My problem was that i didnt load pbx_config so i
guess it wasnt parsing the extensions.conf. Now i can get it to work if i
put..
[sortcalls]
switch = Realtime/@
but does not seam like a true solution, having to
create a context in teh extensions.conf for every context i want to put in
How can i let my iptables let rtp traffic through on both nic's
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Hi list,
how can I realize explicit call transfer? I want to transfer a call
which I answered to another phone and it the other one answers I want to
hang up so that my resources are freed.
Is that possible with Zaptel or which channel can I use else?
TIA, Christophorus
On 10:17, Thu 10 Aug 06, Siqhamo Sifo wrote:
How can i let my iptables let rtp traffic through on both nic's
/sbin/iptables -A INPUT -p udp --dport 11000:2 -j ACCEPT
the 11000:2 was taken from /etc/asterisk/rtp.conf
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG
Dear list,
we are using MixMonitor within our dialplans to records in/outbound calls.
Now we have the following problem:
when someone transfers a call to a different user via an attended transfer,
MixMonitor stops recording. We see this behaviour also when in the Dialplan
a call is placed to a
On Thursday, August 10, 2006 10:43 AM Christophorus Laube wrote:
Hi list,
how can I realize explicit call transfer? I want to transfer a call
which I answered to another phone and it the other one answers I want
to hang up so that my resources are freed. Is that possible with
Zaptel or
Hi all,
I just search for the Load Balance and HA solution for the Asterisk
servers. I visited http://www.vovida.org/ and there is a Load Balancer.
Did anyone try that application? If yes, please give comment about it.
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Hello,
We bought Sangoma A101 card and after some troubles successfully
compiled modules (Kernel 2.4.32, wanpipe 2.3.3-3 and zaptel 1.0.8
drivers). wanpipemon can reach w1g1 interface, but asterisk (1.2.8)
not... When I'm trying to reload chan_zap so error occurs:
christine*CLI reload chan_zap.so
Just a question:Don't you need type=user to receive in this trunk?As far as I know, peer is where you dial calls, and user is where calls can be placed.To outbound a call from you * box via SIP trunk, this trunk must be type=peer or type=friend
To inbound calls to * box via SIP trunk , this trunk
Carlos Chavez wrote:
I am trying to get my Asterisk server to talk to a Panasonic D500 PBX
using an E1 connection. The card for the Panasonic uses MFC/R2 and I
have installed Unicall. Calls from the Asterisk server to the Panasonic
go through without a hitch and I can call any
Thierry Querette wrote:
Hi Carlos,
I had the same problem and spent a lot of time studying the MFC/R2
protocol but the problem is in the libmfcr2 package version!!
Try using the packages in:
http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre7
And not in pre9.
Both pre7
Hey,
I have ALWAYS seen the first couple of seconds of audio being clipped
in my SIP calls. An easy way to see this is to call the echo test.
I don't hear the beginning of the playback.
I tried installing a wait in front of the playback, but it didn't
affect the clipped audio.
This
You want to look up REN or Ringer equivalence number .
The old PSTN phone that had a mechanical bell required a certain wattage to run
the motor.
That was later called 1 REN.
It is a measure of the electrical draw from the device when ringing.
The ringing is triggered by a voltage spike, but
Hi All,
I'm using Realtime for SIP users and I looking to find a way to be able
to authenticate users based on both the username and IP of the incoming
call the reason being I have different users connecting from same IP
but using different usernames.
I have read that setting type=peer is
Kanelbullar wrote:
Hi all,
Those of you who are using or who have used the Unicall channel for
MFC/R2 may be familiar with 'protocolvariant' field, in the
unicall.conf file. It changes from country to country and even in the
same country it may change from carrier to carrier.
I googled
username + secret
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Thursday, August 10, 2006 7:53
AMTo: asterisk-users@lists.digium.comSubject:
[asterisk-users] Realtime SIP Authentication
Hi All,I'm using Realtime for SIP users
Hi. can someone help me with where in the D-Link menu I'd go about configuring it to allow people to access my internal Asterisk server.I just want to make a call to this stanaphone number and see something happen on Asterisk.
I have a Dinky DI-524 Thanks very much.-- Anything else, let me
Hi
All
I am new member to asterisk mailing list.
I have complied the asterisk and it is running fine.
I have configured two extensions in extensions.conf
exten = 228,1,Dial
exten = 234,1,Dial
and configured the xlite soft phone. when I am calling from 234 to
228 it is unable to establish
Steve,
I have the exact same problem with a sangoma a104d so I don't think it is
related to the card. I am trying to figure that one out as well,
fortunately I only have one DID on that trunk so I used _.* to route
everything...
Sorry this doesn't help other than to let you know that it
Hi,
When I run iax2 show netstats... what is each side? Obviously Local
is me and Remote is the other side... but which is which direction?
That being is local me -- remote or is local remote -- me?
LOCAL -
REMOTE
Mr. Jones wrote:
I have had the same experience with a Grandstream order from them - 7
days and no product.
They even told me it was shipping Monday, but couldn't produce a
tracking number on Tuesday.
I ordered a T1 card from them on July 17. No trace of it. I've sent
them four e-mails,
Hello Linga,
you could download and install the ESCAUX net.PBX Free Edition and
create whatever device or call flow you want with the web interfaces.
Afterwards, in the /etc/asterisk/ directory, take a look at the
generated configuration files (gen_sip.conf, gen_extensions.conf,
Hi there
We're running Asterisk 1.2.1 (I know, it's old; we have an upgrade
planned but can't do it just yet) on Debian testing. Every now and
Asterisk and the box are dying -- no SSH login, no calls, nothing. The
last lines logged are:
Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Executing
This is an issue I'm having as well. Here is what I've discovered.
Call comes in on a T1 line. Call is sent to a SIP phone (say 4000) based on
the extensions.conf setup. User of phone 4000 has set a forward in the phone
to an external number, 1-555-555-. There is nothing telling Asterisk to
Hello
We have the same issue
asterisk[5303]: NOTICE[5303]: channel.c:1917 in ast_read: Dropping
incompatible voice frame on Local/[EMAIL PROTECTED],2 of format alaw
since our native format has changed to slin
We are using asterisk 1.2.10 + zaptel 1.2.7 + libpri 1.2.3 on a redhat 9 P4
2 Go RAM
Hey there all,
i have an app that calls our customers when the status of their
machines change. I am using a Zap channel to dial out on an analog line.
I do this by putting a drop.call file in /var/spool/asterisk/outgoing
and connecting it to an extension which fires up a python script.
the
Dana,
Thank you very much for your very detailed response. Yuo are the second one to
have recommended going with a PCI card. I think that I will take your advice
and do that instead of the SPA-3000. Even if I need two FXO ports some day,
the price of two SPA-3000 cards (and all the extra
About the ANI problem, in Brazil I use the following parameter for protocolvariant.protocolvariant=br,20,16,16I have the following configuration:Astetrisk E1 - PBXE1Telco
But Steve, after changing versions it really started to work without any modification in the .conf files. It
I walked into a new potential * install yesterday. They are
running a Samsung Prostar DCS. Does anyone have any experience with these out
there that you could relay some things to look out for when integrating this
until the migration is complete? Or what would be the best way to integrate
Hi
Sorry, I should have mentioned that we're only running SIP. Our calls
to the PSTN are routed through a VoIP carrier and all of our clients
are SIP.
Which version of Asterisk are you using? Is this killing your box? If
it is, have you established why? CPU being killed, memory starvation,
Had the same issue here as well, our company was in a
dire need for some digium cards due to an internal
system failure. So we ordered some replacements from
voiplink for overnight delivery, we loved the pricing
they had on the website and even called to confirm
they had stock and could ship that
What about using a *72 application to forward the calls rather than
the phone itself.
on Thursday 08/10/2006 M D([EMAIL PROTECTED]) wrote
Hi
Sorry, I should have mentioned that we're only running SIP. Our calls
to the PSTN are routed through a VoIP carrier and all of our clients
are
Is there a command for setting of a DID number?
Eg below I can set callerid
[custom-fromiaxfwd]
exten = s,1,Set(CALLERID(number)=2125316214)
Butw what I would prefer to do is set DID like this
(it doesnt work
[custom-fromiaxfwd]
exten =
I have always had excellent service from Atacomm. Not always the absolute lowest price, but they pretty much ship when they say they will.On 8/10/06, Don Tasker
[EMAIL PROTECTED] wrote:Had the same issue here as well, our company was in a
dire need for some digium cards due to an internalsystem
Here is their contact info:
http://www.voiplink.com/Contact_Us_Voiplink_com_s/11.htm
Here is the CA department of Consumer affairs website:
http://www.dca.ca.gov/
I recommend that you all work together on this, all filing complaints
with the dca of CA.
Please don't forget to report back on the
At 07:59 AM 8/10/2006, you wrote:
is there a way that asterisk can detect when someone speaks ? Like
answering a phone? i dont need speech recognition or anything like
that, just something that lets me know that any sound is originating
from the other end.
Play a recording that says Press 1
A venture capital funded private company buying a public company which makes
the same products?! I seriously doubt that will happen!
I can thing of a LOT of companies that might want to buy Sangoma some day
but Digium is not one of them. Besides, from the perspective of Asterisk,
having another
shadowym wrote:
A venture capital funded private company buying a public company which makes
the same products?! I seriously doubt that will happen!
I can thing of a LOT of companies that might want to buy Sangoma some day
but Digium is not one of them. Besides, from the perspective of
Curt Shaffer wrote:
I walked into a new potential * install yesterday. They are running a
Samsung Prostar DCS. Does anyone have any experience with these out
there that you could relay some things to look out for when integrating
this until the migration is complete? Or what would be the best
Ira wrote:
At 07:59 AM 8/10/2006, you wrote:
is there a way that asterisk can detect when someone speaks ? Like
answering a phone? i dont need speech recognition or anything like
that, just something that lets me know that any sound is originating
from the other end.
Play a recording that
Dean Collins wrote:
Is there a command for setting of a DID number?
Eg below I can set callerid
[custom-fromiaxfwd]
exten = s,1,Set(CALLERID(number)=2125316214)
Butw what I would prefer to do is set DID -like this (it doesn't work
[custom-fromiaxfwd]
exten =
My Asterisk is 1.2.9.1 but I've recreated this on 1.2.7 and 1.2.8. Not tried
1.2.10 yet.
This only happens on forwarded calls for me as well. I've not let it run too
long to see if the server dies eventually. I don't believe it will because
once the caller hangs up the errors stop and my server
Kannel is good software that is able to handle this (http://www.kannel.org).
Though if this will be your only SMS service that you provide, most SMS
gateways offer to implement a http get/post interface to send/receive
messages.
An example is http://www.clickatell.com/ but if you google around a
Hello,
I am using A2Billing which comes with TrixBox 1.1.1 . I am creating SIP
account from A2Billing for IP Phone. Everything is working fine. What I need
is to assign Voicemail box to every phone, which I think cannot be done
through A2Billing right now. Therefore I need to know some utility or
Can you do buddies with Cisco phones running SIP? I can't find anything
that says yes or no. I can set it up on the * server, but I don't know
what to do on the 7960's themselves.
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If you're using Trixbox, why not use the interface that comes with it, FreePBX? System Administration - FreePBX - Setup - Extensions.AlexOn 8/10/06,
Wasif [EMAIL PROTECTED] wrote:
Hello,I am using A2Billing which comes with TrixBox 1.1.1 . I am creating SIPaccount from A2Billing for IP Phone.
I have just purchased an IAXy and cannot get an analog telephone to work
with it. The provisioning is succesful and the asterisk server sees it
and you can call it.
Symptoms:
- network status light is on solid
- telephone status light flashes every 7 or 8 seconds
- no dialtone and no response
Hi Eric,
No I know what I want. I want to set the DID to be 212-531-6214 as my
current provider doesn't send a DID number.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Thursday, 10
http://snakesonaplane.varitalk.com
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2006/8/10, Peder @ NetworkOblivion [EMAIL PROTECTED]:
Can you do buddies with Cisco phones running SIP? I can't find anything
that says yes or no. I can set it up on the * server, but I don't know
what to do on the 7960's themselves.
What about a google look for asterisk cisco 7960 config
Any one know how can I use asterisk to record phone call i.e situration like
this
T1 - channels bank - 24 lines - PBX - Recording using asterisk -
24 phones.
Sam
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well, i dont know about how to get a digital line out of the office,
but that would be cool. Clean up some other stuff too. Ah-well, guess i
will settle for door number 1. The press 1 to continue loop.
thanks, guys, let you know how it turns out.
shawnOn 8/10/06, Eric ManxPower Wieling [EMAIL
Then you would use a Goto(custom-fromiaxfwd,2125316214,1) instead of the
Set(CALLERID
In Asterisk there is no difference between a DID and an extension.
Dean Collins wrote:
Hi Eric,
No I know what I want. I want to set the DID to be 212-531-6214 as my
current provider doesn't send a DID
Did it about 10x to friends. Pretty funny.On 8/10/06, Hugh L. Johnson [EMAIL PROTECTED] wrote:
http://snakesonaplane.varitalk.com
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Handle incoming calls to s extension and in the next priority set EXTEN var to your DID then make a goto to desired context.Hope it helps,MoutaPTOn 8/10/06,
Dean Collins [EMAIL PROTECTED] wrote:
Hi Eric,No I know what I want. I want to set the DID to be 212-531-6214 as mycurrent provider doesn't
Hi Mouta, sorrycan you elaborate a little
(maybe something a little more basic).
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta
Sent: Thursday, 10 August 2006
2:16 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Eric can you write that out with a little more explanation as I'm not
sure where that fits.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Thursday, 10 August 2006 2:10 PM
To:
SPONSORED THIS MONTH BY: SOUND CHOICE COMMUNICATIONS LLC Keep in touch
with the World(TM)
With Support from ONVOY Be seen. Be heard. Be informed. Be
connected(TM)
Hello,
The next meeting is this Saturday, and 11:30am at Onvoy in the west metro.
Onvoy is located at Hwy169 and I-394 in the
It would be great if some swift asterisk coder would write a little application
that could monitor the channel for call progress.
so on cases of busy, ring (with timeout) and TALK detection an appropriate response
could be taken for these analog systems.
I would think with all the code in
here here !
skOn 8/10/06, Jerry Geis [EMAIL PROTECTED] wrote:
It would be great if some swift asterisk coder would write a little applicationthat could monitor the channel for call progress.so on cases of busy, ring (with timeout) and TALK detection an appropriate response
could be taken for these
Was going to say I've never had that issue.. but we use PRI.
On 8/10/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Ira wrote:
At 07:59 AM 8/10/2006, you wrote:
is there a way that asterisk can detect when someone speaks ? Like
answering a phone? i dont need speech recognition or
I read both of those links and I don't see any mention of SIP buddies on
either one.
Adrià Vidal wrote:
2006/8/10, Peder @ NetworkOblivion [EMAIL PROTECTED]:
Can you do buddies with Cisco phones running SIP? I can't find anything
that says yes or no. I can set it up on the * server, but
Then in the tftp config file for the phone add speeddials
for the 6000 extension (cant recal how it is done, there are
examples in the default file and on the wiki)
I found out you really have to define the speeddials in the
tftp files. speeddials configured with the phone menu or
webinterface
Hello, I was looking for some FXS gateway with 48
ports.
I find some products in different webs but prices
are between 3.000$ and 4.000$. Someone that have search about that kind of
hardware find some website with great prices?.
Many thanks.
___
Asteris 1.2.7
Redhat 9
Hi,
Is this the correct syntax for setting CALLERID:
exten = _1,n,Set(CALLERID(all)=The Smiths 212-876-9345)
I'm able to get the number to change but the name is always Unknown
Name. I've tried numerous combinations of quotes, but just cannot
get the name...
Thanks,
H
Xorcom has good produtcs for that:For 48 FXO you can combine
Astribank-32 with Astribank-16 and they connect to the Asterisk server via a normal USB-2 cable.Take a look at:http://www.xorcom.com/products.html
ThierryOn 8/10/06, Javier Matos Odut [EMAIL PROTECTED] wrote:
Hello, I was
hugolivude wrote:
I'm able to get the number to change but the name is always Unknown
Name. I've tried numerous combinations of quotes, but just cannot
get the name...
I use Caller Name401
Note, no space between the closing and the character. Seems to work
for me and Polycom phones.
One way to do this would be some rhino channles banks
http://shop.resv.net/Shops/ViewItem.aspx/27934028032-45519589888.htm
you could end up with 48 prots for $2,390-ish. That's
fairly cheap way to do it.
Hello, I was looking for some FXS gateway with 48
Are you trying this for PSTN? it wont work, you can't change the name
on PSTN, the last CO looks up the name for the number it gets.
On 8/10/06, hugolivude [EMAIL PROTECTED] wrote:
Asteris 1.2.7
Redhat 9
Hi,
Is this the correct syntax for setting CALLERID:
exten = _1,n,Set(CALLERID(all)=The
dunno that there is an all datatype is there?
Set(CALLERID(name)=John Doe)
Set(CALLERID(number)=555121212)
- Original Message -
From: hugolivude [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 10,
Why is it Don that you don't know? look it up, it's in the CLI:
pbx*CLI show function CALLERID
pbx*CLI
-= Info about function 'CALLERID' =-
[Syntax]
CALLERID(datatype)
[Synopsis]
Gets or sets Caller*ID data on the channel.
[Description]
Gets or sets Caller*ID data on the channel. The
Don,Found this in the developer docs:Set(CALLERID(all)=...) or Set(CALLERID(ani)=...)
http://www.asterisk.org/doxygen/app__setcallerid_8c.htmlAll should work... But I'd also ask the same question as C F: Where are you trying to set CID for? Are you trying to set it on calls that come into your
I think Don's point was that he wasn't sure that all existed. That being the case, if the OP was using all, it wouldn't work and he wouldn't get the desired result. It's hardly lying to say that one is unsure of something.
AlexOn 8/10/06, C F [EMAIL PROTECTED] wrote:
Why is it Don that you don't
The point was not if he was lying or not, but the fact that he *wasn't
sure as you put it. If you are not sure then look it up. He was even
told by the OP that it existed.
On 8/10/06, Alex Robar [EMAIL PROTECTED] wrote:
I think Don's point was that he wasn't sure that all existed. That being
O.K looks like you are talking about presence, take a look about the hint app.
But didn't now how to make the Cisco check the hint... get us informed
about you advance.
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If you want it to be installed in asterisk, you can use an Asterisk
application that listens for the incoming SMS requests.
Otherwise, I suggest you better use a cgi/php/asp script or whatever flavour
web scripts you like to be using and make an Asterisk manager request OR
use .call files.
An
Thanks everyone. I appreciate the warning about POTS. I already
established in another thread that I won't be able to do that. I'm
now using SIP when I need to set CALLERID. As I mentioned I can set
the number to anything I want and it works, so the Set(CALLERID(all)
is having some effect,
Maybe cause I was trying to help someone out AND I DIDN'T care if it
existed...Cause if you look back I wasn't the original author of this
question...thus I was showing one way to do it...I had no reason to go look
it up cause it wasn't my problem...wtf...
- Original Message -
From:
Thank you...someone understood...hehe
- Original Message -
From:
Alex
Robar
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, August 10, 2006 6:21
PM
Subject: Re: [asterisk-users] Correct
syntax for Set(CALLERID(all)...
I think
Sometime's I wonder if some of these people read
before they reply...hehe
Thanks for clarifying it for them...I was out to
dinner with my wife...came back...and it was like I asked the question and I was
the idiot for not looking.
- Original Message -
From:
Alex
Robar
ok maybe I can explain my problem better. There two trunks both have the same
details except one is type=peer (and only does ulaw) and the other is type
friend (and does ulaw/alaw/g729). Incoming calls should be only going into
the type=friend trunk, NOT into the type=peer trunk. Both should be
Cheers Don. I appreciate the fact that you took the time to read my
post and offer some help. Helping out is what it's all about IMO. I
hope that you will continue to do so...
Yours,
H
On 8/10/06, Don [EMAIL PROTECTED] wrote:
Sometime's I wonder if some of these people read before they
If you ITSP is sending the call to the PSTN (please note we are NOT
talking about POTS, but PSTN), then it wont work either.
On 8/10/06, hugolivude [EMAIL PROTECTED] wrote:
Thanks everyone. I appreciate the warning about POTS. I already
established in another thread that I won't be able to do
Shaun Hofer wrote:
ok maybe I can explain my problem better. There two trunks both have the same
details except one is type=peer (and only does ulaw) and the other is type
friend (and does ulaw/alaw/g729). Incoming calls should be only going into
the type=friend trunk, NOT into the type=peer
I have spent the best part of half the morning googling a solution to
this but nothing has jumped out at me.
Is there a simple method of allowing dynamic changes to the extensions
via a web interface without having to go the @home method.
All I want is to make a webpage to select which person
Hello All,I would like to know is there any database where i can lookup by providing a phone number whether it can be terminated via SIP or Mobile or PSTN???I know there is enum lookup available which will only terminate if the number is registered with the particular enum provider or
e164.org.
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