Check who is dialing this line by CallerID, if it is not your user - just
drop the call.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of gilbert
saunders
Sent: Thursday, April 17,
He said outgoing calls. Its simple. Just put it in a separate zap group,
structure your dialplan (with AGIs or GotoIfs) so that only a particular
user dials on it.
On Thu, 2008-04-17 at 09:17 +0300, Mindaugas Kezys wrote:
Check who is dialing this line by CallerID, if it is not your user –
just
Simon wrote:
Hi There,
We have our Asterisk box using a external SIP provider for outgoing
calls over our DSL line. This seems to be going well... But i do have
the ability to set some QOS ports in our linksystem DSL router... Its
faily basic, so im wondering if it will help at all...
We
--- Kevin P. Fleming [EMAIL PROTECTED] wrote:
Vieri wrote:
So basically I'm wondering if the Asterisk
make/configure process could do steps 1 and 2
automagically for me.
I can't find any other Linux distribution that
provides libilbc, so this
would be a very Gentoo-specific change
On Wed, 16 Apr 2008, lordfuknowsyou wrote:
My thoughts now are to actually do a hangup at the end of the RxFAX and
rely on a 'h' extension to pick it up and carry on with the 2nd half
(which is PDFing and emailling the fax), but I'm concerned I'm going to
lose the channel variables as it
Hi, i have two computer with asterisk.
One is a SIP proxy that Dial() the other.
It is possible to be sure that the proxy does not make transcoding in
any case and Hangup() the call if the Second asterisk does not support
the codec ?
Thanks
___
--
Hi, i have two computer with asterisk.
One is a SIP proxy that Dial() the other.
It is possible to be sure that the proxy does not make transcoding in
any case and Hangup() the call if the Second asterisk does not support
the codec ?
Thanks
___
--
Hi all,
I have been seeing a lot of the following warning messages on my asterisk
cli. Can naybody tell why these messages are showing up. I am using only SIP
to make calls from m asterisk.
[Apr 17 04:52:24] WARNING[2512]: chan_sip.c:6480 determine_firstline_parts:
Bad request protocol Bad event
Hello. I'm trying to use gmail's imap feature w/ asterisk imap voicemail.
I compiled c-client with the following settings: make lr5 IP6=4
and asterisk with: ./configure --with-imap=/usr/src/imap-2007a/
However if i enable any if the imap settings in voicemail.conf, asterisk
starts acting funny
In article [EMAIL PROTECTED],
Rizwan Hisham [EMAIL PROTECTED] wrote:
I have been seeing a lot of the following warning messages on my asterisk
cli. Can naybody tell why these messages are showing up. I am using only SIP
to make calls from m asterisk.
[Apr 17 04:52:24] WARNING[2512]:
Vieri wrote:
Did you try a show channels to see if there were
stale channels for peer 200?
I had the same problem you describe but it was due to
hung channels (used * 1.4.18.1 with rtp*timeout and
saw inuse peers during the pre-timeout periods even
though the agents weren't on a call).
I just saw the sip debug and its showing that for every notify request,
asterisk is sending a bad request response.
here is the debug
--- SIP read from 70.80.000.00:1031 ---
NOTIFY sip:69.90.111.11:9060 SIP/2.0
Via: SIP/2.0/UDP 70.80.000.00:1032;branch=z9hG4bK-ade549bd
From: Blake sip:[EMAIL
Hi,
I have 2 wireless phones. I tried to register both the phones with the same
number say 3000 to asterisk. But at any time i am able to see that only one
phone is being registered. I want to test the call forking feature. How do I
do this? Please help me in this regard.
Thanks Regards,
Hi all,
i want to buy a pci or whatever card for asterisk to plug in my telephone
line into it and use asterisk as a pbx. i have only one telephone line at
home. can you recommend me a simple cheap card which i can buy in pakistan.
I live in pakistan, and i dont know any dealers here who sell
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Simon wrote:
| Is this worth doing? If so, what ports should i specifiy?
http://www.bricklin.com/qos.htm
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.6 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
Asterisk only allows a single contact per SIP account so to do forking
you'll need to use two SIP accounts and put them both in the Dial
command. Or you could use OpenSER.
Regards,
Greyman.
___
-- Bandwidth and Colocation Provided by
Hello. I'm trying to use gmail's imap feature w/ asterisk imap voicemail.
I compiled c-client with the following settings: make lr5 IP6=4
and asterisk with: ./configure --with-imap=/usr/src/imap-2007a/
However if i enable any if the imap settings in voicemail.conf, asterisk
starts acting
On Thu, 2008-04-17 at 07:16 -0400, sil wrote:
Simon wrote:
| Is this worth doing? If so, what ports should i specifiy?
http://www.bricklin.com/qos.htm
Yeah, well, that's all fine and dandy as long as more capacity is an
option. Many people are already subscribed to the most capacity
Hi,
I'm hoping that somebody could possibly assist me with this. I've tried
everything and I believe that my settings and configurations are 100% -
CentOS 5.1 - 2.6.18-53.1.14.el5
Asterisk 1.4.19
libpri-1.4.3
zaptel-1.4.9
Connected via a Digium TE122P to a E1 PRI
Incoming on any one of the
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Brian J. Murrell wrote:
|
| Yeah, well, that's all fine and dandy as long as more capacity is an
| option. Many people are already subscribed to the most capacity
| available to them and using it.
|
| b.
Apparently man people don't understand
5060?
[EMAIL PROTECTED] ~]# netstat -an | grep 5060
udp0 0 0.0.0.0:50600.0.0.0:*
how then do I tell it to use imaps?
On Thu, Apr 17, 2008 at 2:38 PM, Yehavi Bourvine +972-8-9489444
[EMAIL PROTECTED] wrote:
Hello. I'm trying to use gmail's imap feature w/
On Thu, 2008-04-17 at 07:54 -0400, sil wrote:
Apparently man people don't understand that those QoS settings on
routers mean little most of the time. Most providers resell QoS as a
premium service, so while many waste their time painting their packets
those markings get stripped.
Maybe your
--- Nestor A. Diaz [EMAIL PROTECTED] wrote:
Vieri wrote:
Did you try a show channels to see if there were
stale channels for peer 200?
I had the same problem you describe but it was due
to
hung channels (used * 1.4.18.1 with rtp*timeout
and
saw inuse peers during the pre-timeout
Rizwan Hisham wrote:
Hi all,
i want to buy a pci or whatever card for asterisk to plug in my
telephone line into it and use asterisk as a pbx. i have only one
telephone line at home. can you recommend me a simple cheap card which
i can buy in pakistan.
I live in pakistan, and i dont
Hi guys,
What are some reliable sip to FSX gateways with four ports and eight ports?
I've used some Linksys and Grandstream devices and I find that at
unexplained times there will be echo on the line. Sometimes this happens on
the end where the devices is placed and sometimes this happens on the
In article [EMAIL PROTECTED],
Rizwan Hisham [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
I just saw the sip debug and its showing that for every notify request,
asterisk is sending a bad request response.
here is the debug
--- SIP read from 70.80.000.00:1031 ---
NOTIFY
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Hash: SHA1
Brian J. Murrell wrote:
| Maybe your understanding of QOS and mine is different. Of course I have
| no illusions that I can assign a priority to my packets that is going to
| be meaningful to anyone once they leave my network.
|
| But certainly at
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users
On Thu, 2008-04-17 at 08:36 -0400, J. Oquendo wrote:
Brian J. Murrell wrote:
| But certainly at my choke point which is of course my Internet uplink,
^^^
I
| can apply QOS (i.e. traffic shaping, which is what the OP's
On Wed, Apr 16, 2008 at 7:18 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Ex Vito wrote:
Tested with no 4K stack kernel and stackcleanup svn branch
zaptel version. Correct, the kernel no longer complains about
the soft hangup.
However the system still hangs (console
I think videoreps.net It´s not free.
But, I discover that I really need is click-to-talk, excuse me.
On Wed, Apr 16, 2008 at 5:05 PM, Bob G [EMAIL PROTECTED] wrote:
Introducing Click-to-Call http://1ezphone.com/
Posted: 16 Apr 2008 9:55 AM PDT
The 1EZphone browser softphone has created
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Brian J. Murrell wrote:
| Not at all little. If you have a lot of low priority outgoing traffic
| (i.e. p2p) saturating your link, uplink traffic shaping will mean the
| difference between a completely unintelligible call and something very
|
On Thu, 2008-04-17 at 09:25 -0400, J. Oquendo wrote:
Is it? So you're telling me if you're saturated on the way in, fixing up
your packets on the way out is the solution.
I think I've made it clear that my argument is only about uplink shaping
and the requirement for it given the asymmetric
Steve Rawlings wrote:
exten = 596,n,ChanSpy(|g(2000))
...snip...
This worked fine with 1.4.18.1. With 1.4.19 if I dial 596 I get answered
but there's no spying, the only way I could get this to work was with -
exten = 596,n,ChanSpy(|b)
but this spied on all channels, not just those
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Brian J. Murrell wrote:
| I think I've made it clear that my argument is only about uplink shaping
| and the requirement for it given the asymmetric nature of a lot of last
| mile connections existing today. Funny enough that is *exactly* what
|
On Thu, Apr 17, 2008 at 2:36 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Apr 17, 2008 at 02:20:57PM +0100, Ex Vito wrote:
- Should this be considered a regression ?
Yes, it is a regression, and thus a bug.
Mattf has already offered you to work with him on resolving this.
May I suggest the following read:
A Beginners Guide To Successful VOIP Over DSL
http://www.smallnetbuilder.com/content/view/30340/83/
Which covers both QoS and traffic shaping in small routers. It was
written based upon my own experience with both Asterisk and hosted PBX
providers.
Michael
Hi all,
I've been googling for a solution here and haven't really come up with
anything yet. We're doing an Asterisk install for a local radio station,
and we're looking for a phone that they can use in their control room
hooked up to their mixer board for recording calls. So, when you phone
in
Hi everybody,
I need to use different outbound routes from calls started by different
extensions; I mean, that the extension A when dialing 011543... has
to get access always on the 1st trunk, the extension B when dialing
another number has always to access the outside world on the 2nd trunk,
and
How about avoiding the phone entirely in the playback phase?
Have asterisk record the call to disk in MP3 or Slin, then use a pc with decent
audio card
to read it off the shared disk and feed it to the mixer.
Tim.
- Original Message -
From: Bob Pierce [EMAIL PROTECTED]
To: Asterisk
On Thu, Apr 17, 2008 at 9:44 AM, Sean Bright [EMAIL PROTECTED] wrote:
Steve Rawlings wrote:
exten = 596,n,ChanSpy(|g(2000))
...snip...
This worked fine with 1.4.18.1. With 1.4.19 if I dial 596 I get answered
but there's no spying, the only way I could get this to work was with -
Create different contexts and assign them to the extensions
[trunk1]
exten = .X,1,Dial()
[trunk2]
exten = .X,Dial()
and in sip.conf or iax.conf
[exten1]
...
context = trunk1
[exten2]
context=trunk2
Marco escribió:
Hi everybody,
I need to use different outbound routes from
Hi, everyone.
I'm having a problem with qualify=yes sip.conf option. Sometimes, when a
device registered with asterisk goes offline, I'm not getting a message
about it in /var/log/asterisk/messages log. Sometimes the same happens
with REACHABLE message, when a device comes back online. I'm pretty
My personnal experience is if you`re looking for an inexpensive solution
(SOHO), StreamEngine based routers (a lot of D-Link products are
Streamengine based, for example the DI-724GU and the DIR-655) do a decent
job of providing QoS on the upstream, which is where you (usually) need it
anyways.
The asterisk code is full of fun things where it checks for things like
that in multiple places but doesn't always handle every instance of the
same check in the same way. This is getting resolved piecemeal and will
eventually be minimized as the application develops, but I do not think
things
Mike wrote:
do a decent
job of providing QoS on the upstream, which is where you (usually) need it
anyways.
QOS can only be on outgoing, you can't set the priority of a packet
after you receive it. The only other solution would be the cooperation
of the ISP to provide QOS upstream of you.
ok, thanks, does rtp*timeout work if i have canreinvite=yes ? since rtp
traffic is not passing thought asterisk, or i have to put canreinvite=no ?
slds.
rtp*timeout for sip peers is not a fix but a
workaround.
Try to set both values and reload sip.
Then when you witness what you posted try
On Thu, 2008-04-17 at 11:40 -0400, Chris Mason (Lists) wrote:
Mike wrote:
do a decent
job of providing QoS on the upstream, which is where you (usually) need it
anyways.
QOS can only be on outgoing,
Which is what he meant when he said upstream I believe.
you can't set the priority
Hello,
I need to read the Status-Line (I need to know if it's 603, 503, 404)
after a Dial. I have tried:
exten = s,2,Dial(SIP/[EMAIL PROTECTED],,tTwW)
exten = s,3,Set(t=${SIP_HEADER(Status-Line)})
But t is empty
I have also tried:
exten = s,5,Verbose(*** STATUS: ${DIALSTATUS})
exten =
That makes PERFECT sense and also makes me aware that I need to review
asterisk theory :-P
I'll put it under test and let you know how it works.
Thanks a lot!
Marco
Rodrigo Gonzalez ha scritto:
Create different contexts and assign them to the extensions
[trunk1]
exten = .X,1,Dial()
Hi,
I want to know if I am running two machines each with its own Asterisk on my
LAN, show I change the port of one of the Asterisk to something like 5061?
Otherwise, how does an external SIP client ( like IPkall.com) knows how to
route DID call to Asterisk? What is the solution for this kind
I need a refresher course on how many licenses I need to buy. I have
an Asterisk server that receives calls by SIP (G729) and then sends them
to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if
the license is per channel or per call so I do not know if I need 32 or
64
Steve Totaro wrote:
Should one have to change their dialplan for functionality to remain
the same in the same version?
I wasn't suggesting it wasn't a regression, just making the OP aware
that he can pass multiple arguments to a dialplan application (i.e.
ChanSpy(|bg(2000)))
He mentioned
Afaik its per encode / decoder pair.
In this case you will need 32 simultaneous encoders / decoders between
g729 and slin, so you would need 32 licenses.
Contact digium sales/support directly and you will know for sure :)
Zoa
Carlos Chavez wrote:
I need a refresher course on how many
http://store.digium.com/productview.php?product_code=G729CODEC
http://www.digium.com/en/docs/G729/g729policy.php
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
On Thu, Apr 17, 2008 at 11:14 AM, Carlos Chavez [EMAIL PROTECTED] wrote:
I need a refresher course on how many licenses
On Thu, Apr 17, 2008 at 12:03 PM, Pete Kay [EMAIL PROTECTED] wrote:
Hi,
I want to know if I am running two machines each with its own Asterisk on my
LAN, show I change the port of one of the Asterisk to something like 5061?
Otherwise, how does an external SIP client ( like IPkall.com) knows
Is there a way to specify per user attachment options for voicemail, from
within users.conf?
I know I can enable or disable it globally in voicemail.conf, but I have
certain users that like the attachment feature, and others that don't.
Also, can you enable/disable per user the deletion if
you surely are using port forwarding right now, something like this:
wan(5061) -you_router_here--
lan(5061)---Asterisk_box_1(5061)
so you only need to add this :
wan(5062) -you_router_here--
lan(5061)---Asterisk_box_2(5061)
just tell your provider that second
On Fri, 18 Apr 2008, Pete Kay wrote:
I want to know if I am running two machines each with its own Asterisk on my
LAN, show I change the port of one of the Asterisk to something like 5061?
It is common to have multiple instances of Asterisk listening to the same
port number on the same LAN.
Guys,
Sean Bright wrote:
Steve Totaro wrote:
Should one have to change their dialplan for functionality to remain
the same in the same version?
I wasn't suggesting it wasn't a regression, just making the OP aware
that he can pass multiple arguments to a dialplan application (i.e.
We're having some difficulty tying together our Cisco and Audiocodes
syslogs with our Trixbox asterisk logs.
We'd like to have some way to split out a single call from all the
activity going on at one moment.
Obviously NTP is the first step for this, but we haven't found any
means to tie the
My own Chanspy(g(GROUPNAME)) works 2 times out of three (roughly). The
other time, it crashes Asterisk. Using 1.4.19 too.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Rawlings
Sent: Thursday, April 17, 2008 14:10
To: Asterisk Users
Ah. My apologies for the confusion. Not that it helps you a great
deal, but I am running ChanSpy successfully in production (as we speak)
with 1.4.19 with no crashes or the like:
ChanSpy(SIP/11,g(Spyable))
Maybe its only a problem if no channel spec is passed?
Steve Rawlings wrote:
--- Nestor A. Diaz [EMAIL PROTECTED] wrote:
ok, thanks, does rtp*timeout work if i have
canreinvite=yes ? since rtp
traffic is not passing thought asterisk, or i have
to put canreinvite=no ?
In my setup it doesn't really matter since calls are
coming in through PSTN-IVR-QUEUE-SIP
At 05:59 AM 4/17/2008, you wrote:
Not at all little. If you have a lot of low priority outgoing traffic
(i.e. p2p) saturating your link, uplink traffic shaping will mean the
difference between a completely unintelligible call and something very
acceptable.
My network looks like this:
Cable
Hi,
My dialplan works fine with one user (asking for the sharp key to be
pressed to continue, and others), but when 2 users are calling at the
same time if one press key # the two users are jumping to the next step.
Anyidea ?
FYI, I'm using Asterisk 1.4.10.
--
Cyril SCETBON
Logs?
On Thu, Apr 17, 2008 at 11:47 PM, Cyril SCETBON [EMAIL PROTECTED]
wrote:
Hi,
My dialplan works fine with one user (asking for the sharp key to be
pressed to continue, and others), but when 2 users are calling at the
same time if one press key # the two users are jumping to the next
Mike wrote:
My own Chanspy(g(GROUPNAME)) works 2 times out of three (roughly). The
other time, it crashes Asterisk. Using 1.4.19 too.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Rawlings
Sent: Thursday, April 17, 2008 14:10
Raúl Gómez C. wrote:
Hi list,
snip
I think this is a very common scenario so, how are you doing to handle this
situation???
What if you were to set an account code to the extension that is
requesting the long-distance call?
So person at extension 111 requests a long distance call to
J. Oquendo wrote:
Its fine and dandy, but the problem is you're still getting 5 packets.
You're still saturated period. No QoS in the world outside of your
provider and more bandwidth can alleviate that. Your provider is not
going to care what you do once its passed to the CPE. So look at it
J. Oquendo wrote:
it does, when someone can realistically point this out please let me
know so I can switch from a DS3 to T1 and save money.
Use the T1 for voice and get a DSL modem for your data use? :)
___
-- Bandwidth and Colocation Provided
Apparently, there is a SIP(diversionheader) field that fixes the problem
below, but I cannot find any docs or examples of how to use it in my
dialplan. Any help would be appreciated. We have a Cisco CallManager
where users forward their numbers, so PSTN-PSTN calls get this error...
-Greg
---
On Thu, Apr 17, 2008 at 2:15 PM, Henry Cobb [EMAIL PROTECTED] wrote:
We're having some difficulty tying together our Cisco and Audiocodes
syslogs with our Trixbox asterisk logs.
We'd like to have some way to split out a single call from all the
activity going on at one moment.
Obviously
On Thu, Apr 17, 2008 at 5:14 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
Raúl Gómez C. wrote:
Hi list,
snip
I think this is a very common scenario so, how are you doing to handle this
situation???
What if you were to set an account code to the extension that is
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Michelson
Sent: Thursday, April 17, 2008 17:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Mike wrote:
My own
I saw a patch attached to that bug report, just download it run patch
and then make clean make install, restart asterisk and you should be
smokin.
Mike wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Michelson
Sent: Thursday,
Hi,
I have searched through the archives on this mailing list, but didn't
find a solution to the outboundproxy problem. Can someone please help?
I wish to configure Asterisk such that all outgoing SIP requests get
relayed to an outboundproxy, instead of the actual recipient directly.
In my
There are lots of different ways to configure Asterisk and SER to get
them working together depending on what you want to do.
The link below is not a bad starting point.
http://www.voip-info.org/wiki-Asterisk+at+large
Asterisk has outboundproxy and outboundproxyport settings that can be
used in
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