On 01/10/09 00:57, Kirill 'Big K' Katsnelson wrote:
Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID
calls, originating and transferring.
A provider offers both SIP and IAX trunking. Cateris paribus, what is
the preferred solution to choose? What points to consider?
We
Alan Lord (News) wrote:
On 01/10/09 00:57, Kirill 'Big K' Katsnelson wrote:
Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID
calls, originating and transferring.
A provider offers both SIP and IAX trunking. Cateris paribus, what is
the preferred solution to choose?
Hello
I am thinking to develop a softphone that is integrated into web.(in form of
APPLET or some thing else)
Ie a user with with just a PC with Net Browser(fire fox etc) Installed can
make call..
Is there some thing developed before like this that is open source ??
--
Best Regards
Shakeel
Bumping this in the hope that it is seen by people who missed it before.
Ishfaq Malik wrote:
We have a customer who connects PBX boxes (Avaya etc.) to our asterisk
server (1.4.17) as a SIP extension. This customer needs the dialled
number sent to the PBX as well as number that the call is
Hi All,
While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use it in
my dialplan.
I had 2 sip extensions 555 and 666 and I called from 555 to 666, but
unfortunately no value for ${RTPAUDIOQOS} appeared on Asterisk CLI.
Would you please let me know what is wrong with my dialplan
ABBAS SHAKEEL a écrit :
Hello
Hi
I am thinking to develop a softphone that is integrated into web.(in form of
APPLET or some thing else)
Ie a user with with just a PC with Net Browser(fire fox etc) Installed can
make call..
Is there some thing developed before like this that is open
Thanks.
But Can i enhance it in such away that it can make calls to asterisk as part
of a web application ??
user can call from webapplication
i think mozphone is a plugin for mozilla...
On Thu, Oct 1, 2009 at 2:00 PM, Administrator TOOTAI ad...@tootai.netwrote:
ABBAS SHAKEEL a écrit :
On 1/10/09 5:56 PM, das sandesh wrote:
Hi All,
I have a problem, when I was doing a performance testing using an
asterisk server: Quadcore processor, 4GB RAM, CentOS5.2, after 150-151
calls all the other calls are giving busy, I tried to do ulimit related
stuff, like increasing the soft and
On 1/10/09 9:24 PM, Ishfaq Malik wrote:
Bumping this in the hope that it is seen by people who missed it before.
Ishfaq Malik wrote:
We have a customer who connects PBX boxes (Avaya etc.) to our asterisk
server (1.4.17) as a SIP extension. This customer needs the dialled
number sent to the
Hello. I set up an Asterisk box a couple days ago and was having problems
with not being able to hear SIP clients. After some troubleshooting we have
determined that hte INVITE is sending my local(192.168) IP. How would I get
* to send the public IP instead of the local one? I have changed every
On 1 Oct 2009, at 10:43, Mike Bessette wrote:
Hello. I set up an Asterisk box a couple days ago and was having
problems with not being able to hear SIP clients. After some
troubleshooting we have determined that hte INVITE is sending my
local(192.168) IP. How would I get * to send the
OK. Here is the relevant section of my sip.conf
[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes,
this
can also be set to 'osp'
; if asterisk was compiled
This week Steve Sokol stops by to describe and field questions about
Digium's new affordable speech recognition solution. Later on in the
call, we'll also be looking at iVoIP, clients and uses for mobile
VoIP.
Join us on IRC anytime #voip-users-conference
During the conference, call via SIP g711
Hello,
Has anyone encountered that when Asterisk sends RTCP messages, it stops
sending RTP packets until it gets an answer?
Can that be fixed?
Thanks.
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Hi All
I having an intermittent problem with the above mobile gateway and would
appriciate some advice
basically 1 in 10 calls fail at some point during the call, the duration of
the calls ate completely different
call progression
Call comes in from Zap channel and dials a mobile number on the
We had many problems with IAX2, changing to SIP solved them all.
Let me paste link to wise-words which clearly illustrates our experience:
http://wiki.kolmisoft.com/index.php/Why_we_do_not_suggest_to_use_IAX2
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
Using 1.4 svn, I want to implent the busy application.
With the following dialplan:
[inboundqueue]
exten = _X.,1,Answer()
exten = _X.,n,Goto(dropcall,1)
...
exten = dropcall,1,Busy(10)
exten = dropcall,n,hangup()
If I call any number in the inboundqueue, I get the following:
[Oct 1
Hi Matt,
When I get can more that 150 calls, i get a busy signal (Congestion) for the
calls above 150 - says your call cannot be completed now, its allowing
only 150 callsIs there any thing related to field descriptors from linux
point of view that I need to increase inorder to increase the
how can i used this patch with digium cards,
i have digium card and also having some issue in recording ,
can you give me procedure for it?
regards
Dhaval
On Thu, Oct 1, 2009 at 7:37 AM, Martin asteriskl...@callthem.info wrote:
That's nice. At least now peopel that want to do call recording
Mike –
It looks like you have externip set but no localnet setting.
You need to set localnet for your internal networks so that Asterisk knows when
to properly apply the externip setting.
sl
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OK so basically just uncomment the the localnet settings hten?
On Thu, Oct 1, 2009 at 8:15 AM, Scott L. Lykens slyk...@verimedservices.com
wrote:
Mike –
It looks like you have externip set but no localnet setting.
You need to set localnet for your internal networks so that Asterisk knows
Ishfaq Malik wrote:
Bumping this in the hope that it is seen by people who missed it before.
Ishfaq Malik wrote:
We have a customer who connects PBX boxes (Avaya etc.) to our asterisk
server (1.4.17) as a SIP extension. This customer needs the dialled
number sent to the PBX as well as
Cyprus VoIP wrote:
Has anyone encountered that when Asterisk sends RTCP messages, it stops
sending RTP packets until it gets an answer?
There is no such thing as an RTCP 'answer'.
Can that be fixed?
If it is a real problem, of course it can be fixed. The first step to
doing so would be to
Julian Lyndon-Smith wrote:
Using 1.4 svn, I want to implent the busy application.
With the following dialplan:
[inboundqueue]
exten = _X.,1,Answer()
exten = _X.,n,Goto(dropcall,1)
...
exten = dropcall,1,Busy(10)
exten = dropcall,n,hangup()
If I call any number in the
Hei!
Here's my problem. I have an Asterisk with SS7 and SIP trunks. Asterisk
version is 1.6. I'm setting up a custom CDR fields and I was wondering
is there a way to know who initiated a hangup? Asterisk must be aware of
that info somehow, cause in queue_log, that info is present
Mike -
Uncomment and set appropriately for your network. If you're using
192.168.1.0/24 as your internal network then that's what it should be set to.
Be sure to include any private networks that may interact with the server over
VPN or private circuits as well.
Then be sure to reload or
Found it, I use the g flag in Dial command, that helps :)
Rennes
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rennes Neps
Sent: 1. oktoober 2009. a. 16:05
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Is there
I have an Asterisk 1.4.2 system that has been installed for about 3
months now in our home. We converted all of our phones to SIP phones,
and use two different trunk providers (BroadVoice for incoming
FlowRoute for outgoing).
Most of the time its working flawlessly. But about 1/3rd of the
Still no luck. I'm almost ready to start over with a fresh sip.conf and
extensions.conf. Does anyone kno where I can find one without all the
comments and other fluff?
On Thu, Oct 1, 2009 at 9:22 AM, Scott L. Lykens slyk...@verimedservices.com
wrote:
Mike -
Uncomment and set appropriately
Mike –
Your original post indicates the trouble is with audio.
What kind of firewall are you passing through?
If it’s PIX or ASA, I’ve found the most reliable route is to enable SIP inspect
on the PIX/ASA and remove any externip/localnet configuration from Asterisk.
This way the
Right now I have all firewalls and such turned off. When I have the firewall
enabled, I use the one built in to the Tomato firmware on my Asus router.
How could I determine if this is a PIX/ASA firewall?
On Thu, Oct 1, 2009 at 10:33 AM, Scott L. Lykens
slyk...@verimedservices.com wrote:
Mike
On Thu, Oct 1, 2009 at 7:57 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:
how can i used this patch with digium cards,
i have digium card and also having some issue in recording ,
can you give me procedure for it?
May be Martin can help with that, I don't know how to setup Digium
Moises Silva wrote:
May be Martin can help with that, I don't know how to setup Digium
boards in high impedance mode. It seems the feature may not be exported
via configuration files yet, so changes to the driver may be needed?
That is correct, none of our drivers currently expose a method to
Mike -
If your router/firewall does not have any kind of SIP protocol-specific support
then you need to set up port forwarding on your router.
Forward udp/5060 for signaling, and the matching udp ports as listed in your
rtp.conf, to your Asterisk box. Keep the externip and localnet settings in
At 07:10 AM 10/1/2009, you wrote:
I have an Asterisk 1.4.2 system that has been installed for about 3
months now in our home. We converted all of our phones to SIP phones,
and use two different trunk providers (BroadVoice for incoming
FlowRoute for outgoing).
Most of the time its working
I'm using the latest asterisk-1.4.26.2 and no zaptel trunks used, all SIP.
I have one user that is having problems once he connects to asterisk.
He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk)
which goes to my asterisk IVR.
If he presses a dtmf during any message, the
anyone can just grab the PEF framer datasheet and tweak the driver though...
last I checked there's a whole section devoted to high impedance in
the datasheet
Martin
On Thu, Oct 1, 2009 at 9:56 AM, Kevin P. Fleming kpflem...@digium.com wrote:
Moises Silva wrote:
May be Martin can help with
Maybe the GSM carrier is disconnecting you ???
Just a wild guess. They sometimes do that if they have to free
the channel ... for a better paying customer :)
Martin
On Thu, Oct 1, 2009 at 6:09 AM, robert boardman
robert.board...@gmail.com wrote:
Hi All
I having an intermittent problem with
Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in
sip.conf but not iax.conf
Thanks
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now:
I actually see the TOS setting in iax.conf, but the default (commented out)
is EF - which doesn't even match a valid bit combination according to
voip-info wiki
If this is the right place, what TOS value are people using succesfully over
an ADSL connection?
_
From:
I checked the source for reading of configuration options but I didn't see
anything in vm_execmain()
This is the line of code that is bothering you
cmd = get_folder2(chan, vm-savefolder, 1);
On Mon, Sep 28, 2009 at 8:41 AM, Mike l...@virtutel.ca wrote:
I am looking to
Did you look at this wiki -
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf ?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Thursday, October 01, 2009 1:36 PM
To: 'Asterisk
Michelle Dupuis wrote:
I actually see the TOS setting in iax.conf, but the default (commented out)
is EF - which doesn't even match a valid bit combination according to
voip-info wiki
If this is the right place, what TOS value are people using succesfully over
an ADSL connection?
That link is great thanks.
From what I read elsewhere, ToS is just the first 3 bits which should be
honored by DSCP (first 5 bits)- even old equip should be DSCP
compatible...or I need to do more reading :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Yea, kinda, sorta. The DSCP is six bits, which occupy six of the 8 bits
in what is/was the type of service byte in an IP packet. Three of the 6
DSCP bits reside over the old precedence field and three reside over the
old low delay, high throughput and high reliability fields (those three
On 2/10/09 12:41 AM, das sandesh wrote:
Hi Matt,
When I get can more that 150 calls, i get a busy signal (Congestion) for
the calls above 150 - says your call cannot be completed now, its
allowing only 150 callsIs there any thing related to field
descriptors from linux point of view that
Hello All,
Can anyone help me with False Answer Supervision problem with TDM410P
card... I have Asterisk 1.4.25 with DAHDI 2.1.0.4 installed and
everything works fine except the Answer supervision...
When the call hits Asterisk it sends the call to one of the TDM410 card
and the call is
Assuming you're using POTS, you probably won't have much luck with this. If
you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and asterisk
won't process the line until you pick up and punch a dtmf key. If you are
using E1 or PRI, there is more hope for you.
-Original
Ira writes:
Very similar to what I have. Also Flowroute for outgoing but others
and a TDM400 for incoming. Since upgrading to 1.6.2 from 1.2.28 or so
and figuring out DAAHDI and HPEC on the new version there have been
no echo issues at all. Also cable modem but only the slow version.
There
The extension does exist, as the other caller is redirected to the room.
Here's the relevant lines in extensions.conf:
[dynamic-nway]
exten = _XXX,1,Answer
I've been trying to get this to work on and off for a while now, and it's
time to get serious. If someone would like to get paid for
At 02:53 PM 10/1/2009, you wrote:
I am curious about the fact that you said after upgrading to 1.6.2, your
problems went away. I didn't start with that version because it wasn't
the current production version at the time. Do you think it would be
beneficial to migrate to that version for me?
I
Danny,
Thanks for your reply...
Yes these are POTS line and I am not calling myself... Any other
suggestions?
Cheers,
Nitesh
Danny Nicholas wrote:
Assuming you're using POTS, you probably won't have much luck with this. If
you are calling yourself, you can do Dial(DAHDI/8c/3602045,20) and
Hi All,
I am using Asterisk 1.4.26.2 and I am getting the following problem
making connections to this server. My other servers are Version 1.2.x
which have no problems and this 1.4.26.2 server can call the other 1.2.x
servers.
The error is:
chan_iax2.c:4251 handle_call_token: Call
if a user calling you hears echo of himself then it's the fault of
your sip device/sip phone.
The manufacturer must be using a cheap or an open source echo canceller ...
try getting a different sip device made by some 'normal' company like
polycom or linksys/cisco
Martin
On Thu, Oct 1, 2009 at
Are you in US ?
do you have the proper keywords in zapata.conf/chan_dahdi.conf like
callprogress=yes etc ?
Martin
On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha nit...@vipernetworks.com wrote:
Danny,
Thanks for your reply...
Yes these are POTS line and I am not calling myself... Any other
I'm quite new to all this but I was under the impression that most
electrically induced echo was at the physical interface to the PSTN. If
one is using SIP trunking, I would think this would point to a carrier
issue.
We also hit an interesting problem with echo today but I don't think
this is
Are you saying there are half duplex phones out there with half
duplex speakerphones ?
All analog phones are full duplex ...
Anyways the echo can be created by the analog phone even when it's
connected to the
sip ata or even the sip phone ... then you usually have acoustic echo
which goes
Indeed there are! - John
On Thu, 2009-10-01 at 20:18 -0500, Martin wrote:
Are you saying there are half duplex phones out there with half
duplex speakerphones ?
All analog phones are full duplex ...
Anyways the echo can be created by the analog phone even when it's
connected to the
I had a problem between my 1.6.0 server and a 1.4 server trying to call
through iax and I just put
requirecalltoken=no in the stanza and that fixed the problem.
Klaverstyn, David C david.klavers...@intergraph.com wrote:
Hi All,
I am using Asterisk 1.4.26.2 and I am getting the following
Thanks Martin,
Well the Asterisk is in Fiji and we have check with the Telco on
Reverse Polarity and they said it is setup...
Here is my chan_dahdi.conf:-
#include dahdi-channels.conf
[channels]
language=en
context=incoming
signalling=fxs_ks
busydetect=yes
callprogress=yes
usecallerid=yes
I tried your recommendation. I don't get an error with that but the
call is cancelled with a debug of:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 00011ms SCall: 00269 DCall: 3 [192.168.25.250:4569]
AUTHMETHODS : 3
CHALLENGE
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