are configured. The gui interfaces using Asterisk
Manager, so you get the Same IO from the gui that you would get from a
native manager session.
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