Re: [asterisk-users] Asterisk version survey

As I said before. I would like to see a much more comprehensive survey complete with names or nicknames if supplied. Asterisk version, in-house patches or programming, OS, preferred hardware (server, phones, cards) Then more bio on the respondent. Years in data and also telephony field,

Re: [asterisk-users] Asterisk version survey

Well, $9 would pay for up to 500 answers. I also found a free one I'm looking at now, but you never get anything really good free :) > If $9 can put that survey together in a comprehensible set of questions > and results, I will pay the $9. Let me see if I can put what you ask for together on the

Re: [asterisk-users] Asterisk version survey

NEW Asterisk favorite OS survey: http://www.esurveyspro.com/Survey.aspx?id=1f38482e-c3fa-4384-8b8a-65a9f40b2cd8 The above is a compromise between three elements: 1) The amount of time and patience I have 2) The information I think most people are willing to proovide 3) Steve's requests Please ha

Re: [asterisk-users] Asterisk version survey

randulo wrote: > Well, $9 would pay for up to 500 answers. I also found a free one I'm > looking at now, but you never get anything really good free :) > >> If $9 can put that survey together in a comprehensible set of questions >> and results, I will pay the $9. > > Let me see if I can put what

Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4

Julio, Thanks for your suggestion, at this stage I would like this version of g729 running in my box.. but,,, is good to know the paid version works without any problems in this machine for the next stage. Best Regards, Fernando Julio Arruda wrote: Fernando Berretta wrote: Dear Mindauga

[asterisk-users] Asterisk behind a PIX firewall?

Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 1-2 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 1-2.The phone reg

Re: [asterisk-users] Asterisk version survey

y the way, CentOS is winning at the moment. I don't see how to publish the results live yet, but I'll do a static report as soon as there are more. On Nov 27, 2007 2:25 PM, randulo <[EMAIL PROTECTED]> wrote: > On Nov 27, 2007 2:06 PM, Darrick Hartman (lists) > <[EMAIL PROTECTED]> wrote: > > Can yo

Re: [asterisk-users] Asterisk version survey

On Nov 27, 2007 2:06 PM, Darrick Hartman (lists) <[EMAIL PROTECTED]> wrote: > Can you just install limesurvey on a server some place? It would allow > you to do however many future surveys you want to do. I'll look into it for my own use, thanks. Wufoo is way ahead of most of these, but it ain't

Re: [asterisk-users] Asterisk behind a PIX firewall?

Matt wrote: > Is there anything special that anyone here has had to do to get an > Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? > > Ports 1-2 UDP are open on the PIX and forwarding to the Asterisk > server. The Asterisk server's RTP.CONF is set to use 1

Re: [asterisk-users] Asterisk behind a PIX firewall?

> Just checking NAT=yes, canreinvite=no ? Correct, I have those settings set for this phone. Asterisk has been reloaded even restarted. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSC

[asterisk-users] Sip to ATA?

Currently running two POTS lines into an asterisk system. Analog and SIP on premises. Being in the sticks, the POTS service is abysmal for quality, especially in the rain. Recently, cable has become available with VOIP phone. The cost savings are attractive as it can replace several indepen

[asterisk-users] SIP port 5060 closed - how do I open it?

Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and

Re: [asterisk-users] Asterisk behind a PIX firewall?

Matt wrote: > > > > Just checking NAT=yes, canreinvite=no ? > > > Correct, I have those settings set for this phone. Asterisk has been > reloaded even restarted. > > Is this a dual NAT situation? NAT on the phone side and NAT at the PIX? If so, I fear it will never work, yo

[asterisk-users] Asterisk API Manager

Hi, Does Asterisk manager allow multiple clients to connect to an Asterisk instance using the same user account ? Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update opt

Re: [asterisk-users] Asterisk behind a PIX firewall?

Is this a dual NAT situation? NAT on the phone side and NAT at the PIX? > If so, I fear it will never work, you might get one way audio though. > > I live OpenVPN bridges for double NAT situations, of course you could > try IAX2 but I have seen too many sound quality issues surrounding IAX2 > so

Re: [asterisk-users] Asterisk behind a PIX firewall?

Matt wrote: > > > Is this a dual NAT situation? NAT on the phone side and NAT at the > PIX? > If so, I fear it will never work, you might get one way audio though. > > I live OpenVPN bridges for double NAT situations, of course you could > try IAX2 but I have seen too many

Re: [asterisk-users] Asterisk behind a PIX firewall?

> > This is a dual NAT situation. PIX on Asterisk side, and Netgear on phone > side. HOWEVER.The Asterisk box has it's own IP but it is being > tunneled through the PIX.I guess the PIX must be messing something up? > If I remove the phone from behind the Netgear... then I get the

Re: [asterisk-users] Asterisk behind a PIX firewall?

On Nov 27, 2007 9:59 AM, Matt <[EMAIL PROTECTED]> wrote: > This is a dual NAT situation. PIX on Asterisk side, and Netgear on phone > > side. HOWEVER.The Asterisk box has it's own IP but it is being > > tunneled through the PIX.I guess the PIX must be messing something up? > > > > >

Re: [asterisk-users] Asterisk behind a PIX firewall?

Matt wrote: > This is a dual NAT situation. PIX on Asterisk side, and Netgear on > phone side. HOWEVER.The Asterisk box has it's own IP but it > is being tunneled through the PIX.I guess the PIX must be > messing something up? > > > > If I remove the phone from be

Re: [asterisk-users] Asterisk behind a PIX firewall?

> > It is being tunneled or forwarded? Does the Asterisk box have a public > IP or does the PIX have the public which just forwards to the private? > > If it is just forwarding, it will never work without either putting one > side on a public IP, using a VPN solution, or IAX2. > It IS being forwa

Re: [asterisk-users] Asterisk behind a PIX firewall?

Matt, If your phone is using SIP, then you should enable sip inspection (7.x code or above) or fixup sip (6.x code) and have a rule that allows source (wherever you need) inbound on the outside interface to TCP 5060 (SIP port). The sip inspection or fixup should enable the proper ports for the re

Re: [asterisk-users] Asterisk behind a PIX firewall?

> > Trust me on this, I have tried almost everything to get it to work, the > best you can hope for is one way audio in a dual NAT. > > The answer has to do with where the packets are sent from and where they > seem to be sent from. > > If you are not familiar with OpenVPN, you should check it out.

Re: [asterisk-users] SIP port 5060 closed - how do I open it?

Zaheer, If a "netstat -an|grep -I LISTENING" shows that a LISTENING port for 5060 is there, then the problem isn't Asterisk, but some firewall system on the server is blocking access from outside. If its not there, then come back to the group.. Adrian Marsh ___

Re: [asterisk-users] SIP port 5060 closed - how do I open it?

Correction: netstat -an|grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* Adrian Marsh From: Adrian Marsh Sent: 27 November 2007 15:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] SIP p

Re: [asterisk-users] Asterisk behind a PIX firewall?

Try to just open port 5060 for SIP signaling on the PIX and also enable the INSPECT SIP rule. That way, your PIX firewall will inspect SIP signalling and open the necessary UDP ports for the RTP. If you have NAT uptream in the network, you should see if in the layer 4 the IPs shown in the SIP mess

[asterisk-users] Attended transfer to Queue

Hi, I will confess immediately that this is only tested on 1.2.24, and I would be interested to know if it happens on 1.4, but I cannot find a bug-tracker entry which represents this issue. Consider a PSTN call which comes into asterisk, and is bridged to a SIP phone. The phone operator then plac

[asterisk-users] Finding the status of an 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...

Re: [asterisk-users] Asterisk API Manager

Yes, but you should use astmanproxy instead and don't bother Asterisk with multiple manager connections. On Nov 27, 2007 8:24 AM, Anthony Chapellier <[EMAIL PROTECTED]> wrote: > Hi, > > Does Asterisk manager allow multiple clients to connect to an Asterisk > instance using the same user account ?

Re: [asterisk-users] Voice mail & Uniden UIP-200 phones

Yep, that fixed it. Just shaking my head as to why the behavior changed... Lyle CunningPike wrote: > Try dtmfmode=inband > > CP > > Lyle Giese wrote: > >> I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 >> with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of

Re: [asterisk-users] Asterisk behind a PIX firewall?

Shlomo, My understanding is I have to do a no fixup sip 5060. This from Cisco. Without doing the no fixup the registration ports get all mangled. On Nov 27, 2007 10:11 AM, Shlomo Dubrowin <[EMAIL PROTECTED]> wrote: > Matt, > > If your phone is using SIP, then you should enable sip inspection (7.

Re: [asterisk-users] Asterisk behind a PIX firewall?

You can also create the vpn using the existing pix and netgear, eliminating more hardware and points of failure. - Original Message - From: "Ricardo Carvalho" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, November 27, 2007 7:30:35 A

Re: [asterisk-users] SIP port 5060 closed - how do I open it?

Hi Adrian here is what I got when I ran the command you suggested: netstat -an|grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* I also ran netstat -l and got the following: Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address

Re: [asterisk-users] Asterisk behind a PIX firewall?

On Nov 27, 2007 9:08 AM, Steve Totaro <[EMAIL PROTECTED]> wrote: > > Matt wrote: > > > > > > > > Just checking NAT=yes, canreinvite=no ? > > > > > > Correct, I have those settings set for this phone. Asterisk has been > > reloaded even restarted. > > > > > > Is this a dual NAT situation?

Re: [asterisk-users] Asterisk behind a PIX firewall?

Enabling the fixup breaks the registration. On Nov 27, 2007 10:30 AM, Ricardo Carvalho <[EMAIL PROTECTED]> wrote: > Try to just open port 5060 for SIP signaling on the PIX and also enable > the INSPECT SIP rule. That way, your PIX firewall will inspect SIP > signalling and open the necessary UDP

Re: [asterisk-users] Finding the status of an extension

Yehavi Bourvine +972-8-9489444 wrote: > 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

[asterisk-users] Snom phones, blinking lights and call pickup

Hi! I have the following questions/problems with * 1.4. We have several Snom phones (320 and 360). Hints are configured in extensions.conf (core show hints shows the correct values). My Snom phone is registered to some numbers (validated by using sip show subscriptions). I see the lights blink

[asterisk-users] Restricting the manager interface to a number?

Hi! Some persons are using a TAPI driver to connect via the manager interface to the asterisk (1.4). While I can give every user his own password, I didn’t find a way to restrict a user to a certain phone number, so that he can only dial with his number via the TAPI driver and can only answer

Re: [asterisk-users] SIP port 5060 closed - how do I open it?

Hi Adrian here is what I got when I ran the command you suggested: netstat -an|grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* I also ran netstat -l and got the following: Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address

Re: [asterisk-users] Snom phones, blinking lights and call pickup

Stephan Seitz wrote: > We have several Snom phones (320 and 360). Hints are configured in > extensions.conf (core show hints shows the correct values). My Snom phone > is registered to some numbers (validated by using sip show > subscriptions). I see the lights blinking if someone calls the sub

Re: [asterisk-users] Asterisk behind a PIX firewall?

C F wrote: > On Nov 27, 2007 9:08 AM, Steve Totaro <[EMAIL PROTECTED]> wrote: >> Matt wrote: >>> >>> >>> Just checking NAT=yes, canreinvite=no ? >>> >>> >>> Correct, I have those settings set for this phone. Asterisk has been >>> reloaded even restarted. >>> >>> >> Is this a dual NAT situ

Re: [asterisk-users] Asterisk behind a PIX firewall?

On Tue, 27 Nov 2007, Matt wrote: > Shlomo, > My understanding is I have to do a no fixup sip 5060. This from Cisco. > Without doing the no fixup the registration ports get all mangled. So yet another router with a broken SIP ALG... (Juniper NetScreen is one I had issues with) Gordon _

Re: [asterisk-users] Asterisk behind a PIX firewall?

Steve Totaro wrote: > C F wrote: >> On Nov 27, 2007 9:08 AM, Steve Totaro <[EMAIL PROTECTED]> wrote: >>> Matt wrote: Just checking NAT=yes, canreinvite=no ? Correct, I have those settings set for this phone. Asterisk has been reloaded even restarted. >>>

Re: [asterisk-users] Snom phones, blinking lights and call pickup

On 11/27/07, Stephan Seitz <[EMAIL PROTECTED]> wrote: [snip] > So far, so good. What I want now is that I can see the number of the > caller in my display and can pickup the call by pressing the blinking > lights. I can pickup the call using the pickup extension. Hi, I would first ask yourself

Re: [asterisk-users] hostname in MySQL CDR records

On Thursday 01 November 2007 12:57:10 am Steve Edwards wrote: > On Wed, 31 Oct 2007, Jim Gottlieb wrote: > > I would like to send the CDR records from all our machines around the > > world to a single database. But I need the hostname included with each > > record for monitoring purposes. > > > >

Re: [asterisk-users] Asterisk behind a PIX firewall?

On Nov 27, 2007 11:02 AM, C F <[EMAIL PROTECTED]> wrote: > On Nov 27, 2007 9:08 AM, Steve Totaro <[EMAIL PROTECTED]> > wrote: > > > > Matt wrote: > > > > > > > > > > > > Just checking NAT=yes, canreinvite=no ? > > > > > > > > > Correct, I have those settings set for this phone. Asterisk

Re: [asterisk-users] Finding the status of an extension

Hi, Here is a small AGI script that get you the hint status of the extension simply call AGI(script.agi,SIP/100) !/usr/bin/perl # # page.agi - Original file was allpage.agi by Rob Thomas 2005. # With parts of allcall.agi Original file by John Baker # Modified by Ada

Re: [asterisk-users] Restricting the manager interface to a number?

Stephan Seitz wrote: > Some persons are using a TAPI driver to connect via the manager interface > to the asterisk (1.4). While I can give every user his own password, > I didn’t find a way to restrict a user to a certain phone number, so that > he can only dial with his number via the TAPI dri

Re: [asterisk-users] Snom phones, blinking lights and call pickup

Steve Davies wrote: > On 11/27/07, Stephan Seitz <[EMAIL PROTECTED]> wrote: > > [snip] > >> So far, so good. What I want now is that I can see the number of the >> caller in my display and can pickup the call by pressing the blinking >> lights. I can pickup the call using the pickup extension. >

Re: [asterisk-users] Finding the status of an extension

If you have some SIP phone BLF feature capable, you can try it. With it in the phone you can view the state of those the registered extensions you like, as well with it if you do "sip show subscriptions" in the asterisk CLI, you'll get the list of extensions and the last state of those monitored ex

Re: [asterisk-users] Asterisk API Manager

However I wanted to get periodic infos about queued users (position in queue) only. So I thought I could make a program sending periodic requests to asterisk manager. Is it really bad to bother asterisk manager with frequently and periodic requests sent by mutiple users (we could say maybe 100

Re: [asterisk-users] Snom phones, blinking lights and call pickup

On 11/27/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote: > > All he's asking for are SIP NOTIFYs of the extension state with > dialog-info XML in the body as described in RFC 4235 and elsewhere. > > It's up to the SIP user agent (i.e. the phone) how to deal with it. > (e.g. display only the first or

Re: [asterisk-users] Asterisk behind a PIX firewall?

Matt wrote: > > > On Nov 27, 2007 11:02 AM, C F <[EMAIL PROTECTED] > > wrote: > > On Nov 27, 2007 9:08 AM, Steve Totaro > <[EMAIL PROTECTED] > > wrote: > > > > Matt wrote: > > > > > > > > > > > > J

Re: [asterisk-users] Asterisk behind a PIX firewall?

> > This is good to hear. Now I know it can be done this way, although I > still prefer OpenVPN for it's security and ability to let you do other > things such as AMI or whatever. > > It is kind of hard to portscan 5060 when it is not open. I bet I could > do a portscan on 5060 and of those hits

Re: [asterisk-users] Sip to ATA?

Yes, you would just plug the FXO ports directly into the cable modem and it would work. In my experience the cable media gateways (Arris and Motorola) are more robust than VoIP ATAs. They can provide polarity reversal on disconnect, can power a higher REN, and even have built-in batteries to k

Re: [asterisk-users] Asterisk version survey

At 05:26 AM 11/27/2007, you wrote: >By the way, CentOS is winning at the moment. I don't see how to publish >the results live yet, but I'll do a static report as soon as there are >more. Not much of a surprise, TrixBox and it's predecessors all installed CentOS by default and all the people like

Re: [asterisk-users] Sip to ATA?

At 06:01 AM 11/27/2007, you wrote: >I am hesitant to believe that I can simply plug my TDM400P >(2fxo/2fxs) into these (ATA ?) jacks and call it good. > >Any insight? Am I better off ignoring their phone offering and >setting myself up with an IAX or SIP provider? (and surplus-ing the >card).

Re: [asterisk-users] hostname in MySQL CDR records

You can also use CDR(userfield) parameter and that way you can write in the column userfield of your CDR table of the DB, the hostname of the asterisk for each call. You can try something like the following in the dialplan of each machine: Set(CDR(userfield)=hostname_of_the_machine) Regards, Rica

Re: [asterisk-users] Asterisk behind a PIX firewall?

Matt wrote: > This is good to hear. Now I know it can be done this way, although I > still prefer OpenVPN for it's security and ability to let you do other > things such as AMI or whatever. > > It is kind of hard to portscan 5060 when it is not open. I bet I could > do a port

[asterisk-users] Urgent question.

Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't

Re: [asterisk-users] Urgent question.

In other words, what I need is a way for the upstream switch to somehow think that the B channels are out of service, but without actually taking the B channels out of service and dropping the existing calls. >From within asterisk, zaptel, wanpipe, whatever. Is that possible? On Tue, 27 Nov 20

Re: [asterisk-users] Asterisk behind a PIX firewall?

> > > > No, then you are good, but I would bet my life that there are a good > many systems that use the extension for both password and username and > can be accessed from the net. > O yeah.. I can imagine.. wonder how many open systems are out there :) ___

Re: [asterisk-users] Sip to ATA?

>>> On 11/27/2007 at 12:26 PM, Ira <[EMAIL PROTECTED]> wrote: > At 06:01 AM 11/27/2007, you wrote: > >>I am hesitant to believe that I can simply plug my TDM400P >>(2fxo/2fxs) into these (ATA ?) jacks and call it good. >> >>Any insight? Am I better off ignoring their phone offering and >>settin

Re: [asterisk-users] Asterisk behind a PIX firewall?

Steve Totaro wrote: > Matt wrote: >> This is a dual NAT situation. PIX on Asterisk side, and Netgear on >> phone side. HOWEVER.The Asterisk box has it's own IP but it >> is being tunneled through the PIX.I guess the PIX must be >> messing something up? >> >> >> >> If

Re: [asterisk-users] Urgent question.

Alex Balashov wrote: > Our provider gives us four PRIs as a trunk group hunt group. Meaning, the > provider's switch will cycle through B channels in span 1, 2, 3, ... until > it finds one that is available. > > I have moved spans 2-4 onto another machine. But we have one remaining > box with

[asterisk-users] ResetCDR Options v, a - Asterisk 1.4

"show application ResetCDR" shows 3 option values which may be used. "w" works as expected, but what about "v" and "a"? "v -- Save CDR variables" - but theese are saved anyway "a -- Store any stacked records" - but what are "stacked records"? I found some ResetCDR examples using option "w" or

Re: [asterisk-users] Urgent question.

Well, right; the problem is that this also drops the existing calls in progress. :-) On Tue, 27 Nov 2007, Philipp Kempgen wrote: Alex Balashov wrote: Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3,

Re: [asterisk-users] Urgent question.

Philipp Kempgen wrote: > I think if you just pull the plug the PRI goes to red alarm and > the provider will try one of your other PRIs for new incoming > calls. Of course I don't know how your provider will handle this > so don't trust me. :) The downside of this is that established calls would

Re: [asterisk-users] hostname in MySQL CDR records

On Tuesday 27 November 2007 11:50:09 am Ricardo Carvalho wrote: > You can also use CDR(userfield) parameter and that way you can write in the > column userfield of your CDR table of the DB, the hostname of the asterisk > for each call. You can try something like the following in the dialplan of > e

Re: [asterisk-users] Urgent question.

You should be able to issue a "stop gracefully" command to asterisk. That'll cause it to stop accepting new calls, but will let existing calls continue until complete. -erik On Nov 27, 2007 12:06 PM, Alex Balashov <[EMAIL PROTECTED]> wrote: > > In other words, what I need is a way for the upstrea

[asterisk-users] Copy or Make + Make Install

Hi List; If I have a running Asterisk on one machine and I need to have another Asterisk on another machine, can I copy the files from the first running Asterisk machine to the new machine or I have to do the ./configure + make + make install? If I can copy, then which directories (and files) ne

Re: [asterisk-users] Urgent question.

Generally, best practices would dictate that you do this while the business is closed if it is not a 24/7 operation. I guess that kind of foresight comes with experience. Glad to see things are going smoothly. Thanks, Steve Erik Anderson wrote: > You should be able to issue a "stop graceful

[asterisk-users] Semi-OT Part 2: Videophone

Following the recommendations here I've ordered a couple different Polycom & Aastra phones to play with speaker phones. One of our next big projects is Video. I know Grandstream has a video phone, has anyone used it. Anyone have any other recommendations? Thanks, Ken (Who awaits the vmukti.com

Re: [asterisk-users] Urgent question.

To answer the question, there is currently no way to busy out a channel except to put it in use. There was some discussion about adding this feature at Astricon and on the list fairly recently. Thanks, Steve Steve Totaro wrote: > Generally, best practices would dictate that you do this while t

Re: [asterisk-users] hostname in MySQL CDR records

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anthony Messina wrote: > with a "core show function CDR", it states: > > All of the above variables are read-only, except for accountcode, > userfield, and amaflags. You may, however, supply > a name not on the above list, and create your own >

Re: [asterisk-users] Asterisk version survey

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 randulo wrote: > NEW Asterisk favorite OS survey: > > http://www.esurveyspro.com/Survey.aspx?id=1f38482e-c3fa-4384-8b8a-65a9f40b2cd8 > > The above is a compromise between three elements: > 1) The amount of time and patience I have > 2) The informatio

Re: [asterisk-users] Asterisk API Manager

So you'd be making 100 connections/minute, which is pretty relentless. That's like five connections, five requests sent, five responses received, and five disconnects, /every/ three seconds. And the likelihood of all 100 users to be spread out evenly over a minute doesn't seem very high. I th

Re: [asterisk-users] Semi-OT Part 2: Videophone

We've used the Grandstream video phone quite a bit, and I have to say, I'm considerably impressed with its quality. YES, it's a Grandstream (and has the usual quirks and annoyances that one has come to expect now and again), but the quality of the screen and camera are both excellent, and with

Re: [asterisk-users] Asterisk version survey

Results should be withheld until the end of the specified polling period. Thanks, Steve Totaro 888.777.1888 Matt Riddell wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > randulo wrote: > >> NEW Asterisk favorite OS survey: >> >> http://www.esurveyspro.com/Survey.aspx?id=1f38482e-c3

Re: [asterisk-users] Asterisk version survey

On Nov 27, 2007 8:13 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: > Where are the results? Hey Matt, As I said above, this site doesn't appear to allow public results page. (Actually I said I couldn't find one) I'll show the static one as soon as I can. Wait, this is the Internet. I can go now. O

[asterisk-users] Zaptel 1.2.22 and 1.4.7 released

The Asterisk.org development team has announced the release of Zaptel versions 1.2.22 and 1.4.7. These releases contain (among other things) many bug fixes to the TC400B driver, a bug fix on the wctdm24xxp driver for users with a VPM150M, as well as numerous improvements and fixes to the Xorcom

Re: [asterisk-users] Asterisk version survey

randulo wrote: > On Nov 27, 2007 8:13 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: >> Where are the results? > > Hey Matt, > > As I said above, this site doesn't appear to allow public results > page. (Actually I said I couldn't find one) > I'll show the static one as soon as I can. > > Wait, thi

Re: [asterisk-users] Copy or Make + Make Install

Hello, Only copy the configuration files, extensions.conf, sip.conf, iax.conf , Best regards On Nov 27, 2007 1:27 PM, bilal ghayyad <[EMAIL PROTECTED]> wrote: > Hi List; > > If I have a running Asterisk on one machine and I need > to have another Asterisk on another machine, can I > c

Re: [asterisk-users] Asterisk version survey

On Nov 27, 2007 8:17 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > Results should be withheld until the end of the specified polling period. Gh! Ok, I'll take 'em off. We can give this a few days then , and when things stop, I'll publish. Open to suggestions. As far as the open source p

Re: [asterisk-users] SIP port 5060 closed - how do I open it?

So can anyone tell me why the * server is not listening on UDP port 5060? Thanks, Zaheer _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer K. Master Sent: Tuesday, November 27, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subjec

Re: [asterisk-users] SIP port 5060 closed - how do I open it?

Does a sip.conf exist? Asterisk won't start the SIP UAS if it is missing. On Tue, 27 Nov 2007, Zaheer K. Master wrote: > So can anyone tell me why the * server is not listening on UDP port 5060? > > > > Thanks, > > Zaheer > > > > _ > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On

Re: [asterisk-users] Asterisk version survey

randulo wrote: > On Nov 27, 2007 8:17 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: >> Results should be withheld until the end of the specified polling period. > > Gh! Ok, I'll take 'em off. > > We can give this a few days then , and when things stop, I'll publish. > Open to suggestions.

Re: [asterisk-users] Snom phones, blinking lights and call pickup

On Tue, Nov 27, 2007 at 04:36:33PM +, Steve Davies wrote: I would first ask yourself what you expect to happen if (for example) you have a snom sidecar, and there are 42 lights all blinking at the same time? I do not think that your feature request stands the "usability" test :) Well, of co

[asterisk-users] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?

Hi, I have an older phone with touch screen from Philips. It have it connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number

Re: [asterisk-users] SIP port 5060 closed - how do I open it?

Yes I have a sip.conf, contents as follows: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes allowexternaldomains=no allowexternalinvites=no allowguest=no allowsubscribe=no allowtransfer=yes alwaysauthreject=no autodomain=no callevents=no canreinvite=nonat comp

Re: [asterisk-users] Restricting the manager interface to a number?

On Tue, Nov 27, 2007 at 05:22:59PM +0100, Philipp Kempgen wrote: Currently this type of restriction needs to be done in whatever "proxy" script you use to connect to the AMI. Okay, thanks for the answer. Shade and sweet water! Stephan -- | Stephan SeitzE-Mail: [EM

Re: [asterisk-users] Snom phones, blinking lights and call pickup

Stephan Seitz wrote: > The ISDN PBX in our company building does the following: > - some phones in the office area are monitor phones for a range of >numbers (about 5); In Asterisk's terms that is a "pickup group". > - now one of these numbers are called: >1. the phone with this number

[asterisk-users] Lost setting up IAXmodem after drive crash

A few weeks ago, I lost my Trixbox that was all set up with Hylafax and IAXmodem. I am trying to set it up for email > procmail > faxmail > iaxmodem > asterisk >sipext > ATA (with attached fax). I have followed all the instructions on creating the IAX extension and configuring the IAXmodem co

Re: [asterisk-users] Recommendations for 100 Wifi SIP phone setup

Am Montag, den 26.11.2007, 22:39 -0500 schrieb [EMAIL PROTECTED]: > On Nov 26, 2007 9:52 AM, Alberto Pastore <[EMAIL PROTECTED]> wrote: > > > I also found the Pirelli DP-L10 dual phone to be an excellent sip client > > with good roaming support and discrete battery saving capability. > > (Used in

Re: [asterisk-users] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?

Am Dienstag, den 27.11.2007, 18:00 +0100 schrieb Robert Rozman: > Hi, > > I have an older phone with touch screen from Philips. It have it connected > to Sipura 3000 FXS port and majority of features work ok. > > But phone also has touchscreen and web browser that I'd love to use for > accessin

Re: [asterisk-users] SIP port 5060 closed - how do I open it?

Zaheer, On 28/11/07 9:28 AM, "Zaheer K. Master" <[EMAIL PROTECTED]> wrote: > Yes I have a sip.conf, contents as follows: >From the CLI can you confirm SIP is running by pasting the results of 'module show like sip' Cheers Nick. ___ --Bandwidth and

Re: [asterisk-users] Snom phones, blinking lights and call pickup

Hi! >2. if the call is not answered after a short time, a light (phone > layout is similiar to the Snom phones) begins to blink at the > monitor phones; caller ID and the target number are displayed; if > you press the button next to the blinking light, you pick-up the >

Re: [asterisk-users] Lost setting up IAXmodem after drive crash

oops. I meant to post this to the Hylafax list :-[ And with a reboot, at least it made the call. Now to get the fax to print! Robert Moskowitz wrote: > A few weeks ago, I lost my Trixbox that was all set up with Hylafax and > IAXmodem. > > I am trying to set it up for > > email > procmail > f

Re: [asterisk-users] Urgent question.

If the hunt-group is properly done, you should be able to "busy-out" members of a trunk for maintenance. Otherwise, if the individual trunks have numbers (unpublished) assigned to all the circuits in the group, you could always send a Redirect() to that any of the other trunks' numbers. -Phili

Re: [asterisk-users] Urgent question.

Anyone have an application to robo-dial an outgoing conference call? ;-) You could tie up all your circuits with outbound calls... If you hairpin them at the switch, you shouldn't incur any usage costs... Steve Totaro wrote: > To answer the question, there is currently no way to busy out a cha

Re: [asterisk-users] Snom phones, blinking lights and call pickup

Philipp von Klitzing wrote: >>2. if the call is not answered after a short time, a light (phone >> layout is similiar to the Snom phones) begins to blink at the >> monitor phones; caller ID and the target number are displayed; if >> you press the button next to the blinkin

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