On 04/18/2011 06:36 PM, Paul Belanger wrote:
On 11-04-18 09:46 AM, Jonas Kellens wrote:
Asterisk freezed and only a reboot of the whole server fixed this. Any
command on the Asterisk CLI was not executed because Asterisk was too
busy processing all of these messages that you see in the debug
Hello List,
I have seen from the following link that, for SIP channels there is no audio
communication possible in MeetMe with AGI_BACKGROUND.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
Currently we are using asterisk-1.6.2 and the problem still persists. Is there
any solution
Hello List,
Is it possible to run an AGI script in backgroung for all the associated SIP
channels in ConfBridge Application? If yes how?
This can be done using 'b' parameter in MeetMe for non SIP channels.
Regards,
Rajib
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto:
Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and
internet is offline.
srvlookup = no didn't help.
What about putting my provider's name in /etc/hosts?
Should it solve the problem?
A caching nameserver is not a
In article
2658e54b540d284981ea57e6a549ea70abd1f7c...@inblrk77m1msx.in002.siemens.net,
Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote:
I have seen from the following link that, for SIP channels there is no audio
communication
possible in MeetMe with AGI_BACKGROUND.
On Tuesday 19 Apr 2011, Niccolò Belli wrote:
A caching nameserver is not a viable solution because I want it working
even after a month without internet access.
Then just make your local nameserver authoritative for the domain in question.
You can always firewall off port 53, if the
I have enabled DNS manager in /etc/asterisk/dnsmgr.conf. It helps me.
On 19.04.2011 14:05, Niccolò Belli wrote:
Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto:
Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and
internet is offline.
srvlookup = no didn't help.
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables
the
Hi Rajib,
this is your second post on Meetme with SIP channel and AGI script, Can you
provide your requirement to run an AGI for Meetme , what you want to run an
AGI with meetme.
in confbridge there is nothing option for running AGI in background mode.
let us know what you want to do exactly, on
@digium
1. What happened with the 1.4 patches that still wait on
issues.asterisk.org? e.g. issue #19108
2. What happened with bugfix patches for 1.4 made by the community.
Will those be ignored now?
(e.g. i have one more a memleak fix for 1.4 in preparation, that i can
publish earliest after
2011/4/19 Niccolò Belli darkbas...@gmail.com:
Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto:
Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and
internet is offline.
srvlookup = no didn't help.
What about putting my provider's name in /etc/hosts?
Should it
Make sure ALL IP addresses of the system are in /etc/hosts, as well as the IP
of your provider. Asterisk gets upset if it can't do a reverse lookup of an IP
address on the system.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi all,
I downloaded a copy of bristuff 0.4.0-RC11 on my Ubuntu 10.04.2 LTS server
machine with 2.6.32-24-generic-pae kernel.
The prereq.sh script executes without complaints (BTW on my system,
libncurses-dev evaluates to libncurses5-dev and libz-dev evaluates to
zlib1g-dev).
With the asterisk
On 11-04-19 09:28 AM, Kristijan Vrban wrote:
@digium
1. What happened with the 1.4 patches that still wait on
issues.asterisk.org? e.g. issue #19108
Once a branch moves into security mode; no more bug fixes will be
applied. If a security issue affects the 1.4 branch, a new release will
be
Can someone confirm if the bug present in #18951 has been fixed in 1.6 or 1.8 ?
If not, then I am stuck on my current version of 1.4, and will not be
able to upgrade to either of those two versions, even for security
fixes.
Julian
On 19 April 2011 15:52, Paul Belanger pabelan...@digium.com
Its not really had to install 1.6 or 1.8 on a test box, and see if a phone
connects to it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith
Sent: Tuesday, April 19, 2011 11:02 AM
To:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I have no audio on chan_mobile but this message repeats continuously:
Dropping incompatible voice frame on Mobile/DNA-54f4 of format slin
since our native format has changed to 0x0 (nothing)
Can somebody point me to the right direction?
Hi
I have a weird issue with a new 1.6.2.17.2 box.
At random intervals it just stops responding to RTP and signalling
(both SIP and IAX observed). All calls in progress lose audio both
ways although the console shows the call legs still in progress. No
signalling can be sent or is received. It
The point I was trying to make was that *anyone* on 1.4 who uses
ciscos will be forced to move to 1.6 or 1.8 if they want any security
fixes applying , as the patches will go into 1.4 svn where the bug is
present.
IOW if you uses cisco's and 1.4 then that's the end of the line for
you. No more
Hi.
Dont know if this is an Asterisk or Dahdi or LibSS7 Error. So Im writing to
Asterisk List.
If somebody knows where to search (dahdi lists or libSS7 lists) will be
appreciated.
Im getting this error after a certain time,
My config is:
Hardware: 3 Digium Quad E1 TE4XXP
libss7 version:
Hi All,
Is it possible o know how many call are into hold ?
who are on hold ?
By whom these extension are on hold ?
And after 60 sec asterisk will call them automatically as Call Parking do?
I wan to make this concept to my PBX system...
Thanks in advance
--
-
Thanks and regards
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Tuesday, April 19, 2011 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many calls are into
Hi Bilal,
Probably there is no open source tool or a good ones available. But few of
them I worked with provide up to 2 users free of cost license type of
reporting. Reporting for Call Centers can get very complicated. Once you
explore some of the commercial apps you will notice how extensive
Solution: The problem is not actually with number of threads.. It is with the
stack size.
Just reduce stacksize per thread and it allowed thousand of calls :) Also use
same configuration to sipp client and server.
ulimit -s 1024
From: satish...@hotmail.com
To:
I personally know one of the developers of some commercial billing
system, supporting what you have written.
Not sure for Web site for billing itself, but one of their the
projects, using the billing system is www.sip.tj with user panel is at
cab.sip.tj
All runs Linux. Billing owner can be
On 04/14/2011 05:57 AM, Niccolò Belli wrote:
Il 14/04/2011 12:25, Larry Moore ha scritto:
I made a suggestion on how you could check this i.e. have your incoming
call go directly to the fax extension, my 1.8.3.2 installation
immediately negotiates a T.38 connection in this sceanrio, of course I
On 19/04/11 1:19 AM, Eduardo Leones wrote:
Anyone know a good IAX2 softphone for Windows that has g729 and it is free?
That's not going to happen. g729 is not free so how can a softphone
that uses it be free unless it isn't honouring the licenses.
Don't use this as an excuse to discuss the
On 20/04/11 1:58 AM, Mark Deneen wrote:
2011/4/19 Niccolò Bellidarkbas...@gmail.com:
Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto:
Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and
internet is offline.
srvlookup = no didn't help.
What about putting my
Does anyone have a recommendation for an Open Source IP Address Management
solution please?
There are several commercial players such as BlueCat, BT Diamond, InfoBlox,
VitalQIP. And, Solarwinds makes a module that focuses on IPAM.
Most vendors tie logic into DNS and DHCP into IPAM designs. In
Hi,
Is there a step by step guide to Configure IVR(Inbound and Outbound) in
AsteriskNow using FreePBX ?
Thanks
Kaushal
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New to Asterisk? Join us for a
Hello List,
The requirement is little complicated as it is H/W specific.
Basically we are integrating a radio gateway (SIP) with asterisk. The gateway
will be connected to a meetme room, so that any operator (with IP phone
registered as SIP user to asterisk) can login to the room and listen to
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