Hello,
I am having an issue here that after an attended call transfer there is no
audio on one way; the problem is caused by Asterisk sending two INVITE messages
without waiting for an ack for the first one.
The issue has been reported on bug 9305, has been fixed and the fix is now
included
Hello,
Our Polycom-501 phones are set to retreive their config for the server by a
static configuation defined at the phones (boot servers). Is there any way to
change it remotely? I found no relevant field in the internal WEB browser, nor
anything in the configuration files (sip.conf and
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
Has anyone here implemented Ring back when free in Asterisk?
Here is what I do; the dialplan enclosed is in AEL2 format, but you can get the
idea.
When a call is originated I save the called and callee numbers in a database.
If the user gets busy he/she hangs up and dial *41. I then retreive
Sorry for the hijack, but I was wondering if I could be pointed at how
to get BLF and pickup working with Polycoms? I can use their Buddy
option to get a basic BLF system working, but that does not understand
the ringing state, and does not allow call pickup.
This is what I do (it works most
what do you think is the best way to replicate an asterisk mysql realtime
database ,
i'm setting up a cluster , and i obviously need to ensure the database high
availability
i've been reading about DRBD and the internal replication functionality of
mysql , but i dont really know if its
I am using version 2.2.0.
__Yehavi:
Date: Thu, 6 Mar 2008 15:01:26 +1100
From: Lee, John (Sydney) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT How to Change Polycom Web Admin User:Pass
As far as I recall
Hello,
I've found that my Polycom-501 accepts INVITES from any server in the
world... I would like to restrict it to accept calls only from the servers
listed in its config file, but I cannot find anything in the documentation. Any
idea?
Thanks, __Yehavi:
I could be mistaken, but I think this has to be done physically from
the phone. I don't think you can do this with central provisioning or
from the web interface.
As far as I recall it can be done from the config file only. Here is the
relevant line from sip.cfg:
device device.set=1
If i have this kind of setup, what do i need to make it's load balance.
[ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
| | | |
-
|
Hello,
Here is how I do this. The prerequisits are:
- MySQL to hold the extensions realtime database. MySQL is synchronized
among all servers using the Master/slave replication model.
- The phones are spread by some external algorithm over the Asterisk servers
(statefull load balancer,
What I would like to do is have two identical *
servers which accept registrations of sip extensions
4000-4999.
If I define a rrDNS or LinuxHA then I should have
load-balanced registrations.
However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001 tries to call
Hello,
I've one nokia E65 that works very well with my asterisk box.
The people here don't let me even try it as they are afraid it will consume the
battery more than when it is used the usual way. Is this true?
Thanks, __Yehavi:
Is anyone using a cisco router as an ISDN gateway with Asterisk?
As you might have seen from a couple of my threads, I have been looking at
Fritz! and Cologne cards, both of which require development against a
specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive
and
I'm trying to test IMAP in 1.4.17 and it appears to be not working.
I've compiled imap-2007 with the following on a CentOS 5 box:
make slx EXTRACFLAGS=-I/usr/include/openssl -fPIC
and I've configured and compiled asterisk with the following:
./configure
I've connected some analogic phone to some fxs modules on an analogic card.
I want to disable by default the call waiting sound.
I know that dialing *70 before to call the call waiting is disabled
until the next call, but isn't there a setting or a dialplan command
to set up this
On Friday 04 January 2008 16:45:00 ameel wrote:
Is there a way to disable the b2bua feature in asterisk.
I would like asterisk to work as a sip server and not be involved in the
RTP path between phones.
No. And by the way, b2bua is not a feature. It's is literally what Asterisk
is.
If I
Hello
I'm using eyeBeam, and Asterisk keeps sending my clients a second call,
when they are still in one call (because eyeBeam has lots of channels).
I was using X-Lite (with 3 channels) and Asterisk never sent the client
a second call.
How can I force Asterisk (or eyeBeam) just to send one
What is the easiest (simplest) way to do this?
I do it in two steps: Save the dialled number in Asterisk DB and have a
special extension (*41) which redials it.
Here is the abstract from the dialplan where I save it:
Set(_To=${EXTEN}); // Save the original extension
Hello,
I would like to check whether an extension is busy or not before calling the
Dial() application to it (for example - to play a Busy if it is on
conversation).
How do I check it? In the trunk version there was a function
DEVSTATE(SIP/123), however it does not exist on version 1.4.13...
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My
Sangoma A102D shipped with 2 T1 cables - which I assume are straight
through. Do I need to make crossover cables for this scenario?
As people answered here you need a crossed cable; Note that T1/E1 cables are
different
Hello,
A few days ago I've posted two questions about Polycom phones: How to access
corporate phone directory from the phone and how to use a conference server
with it. After I got zero responses I tried openning a support call in
Polycom's site. Here are the replies I got from them:
-
I've got a few Cisco 7921 wifi phones to use with an Asterisk pilot. For
the purpose of the pilot (i.e. low investment) I want to configure the
phones from the keypad.
Each phone shows settings locked! whenever I try to edit the network
profiles. I can't seem to unlock them! Hopefully
Yehavi Bourvine +972-8-9489444 wrote:
In any case, I'll try this week to upgrade to 1.4.6 version and then add
IMAP
support and inform what happens.
There have been _many_ IMAP related fixes sine 1.4.6. Please try the latest
version, 1.4.13, instead.
--
Russell Bryant
Sorry, I had
Hello,
I have two issues which I would like to know whether someone has an answer to
them:
1. Our institute has over 8,000 phone numbers and I would like to allow
people to search it from the phone. I am willing to write some XHTML
scripts to run through the microbrowser, but I cannot
Isee no one has answered you so far, so I'll try:
I have asterisk ip-pbx on my network, with some grandstream ip phone
and i have cisco gateway that is connetced to VOIP service providers .
Cisco is 3700 series and is using H323 .
i have compiled H323 on asterisk . now i want to
Hello,
I tried a few months ago to use IMAP with Asterisk; I used either 1.4 or the
latest SVN at that time (sorry, don't remember).
After a day I had to remove it since Asterisk crashed, mostly in the IMAP
client code (the code of UW IMAP). My users wants IMAP back (they loved it) but
not
Hello,
Up to a while ago I thought that the released versions are checkpoints of
the trunk versions; however, now I understand they are not, as I see
differences between the two trains. So, what is the relation between them?
Examples for differences:
- When the language is different than
Hello Sean,
Does this clear things up?
Yes! Thanks!
__Yehavi:
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Hello,
I see that most people are using the extensions.conf syntax (most of the
examples and questions here use that syntax). recently I've translated all my
dial plan to AEL syntax and I find it much easier, especially when you need
IFs.
Why most people don't use it? Am I missing
try the astman command.
__Yehavi:
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To
SInce no one else has brought this up, just thought I'd let you know that it
is being worked on...
http://bugs.digium.com/view.php?id=8824
And it works - I am using it for months already.
Note that not all phones support it. Cisco and Policom supports it, while Snom
does not.
Hello,
After upgrading to 1.4.9 the above function does not work anymore; it claims
that child went away while the child is probably not born at all...
Before I open a bug on it, anyone has a clue?
Thanks! __Yehavi:
Hello,
I would like people to use soft phone, but they are used to have the standard
phone handset in their hands... Is there a USB handset or a handset that
connects to the audio card?
Thanks! __Yehavi:
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Hello,
One of our faculties have Microsoft's LCS and would like to connect it to
our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS
talks SIP over TCP with TLS. Anyone can recommend a gateway between these two
protocols?
Thanks!
Hello,
All the examples of SLA talk about Zap channels from one side and SIP on the
other side, while my system is a pure SIP one.
I would like to have two phones having extensions 1 2 defined on them, and
when someone calls extension 1 it rings on both, each one can see its status,
and when
Hello,
A few days ago I've asked about the ability to play a stuttered ringing
tone when the called party is already on the phone. I've found a partial
solution for it.
To describe again the problem: When a user is on a call and someone else
calls him, the caller does not know that the
Hello,
When dialling a SIP phone which is already in a call the caller hears a
regular ringing tone and does not know that the called party is engaged in
another call. Is there a supported way inside SIP to tell the calling party to
play a stuttered ringing tone?
this is a SIP phone configuration issue.
You should tell the UAC to not accept a second call while the line is
engaged (look for a 'Call Waiting' option in the configuration of the UAC)
The UAC will send back a 486 Busy Here error code and the calling
party will get a busy signal from
Hello,
I am trying to apply the called party identification patch (patch 8824) and
managed to make it work with a static data. Where do I take the name of the
called person (the equivalent of CALLERID, but the other way...)?
BTW, one note to the above patch: To make it work the device should
Yehavi wrote:
I am trying to apply the called party identification
patch (patch 8824) and managed to make it work with a
static data. Where do I take the name of the called person
(the equivalent of CALLERID, but the other way...)?
Asnwering myself: I am using realtime extensions, so
Hello,
I would like to implement a few decision making process inside the dialplan
using information stored in MySQL (like LCR, etc.). I see the MYSQL()
application, but as far as I understand I have to connect to the database each
time I want to query it; this seems a CPU eater to me. Is this
Does anyone know the code to dial in Israel on Bezeq to get back the number
of the line that you are on ?
Dial 110 and it will tell you your number.
__Yehavi:
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Hello,
I would like to understand how SLA is working, but all the examples are using
trunk of ZAP type and stations of SIP. However, in my case the stations are SIP
and the outgoing connection to PSTN is SIP also (link to a Cisco gateway).
can anyone send me a simple configuration of how to
Hello from Greece. I have an assignment related to Cisco. Specifically, i
have to connect an asterisk server (VoIP) to a Cisco Router 2851 in a way
that the router will handle some calls. I have not found any manuals
that would be helpful to me. On the other hand, i found some manuals for
Does anybody have callerid name coming in on a Cisco PRI via a Cisco
gateway via SIP to *? I've seen a few people ask and a few people that
say it should work, but I've never seen an actual working config.
I have it working, but it depends on the specific configuration. I have it
working via
Here are the relevant parts from extensions.conf file. It works only for local
extensions whose number id 806xx. Note one thing: When you use the H extension
the generated CDR is wrong - the destination extension is H and not the
original number. I've done some small code change in Asterisk and
Hello,
I've wrote a dialplan script which uses the H extension to do something
similar to what you want. In general it uses the internal ASTDB for this:
- When there is no answer (or busy) the caller hangs up, initiate a new call
with some special code (*41 is used here by the public
On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote:
Hello,
On most SIP phones a conference call is done on the phone and is limited to
3
participants. Polycom phones has a configuration option to use a conference
server instead of the internal conferencing feature. I guess I need
Use Snom phones.
We have had around 6 participants, without problems. In theory you should be
able to have around 12 people on a conference on a snom phone.
I have a few Snom phones here - people do not like them...
Thanks, __Yehavi:
Yehavi,
Can you make a script that uses call files to get everyone into the
conference?
--
Warm Regards,
Lee
Possible, but looks too much cumbersome... However, that's a nice idea.
Thanks! __Yehavi:
Hello,
On most SIP phones a conference call is done on the phone and is limited to 3
participants. Polycom phones has a configuration option to use a conference
server instead of the internal conferencing feature. I guess I need some
conference server; any experience with such a server which
That was exactly what I meant.
Your setup is :
Nortel --- Cisco --- Asterisk
What I was thinking about is:
Nortel --- Asterisk1 --Asterisk2
In previous case, your are using Cisco's QSIG features.
In the latter one, you could use Asterisk QSIG features.
I
Hello,
I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it
on 1.4svn 56126 and it does not recognise this application. Any idea?...
Thanks! __Yehavi:
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do you have some info, about what platforms supports qsig decode? I
found, that first supported in 12.4.(9)T, but I don't know, if only on
28xx or also in older routers (like 3660) with NM-HDV-E1...
I assume that it is a hardware independent but I do not have the hardware to
test...
I see
Do you think it could have been done with another T1/E1Asterisk box between
the Nortel PBX and the other Asterisk server ?
Sorry, I do not understand exactly what you are asking. Do you mean using an
Asterisk with PRI card instead of Cisco? If so, I have no experience with this.
Which
Hello,
I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and
Nortel TX-1. I had problems with name transfer and with the help of Cisco
support I've fixed it. Enclosed here are the definitions needed for it.
BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM.
interesting!
so it means, that you can now see caller id names between sip phones
connected to asterisk and phones connected to pbx?
Yes!
__Yehavi:
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Hello,
I am running version 1.4 with realtime support. I've set (for Snom phones
300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the
database).
- When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem
was that when such a phone received a call
I had a similar problem with Asterisk-1.4; After digging slightly I've found
that sterisk crashed (safe_asterisk restarted it and I did not notice) and this
caused the phones to lose registrations until the next expiration...
I am not sure this is your case, but try TCPDUMP of port 5060 from
anyone know if they have a solution in Cisco for:
1- Connect old PABX (with BRI or PRI) to a cisco router
2- Connect this cisco router in SIP to a Asterisk Server
I am search if cisco can this and what is the modele for this
I am using a Cisco to connect Asterisk via PRI to our
Hello,
Before I dig into the SIP traces maybe someone has a clue:
We have a few Snom phones, and the call-limit on the extension associated with
them is set to 1 (so the other Snom phones can blink the relevant LED).
When I call a SNOM phone (300 with the latest firmware) and the call is
Hello,
Anyone managed to control a Cisco voice gateway (2,811 in my case) using
MGCP? I cannot make the PRI going on-line (while with SIP I can).
If you ask why I want to use MGCP and not SIP: it is because Cisco uses
different Q.sig signalling when you manage it with different protocols,
The users do not want to push the new call softkey or the speaker button
in order to dial a call. They want to be able to just begin dialing the
number.
The users do not want to push the answer softkey after they pickup the
handset in order to answer a call.
Doesn;t it answers when you
Hello,
I am using a Cisco-2,811 as a gateway between the Asterisk PBX and our Nortel
TX-1 university's PBX. It is working but no names are exchanged. From the debug
mode I see that the Cisco sends the display name (which does not appear on the
Nortel's phones) and the Nortel does not bother to
Enclosed bellow is the fragment from extenstions.conf which does two things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi:
; regular local extensions:
; The flow is: If not available or no answer send to mailbox if
Hello,
I would like the IP phones to not accespt SIP requests (like INVITE) from any
device other than its proxy. Snom phones ignore this while Cisco Polycom
accepts the call. Any idea what to do to disable it?
Thanks! __Yehavi:
Hello,
I am looking for a real life example of using SLA lines under Asterisk.
I'll describe my environment and would like to know how I define it in
Asterisk (version 1.4 final).
Suppose I have two multi lines phones. The first phone has extension 1
assigned to it, and the second phone has
You say you can't put 'extension 3' on both phones because only one can
register to it, but that is incorrect. Extensions and SIP devices are
not the the same thing; extensions exist in the dialplan, and SIP
devices are in the SIP channel driver.
Create two SIP friends for the two phones to
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