Re: [asterisk-users] Fax Machine Options

>and to the OP about the hardware, even a cheap grandsteam ATA will >work just fine... that's what I use on my personal fax machine and it >has no issues. I can't recall a time this year a fax has failed. This >is going over the public internet and then also back out to a voip >provider. We do use

Re: [asterisk-users] Fax Machine Options

On Tue, May 20, 2008 at 12:23 AM, Lee Howard <[EMAIL PROTECTED]> wrote: > Andreas van dem Helge wrote: >> Cisco gateway with T.38 support. That's the only real way to do faxing >> through asterisk. > > Although this statement has marginally more truth to it given the > SIP-only context that the ori

Re: [asterisk-users] Which best practices to build and deploy Asterisk on different hardware ?

2008/5/19 Tzafrir Cohen <[EMAIL PROTECTED]>: > On Mon, May 19, 2008 at 09:17:56PM +0200, Olivier wrote: > > 2008/5/19 Tzafrir Cohen <[EMAIL PROTECTED]>: > > > > > On Mon, May 19, 2008 at 07:29:54PM +0200, Olivier wrote: > > > > Hi, > > > > > > > > Today I'm building Asterisk using steps like this

Re: [asterisk-users] Where does menuselect save your choices?

Sherwood McGowan wrote: > Sherwood McGowan wrote: >> Just a quick question, wanted to see if anyone knew where the >> menuselect app stored your choices. >> >> I think it's menuselect.makeopts but I'm not sure...just thought >> someone might know. >> >> Sherwood McGowan >> >> P.S. I'll post here

Re: [asterisk-users] Asterisk on iPhone

try Fring beta to Iphone http://www.fring.com/blog/?p=226 Alexander Lopez wrote: > Can this also run on an IPod touch??? > > I am almost tempted to go buy one and see > > Alex > > > >> -Original Message- >> From: [EMAIL PROTECTED] [mailto:asterisk-users- >> [EMAIL PROTECTED] On Beh

Re: [asterisk-users] Fax Machine Options

Andreas van dem Helge wrote: > Cisco gateway with T.38 support. That's the only real way to do faxing > through asterisk. Although this statement has marginally more truth to it given the SIP-only context that the original poster provided, it is still substantially inaccurate. There are several

Re: [asterisk-users] Fax Machine Options

Cisco gateway with T.38 support. That's the only real way to do faxing through asterisk. I think a VG200 with newer firmware will support SIP + T.38 but don't buy on my suggestion because I've never used that device outside call manager configuration. Or see if your VoIP provider supports T.38 fa

Re: [asterisk-users] Recording problems, reinvites

So why don't you just disable reinvite? Using 1.4.15 here with no issues with MixMonitor. Then again I've *ALWAYS* disabled reinvite because it never works for me. On Mon, May 19, 2008 at 4:33 PM, Trevor Peirce <[EMAIL PROTECTED]> wrote: > Hello, > > I'm wondering if anyone else has been observ

[asterisk-users] mute a call/ re-invite mid-session?

Hello ppl, Is there anyway to control a call mid-way in terms of sending a re-INVITE with say sendonly, etc. to mute one call leg of a bridged call ?? Looked around, so far, doesnt seem to be possible. If it's not, I think it's quite an important feature (re-INVITES mid-session) for a B2BUA. c

[asterisk-users] Fax Machine Options

Is my only solution to add a fax machine to our VOIP only setup by using an IAXy? I should specify the office people want a traditional fax machine in the sense that fax's be sent and received from a physical unit, they don't want an email to fax setup. They have a dedicated sip did provisioned

Re: [asterisk-users] Concept Clarifications

>Sounds like you want a ring-all queue. Appreciate that pointer. So if I make a queue with a strategy => ringall and add all the extensions I want in it, then send the incoming calls from the sip did into it, do the callers experience being put on hold, or do they experience a ringing line until i

Re: [asterisk-users] Recording problems, reinvites

Hi, I am also experiencing problem with MixMonitor in Asterisk 1.4. Most of our recording is overlapping. How can I fix this problem? On Tue, May 20, 2008 at 4:33 AM, Trevor Peirce <[EMAIL PROTECTED]> wrote: > Hello, > > I'm wondering if anyone else has been observing problems lately with > 1.4.

Re: [asterisk-users] DHCP Failure screws up system

Anthony Francis wrote: > Watch your router and see if the arp entries are changing for the IP's > as this happens, I am very willing to bet you have IP address conflicts > on a mass scale and should consider shutting down all of these phones > and bringing them back up so they all get leases out

Re: [asterisk-users] anyone from Joplin, MO

Sherwood, I don't suppose you could pass on some info concerning who this client was? Could you contact me at bmedlock(at)chcchurch(dot)org? Thanks, Bryson > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Sherwood McGowan > Sent: Thursday, May 15

Re: [asterisk-users] Wireless headsets for Polycom phones

John Lee wrote: > > > > > Anyone have recommendations for wireless headsets that work well with > > Polycom phones and Asterisk? > > > I am waiting for delivery one of the following types: > Plantronics CS351 Monaural SupraPlus Wireless Professional Headset > System Voice Tube > (http://www.plantr

Re: [asterisk-users] Asterisk on iPhone

Can this also run on an IPod touch??? I am almost tempted to go buy one and see Alex > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth > Sent: Monday, May 19, 2008 4:59 PM > To: asterisk-users@lists.digium.com > S

[asterisk-users] Zaptel project being renamed to DAHDI

Last week some changes were made in the Zaptel source code repository referencing a new name (DAHDI) for that code. This has caused some discussion (and confusion) in the community, since it occurred before we had made a public announcement of the name change. However, that announcement is now off

Re: [asterisk-users] Fedora 9 + Asterisk

On Mon, 2008-05-19 at 13:38 -0400, OCG Technical Support wrote: > Anyone tried Asterisk with Fedora 9 (recent release)? I've got it running under Fedora 9... 1.4 branch, 1.6.0 branch, and trunk all seem to work fine. Oh, and by the way, thanks to the hard work of Jeff C. Ollie and Dennis Gilmor

Re: [asterisk-users] Wireless headsets for Polycom phones

> > Anyone have recommendations for wireless headsets that work well with > Polycom phones and Asterisk? > I am waiting for delivery one of the following types: Plantronics CS351 Monaural SupraPlus Wireless Professional Headset System Voice Tube (http://www.plantronics.com/australia/aus/products/c

Re: [asterisk-users] BLF Compatible Phones

Grygoriy Dobrovolskyy wrote: > Hmm, i dont like aastra really much, their transfer management is not human > friendly ;) > Fwiw, their new 9i series (9143, et al) have BLF/xfer, speeddial/xfer feature that features one button transfer, eliminating the need for the transfer button for any prepro

Re: [asterisk-users] Concept Clarifications

Joseph L. Casale wrote: > Hi, > My system is setup and working, I can dial out, in and to demo extensions > that play music etc. > I would like to read up on some final topics before I get it running in > production but don't really > know what to look for. > > If an incoming call from a SIP DID

[asterisk-users] Concept Clarifications

Hi, My system is setup and working, I can dial out, in and to demo extensions that play music etc. I would like to read up on some final topics before I get it running in production but don't really know what to look for. If an incoming call from a SIP DID is to ring across {n} phones for exampl

Re: [asterisk-users] Fedora 9 + Asterisk

On May 19, 2008 06:49:23 pm Kevin Smith wrote: > I almost hate to admit this...but I'm still running Asterisk 1.2 on > Fedora 4 :D IMO theres nothing wrong with running an old version of * or an old version of the OS... as long as the box doesn't have a public IP bound to it. If its an internal-

Re: [asterisk-users] Fedora 9 + Asterisk

I almost hate to admit this...but I'm still running Asterisk 1.2 on Fedora 4 :D However, I'm planning on upgrading to 1.4 but it has been working out just fine so far and I just can't find time to upgrade. Otherwise, at least with Fedora I have had no major issues running Asterisk. Most of any

Re: [asterisk-users] Asterisk first time user

I guess take this into consideration if time isn't a real factor (however, I'm sure it is). In my experience I found it best to start learning with the configuration files only then use the GUI. The GUI's are very nice and handy, but sometimes I feel they lack what you could do with manually cr

Re: [asterisk-users] Googles 411 services

On Mon, 2008-05-19 at 11:13 +0100, Adrian Marsh wrote: > Hi Brian, > > Thanks for the reply. I tried searching for your posts, but no luck. I find effective use of the Internet absolutely depends on knowing how to search for stuff. http://www.google.ca/search?q=site%3Alists.digium.com+murrell&i

Re: [asterisk-users] PRI debugging ...

In article <[EMAIL PROTECTED]>, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: > > You might try turning off echo cancellation to see if your D-channel > performance improves. That would be a good test to tell if you should > look into perhaps getting either a faster CPU or a hardware echo > c

Re: [asterisk-users] Asterisk on iPhone

On Mon, May 19, 2008 at 10:48:31PM +0200, Andrea Cristofanini wrote: > http://www.mgamble.ca/oss/iphone_asterisk/ Damn. It's running 1.4, so I can't aim VICIdial at it. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer

Re: [asterisk-users] Asterisk first time user

http://easyvoxbox.org/ is pretty sweet. Thanks, Steve Totaro On Mon, May 19, 2008 at 4:41 PM, Aaron Stranberg <[EMAIL PROTECTED]> wrote: > Thanks for the response, to clarify a bit, I don't mind the hands on > installation but after the system is up and running I would like to have a > GUI front

Re: [asterisk-users] Asterisk on iPhone

http://www.mgamble.ca/oss/iphone_asterisk/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Not hearing first prompt

This does the trick for me. Sorry for not posting it earlier As long as there is no answer the channel takes a second pause by jumping to (wait). As soon as the status is not "NOANSWER" anymore the routine jumps to (go_on) and plays the "special_message" or any other message of our choice.

Re: [asterisk-users] Asterisk first time user

Thanks for the response, to clarify a bit, I don't mind the hands on installation but after the system is up and running I would like to have a GUI front end that I can dump off to less linux friendly folks for creation of new extensions, voicemail setup etc.. Thanks again for the response. -Aar

[asterisk-users] Recording problems, reinvites

Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor

Re: [asterisk-users] Understanding Incoming sip DID handling

Joseph L. Casale wrote: >> Yes, in your dialplan you should have one extension set up for the first >> number and where to send it, and a second for the other. >> > > So, if the sip.conf config sends the did into the [incoming] context > and its phone number is 555-1212, would this be the righ

Re: [asterisk-users] Googles 411 services

Yea... tf.voipmich.com went down ~ 1 week ago when you call through it the calls just ring forever. Some issue seems to happen with tf.voipmich.com at least once a year and it always takes a long time to fix. tollfreegateway.com seems to be working.. On Sat, May 17, 2008 at 1:38 PM, Adrian M

[asterisk-users] Asterisk Jobs Spring Special - Free Postings for Employers!

Greetings, Asterisk-Jobs.com has started the spring special for all it's employers. For the next 2 months all employer postings will be free on the site. We've also added a couple new features in the past months including Print to PDF for applicants, making it easier than ever to save and reply t

Re: [asterisk-users] DHCP Failure screws up system

Watch your router and see if the arp entries are changing for the IP's as this happens, I am very willing to bet you have IP address conflicts on a mass scale and should consider shutting down all of these phones and bringing them back up so they all get leases out of the new servers pool inste

Re: [asterisk-users] Asterisk first time user

On May 19, 2008 03:21:34 pm Aaron Stranberg wrote: > Folks, > We are a small office with remote users less than 20 total phone > extensions, and I am looking for some guidance on choosing between > asterisknow and a centos/ubuntu or any other os with an asterisk + > asteriskgui build out? Lookin

Re: [asterisk-users] Dutch Asterisk mailing list?

On 15:34, Mon 19 May 08, Jaap Winius wrote: > Quoting Michiel van Baak <[EMAIL PROTECTED]>: > > > I can start it if you want. > > Yeah, why not. Eventually, it may help to solve this and other > KPN-related problems, which can only be good for Asterisk adoption in > the Netherlands. I'll sign

Re: [asterisk-users] Dialplan Visualization, Rating System Unveiled!

On Mon, May 19, 2008 at 3:15 PM, John Signorello <[EMAIL PROTECTED]> wrote: > Matthew Gibson wrote: > > Hello Fellow Asterisk Enthusiasts, > > With all the talk lately of dialplan visualization, we've decided to create > a geeky, but fun and useful site. > > Visit http://www.ratemydialplan.com to

Re: [asterisk-users] Which best practices to build and deploy Asterisk on different hardware ?

On Mon, May 19, 2008 at 09:17:56PM +0200, Olivier wrote: > 2008/5/19 Tzafrir Cohen <[EMAIL PROTECTED]>: > > > On Mon, May 19, 2008 at 07:29:54PM +0200, Olivier wrote: > > > Hi, > > > > > > Today I'm building Asterisk using steps like this : > > > > > http://mikeoverip.wordpress.com/2008/03/29/aste

[asterisk-users] VoIP Termination To Togo and Cameroon

Does anyone has termination to Togo and Cameroon? Thanks. You can send me a private message if you do. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http:/

[asterisk-users] Asterisk first time user

Folks, We are a small office with remote users less than 20 total phone extensions, and I am looking for some guidance on choosing between asterisknow and a centos/ubuntu or any other os with an asterisk + asteriskgui build out? Looking to get up and going quick with some method of GUI administr

Re: [asterisk-users] Which best practices to build and deploy Asterisk on different hardware ?

2008/5/19 Tzafrir Cohen <[EMAIL PROTECTED]>: > On Mon, May 19, 2008 at 07:29:54PM +0200, Olivier wrote: > > Hi, > > > > Today I'm building Asterisk using steps like this : > > > http://mikeoverip.wordpress.com/2008/03/29/asterisk-compilation-and-installation-on-debian-etch/ > > > > As you can see,

Re: [asterisk-users] Dialplan Visualization, Rating System Unveiled!

Matthew Gibson wrote: Hello Fellow Asterisk Enthusiasts, With all the talk lately of dialplan visualization, we've decided to create a geeky, but fun and useful site. Visit http://www.ratemydialplan.com to share your diagrams of your Asterisk Dialplans and have other users rate them. To ge

Re: [asterisk-users] Recall: Newbie Asterisk: Install Asterisk as non-root

On Mon, May 19, 2008 at 09:08:05AM +0100, Steven Howes wrote: > On 19 May 2008, at 05:42, Lee, John (Sydney) wrote: > > Lee, John (Sydney) would like to recall the message, "[asterisk- > > users] Newbie Asterisk: Install Asterisk as non-root". > > > Fail. /user chuckles -- Jay R. Ashworth

Re: [asterisk-users] Understanding Incoming sip DID handling

You have DID1 on sip trunk 1 (unlimited channels) You have DID2 on sip trunk 2 (restricted channels) You want all you outgoing traffic to go out sip trunk 1 == Sherwood McGowan wrote: Joseph L. Casale wrote: Hi, What is the method (preferred) way Asterisk handles the incoming sip lines

Re: [asterisk-users] Understanding Incoming sip DID handling

On Mon, 19 May 2008, Joseph L. Casale wrote: > How do I elegantly write something to catch the area code, if the number > dialed was > 1-xxx-555-1212? Or does the incoming number from my sip provider always show > up to me > as an 11 digit number: > > exten => 19495551212,1,`Do Something` for ex

Re: [asterisk-users] Which best practices to build and deploy Asterisk on different hardware ?

On Mon, May 19, 2008 at 07:29:54PM +0200, Olivier wrote: > Hi, > > Today I'm building Asterisk using steps like this : > http://mikeoverip.wordpress.com/2008/03/29/asterisk-compilation-and-installation-on-debian-etch/ > > As you can see, the first requirement is to download various dependencies >

Re: [asterisk-users] Understanding Incoming sip DID handling

>Yes, in your dialplan you should have one extension set up for the first >number and where to send it, and a second for the other. So, if the sip.conf config sends the did into the [incoming] context and its phone number is 555-1212, would this be the right way: exten => 5551212,1,`Do Something`

[asterisk-users] Dialplan Visualization, Rating System Unveiled!

Hello Fellow Asterisk Enthusiasts, With all the talk lately of dialplan visualization, we've decided to create a geeky, but fun and useful site. Visit http://www.ratemydialplan.com to share your diagrams of your Asterisk Dialplans and have other users rate them. To get people started with Dialpl

[asterisk-users] DHCP Failure screws up system

Maybe someone could point in the right direction. I have a small facility that's running around 40 Polycom 301/501 phones, Asterisk 1.4.18 running under Mandriva 2007.1. The phones were assigned a DHCP address in the 10.10.10.x range. Today, the DHCP server failed and to get them back online,

Re: [asterisk-users] BLF Compatible Phones

I have the SPA962 with the SPA932 side car and it works great, once I got it to configure correctly for my Asterisk PBX system, which has a Digium TDM03B WILD CARD installed. I like the Cisco/Linksys SPA models because they have good and easy configuration webpages for the VOIP phones. Date:

Re: [asterisk-users] BLF Compatible Phones

On May 19, 2008 12:51:09 pm Grygoriy Dobrovolskyy wrote: > Hmm, i dont like aastra really much, their transfer management is not human > friendly ;) What do you mean by that? I've run my Aastra's with BLF using both Aastra's 'blf' mode and 'blfxfer' mode... the former is basically attended tran

Re: [asterisk-users] Which best practices to build and deploy Asterisk on different hardware ?

On Mon, May 19, 2008 at 1:29 PM, Olivier <[EMAIL PROTECTED]> wrote: > Hi, > > Today I'm building Asterisk using steps like this : > http://mikeoverip.wordpress.com/2008/03/29/asterisk-compilation-and-installation-on-debian-etch/ > > As you can see, the first requirement is to download various depen

Re: [asterisk-users] Understanding Incoming sip DID handling

Joseph L. Casale wrote: > Hi, > What is the method (preferred) way Asterisk handles the incoming > sip lines? I am currently trying to setup two lines, one has > unlimited in/out channels and the other phone number has only two. > > The provider has given a macro that manages dialing out on the two

Re: [asterisk-users] Dutch Asterisk mailing list?

Erik de Wild: Tripple-o wrote: >>> "What is the most reliable method for Asterisk >>> to detect the Called ID for incoming calls on >>> an analog line in the Netherlands?" >>> > > In Holland you have to pay to receive cid info on the incoming line. > If you don't pay at the mo

Re: [asterisk-users] nokia 770 has a build in mic, asterisk and iphone

And how about touchmods ? 2008/5/19 Erik de Wild: Tripple-o <[EMAIL PROTECTED]>: > > > > > > was almost tempted to try it, but time was short at the time, and > > holding an N770 to my head seemed a bit silly.. (built in mic and > > speakers, but no socket for an external mic) > > > > Gordon > >

[asterisk-users] Understanding Incoming sip DID handling

Hi, What is the method (preferred) way Asterisk handles the incoming sip lines? I am currently trying to setup two lines, one has unlimited in/out channels and the other phone number has only two. The provider has given a macro that manages dialing out on the two possible servers. Would I match o

[asterisk-users] outbound calls on PRI all congested

Hi, I am just trying to but an Asterisk server between a German Telecom (Deutsche Telekom) PMUX and a Siemens HiCom. Inbound Calls (Telekom -> HiCom) work like charm. Outbound calls however all end up being congested: -- Executing Dial("Zap/50-1", "Zap/g1/06151343|180|WT") in new st

Re: [asterisk-users] Dutch Asterisk mailing list?

>> "What is the most reliable method for Asterisk >> to detect the Called ID for incoming calls on >> an analog line in the Netherlands?" In Holland you have to pay to receive cid info on the incoming line. If you don't pay at the moment you can start with that. There are 2 ways for

[asterisk-users] Fedora 9 + Asterisk

Anyone tried Asterisk with Fedora 9 (recent release)? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user

Re: [asterisk-users] Which best practices to build and deploy Asterisk on different hardware ?

I recommend using Rpaths Asterisk Now distribution (and ignore the GUI) . My organization has been very happy with this Digium Supported- Pre Built Distro, which, is essentially Asterisk Business Edition. (Without the support contract) Matt Olivier wrote: Hi, Today I'm building Asterisk

[asterisk-users] Which best practices to build and deploy Asterisk on different hardware ?

Hi, Today I'm building Asterisk using steps like this : http://mikeoverip.wordpress.com/2008/03/29/asterisk-compilation-and-installation-on-debian-etch/ As you can see, the first requirement is to download various dependencies such as gcc, g++. As I'm trying to centralize everything (configurati

Re: [asterisk-users] Using a Loopback Plug for an RJ-45 EthernetInterface for testing a Digium Card

> I can't speak to exactly what the alarm status stuff does if the port > you're looping expects to have a PRI plugged into it: I would expect > "Green, but no actual traffic", but I could be wrong, I'm a bit new on > that front. Just for the record, this is generally a correct statement. I can't

Re: [asterisk-users] nokia 770 has a build in mic, asterisk and iphone

> > was almost tempted to try it, but time was short at the time, and > holding an N770 to my head seemed a bit silly.. (built in mic and > speakers, but no socket for an external mic) > > Gordon I run Asterisk on a Nokia 770 and as a mini pbx it run pretty smooth. I used it, just for fun an

Re: [asterisk-users] BLF Compatible Phones

Hmm, i dont like aastra really much, their transfer management is not human friendly ;) Polycom or Snom is our choice (snom 340) 2008/5/18 Sigma Networks <[EMAIL PROTECTED]>: > [EMAIL PROTECTED] wrote: > > I am new to asterisk and am looking to setup a small office with 4-6 IP > > phones and 4 a

[asterisk-users] Wireless headsets for Polycom phones

Anyone have recommendations for wireless headsets that work well with Polycom phones and Asterisk? Thanks, Mike Clark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update optio

Re: [asterisk-users] Not hearing first prompts

Brent Davidson wrote: > Another solution that works for me is to add "Playback(silence/1)" > just before whatever you are about to do. Something about the > playback command opens the channel up. > > -Brent > > Sherwood McGowan wrote: >> Alan Lord wrote: >> >>> Sherwood McGowan wrote: >>> >>

Re: [asterisk-users] Not hearing first prompts

Another solution that works for me is to add "Playback(silence/1)" just before whatever you are about to do. Something about the playback command opens the channel up. -Brent Sherwood McGowan wrote: Alan Lord wrote: Sherwood McGowan wrote: Hrm...I have encount

Re: [asterisk-users] Asterisk on iPhone

On Mon, 2008-05-19 at 11:33 +0100, Gordon Henderson wrote: > I was almost tempted to try it, but time was short at the time, and > holding an N770 to my head seemed a bit silly.. (built in mic and > speakers, but no socket for an external mic) The N810 comes with a set of headphones that have a

Re: [asterisk-users] PRI debugging ...

Gordon Henderson wrote: > On Fri, 16 May 2008, Gordon Henderson wrote: > >> Have a problem with an ISDN30 line in the UK. > > So following up my own post.. I've not solved this issue, but I think I > know what causes it. > > This was my experiment to put 2 cards in one 1.3GHz system - a TDM400

Re: [asterisk-users] Asterisk temporary hangs when no internet connection

Seems like it would have been much simpler just to correct your /etc/hosts file. Most systems check /etc/hosts before consulting DNS for IP and host information. Loic DIDELOT wrote: > Hello, > sorry for dropping in so late but I encountered the same problems so I > started playing around with I

Re: [asterisk-users] Asterisk temporary hangs when no internet connection

Hello, sorry for dropping in so late but I encountered the same problems so I started playing around with IPTABLES and DNS servers. For me it looks like asterisk has a problem because it seems to do reverse dns lookups (PTR records). For me the following helped: I installed a local dns recursor

Re: [asterisk-users] Is this the forum for Business Edition of Asterisk

- "vision admin" <[EMAIL PROTECTED]> wrote: > Just wanted to know if this is the forum to discuss errors for > Asterisk Business Edition? If you want a resolution to your ABE issue, the best strategy is to contact Digium support. ___ -- Bandwidt

Re: [asterisk-users] Asterisk concurrent calls count

On Mon, May 19, 2008 at 09:54:47AM -0400, Steve Totaro wrote: > As you have conveniently snipped what I had previously proposed to > benchmark then your silly proposal of local channels is sneaky at the > least and at most an attempt to discredit any of the benchmarks. > > Maybe one day Xorcom wi

Re: [asterisk-users] where did the switch statement come from?

The Switch statement is used to bring any external dialplan configuration into an asterisk context, such as realtime configs and even DUNDI. http://www.voip-info.org/wiki-DUNDi To answer your actual question. Rizwan Hisham wrote: > where can i find details about both switch statement and dundi

Re: [asterisk-users] Is this the forum for Business Edition of Asterisk

On Mon, May 19, 2008 at 10:17 AM, <[EMAIL PROTECTED]> wrote: > Just wanted to know if this is the forum to discuss errors for Asterisk > Business Edition? > Call Digium directly, that is what ABE is for. Thanks, Steve Totaro ___ -- Bandwidth and Colo

Re: [asterisk-users] Newbie Asterisk: Install Asterisk as non-root

James Sneeringer wrote: >> Also note that asterisk.conf options override command-line options (and >> not the other way around, as you might have learned to expect from most >> other applications). > > Some asterisk.conf options, such as runuser and rungroup, don't appear > to work at all. I can

[asterisk-users] Is this the forum for Business Edition of Asterisk

Just wanted to know if this is the forum to discuss errors for Asterisk Business Edition? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digi

Re: [asterisk-users] Newbie Asterisk: Install Asterisk as non-root

On Sat, May 17, 2008 at 9:21 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Fri, May 16, 2008 at 06:32:30PM -0500, James Sneeringer wrote: >> The safe_asterisk script monitors the actual asterisk process, and if >> it dies for some reason, > > Not "for some reason". For instyance, if asterisk de

Re: [asterisk-users] Asterisk concurrent calls count

On Mon, May 19, 2008 at 9:41 AM, Steve Totaro <[EMAIL PROTECTED]> wrote: > On Mon, May 19, 2008 at 7:39 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: >> On Mon, May 19, 2008 at 07:27:00AM -0400, Steve Totaro wrote: >>> On Mon, May 19, 2008 at 3:03 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: >>> > O

Re: [asterisk-users] Asterisk concurrent calls count

On Mon, May 19, 2008 at 7:39 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Mon, May 19, 2008 at 07:27:00AM -0400, Steve Totaro wrote: >> On Mon, May 19, 2008 at 3:03 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: >> > On Mon, May 19, 2008 at 12:13:05AM -0400, Steve Totaro wrote: >> > >> >> I am c

Re: [asterisk-users] Dutch Asterisk mailing list?

Quoting Michiel van Baak <[EMAIL PROTECTED]>: >> Would anyone here happen to know of the existence of a Dutch Asterisk >> mailing list? If so, where can it be found? > > Not that I know off. I didn't think so. Either that, or its existence is a well kept secret! > I can start it if you want. Ye

Re: [asterisk-users] Asterisk concurrent calls count

Alexander Olekhnovich wrote: > Thanks very much for your examples > > On Fri, May 16, 2008 at 8:59 PM, Sherwood McGowan > <[EMAIL PROTECTED] > wrote: > > Alexander Olekhnovich wrote: > > Hi Asterisk Users, > > > > I'm interested in how many concurrent call

Re: [asterisk-users] One way sound when Using Dial cmd without "t" option (SOLVED) Need explanation

The t, much like reinvite = no keeps asterisk listening to the audio stream to detect dtmf input if dtmf mode is in-band, what is happening is that the sip reinvite is failing, due to a firewall rule or a routing problem and you end up with only one connected RTP stream. Asterisk does not "requir

Re: [asterisk-users] where did the switch statement come from?

where can i find details about both switch statement and dundi. On Mon, May 19, 2008 at 4:37 PM, Alexander Lopez <[EMAIL PROTECTED]> wrote: > The switch statement allows you to 'include' a context from another > machine into your machine. > > > > Problems with it was if the other machine was una

Re: [asterisk-users] Asterisk concurrent calls count

On Mon, May 19, 2008 at 07:27:00AM -0400, Steve Totaro wrote: > On Mon, May 19, 2008 at 3:03 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > On Mon, May 19, 2008 at 12:13:05AM -0400, Steve Totaro wrote: > > > >> I am complaining that they should be provided by Digium. I have an > >> early source

[asterisk-users] zaptel errors update

2 of our ISDN lines attached to a te410 card (only 2 in use on that card) both started to give 'reject frame errors' after an zaptel 1.4 update. prev. i was using a version about 2 month old - 'make update' updated it to rev '4301'. I contacted our telco and they say line is clean, not a sing

Re: [asterisk-users] where did the switch statement come from?

The switch statement allows you to 'include' a context from another machine into your machine. Problems with it was if the other machine was unavailable, or even slow to respond, your machine would hang until it timed out. DUNDI has since replaced the functionality of the switch statement

Re: [asterisk-users] (Newbie)Security Risks in opening IAX & Sip Ports

On Mon, May 19, 2008 at 6:59 AM, Shaun <[EMAIL PROTECTED]> wrote: > Hi All, > > Please direct me to solve this concern of my server host. > > Tx > > Shaun > Sent via my BlackBerry from Vodacom - let your email find you! > > There are always risks Thanks, Steve Totaro

Re: [asterisk-users] Asterisk concurrent calls count

On Mon, May 19, 2008 at 3:03 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Mon, May 19, 2008 at 12:13:05AM -0400, Steve Totaro wrote: > >> I am complaining that they should be provided by Digium. I have an >> early source of some funding for benchmarking, so it certainly will >> not be free.

[asterisk-users] (Newbie)Security Risks in opening IAX & Sip Ports

Hi All, Please direct me to solve this concern of my server host. Tx Shaun Sent via my BlackBerry from Vodacom - let your email find you! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUB

[asterisk-users] where did the switch statement come from?

Hi all, I have looked up the applications and function in asterisk but i could not find the help for the switch statement which is used in several places in sample extensions.conf file. i am using asterisk 1.4.2. On voip-info.org the switch statement seems to be used to connect 2 asterisk servers,

Re: [asterisk-users] PRI debugging ...

On Fri, 16 May 2008, Gordon Henderson wrote: > Have a problem with an ISDN30 line in the UK. So following up my own post.. I've not solved this issue, but I think I know what causes it. This was my experiment to put 2 cards in one 1.3GHz system - a TDM400 with 2 x FXO and 2 x FSX and a TE120P

Re: [asterisk-users] Asterisk on iPhone

On Mon, 19 May 2008, Steve Davies wrote: > 2008/5/19 Steven Howes <[EMAIL PROTECTED]>: >> >> On 18 May 2008, at 23:42, Andrea Cristofanini wrote: >>> Hi >>> I just saw this now ! >>> does the microphone and speaker works ? >>> Can you use it like softphone for recive calls ? >>> Regards Andrea >>

Re: [asterisk-users] Asterisk on iPhone

You might be looking for that instead http://sip.free.fr/index.html.en Andrea Cristofanini a écrit : > Hi > I just saw this now ! > does the microphone and speaker works ? > Can you use it like softphone for recive calls ? > Regards Andrea > C F ha scritto: > __

Re: [asterisk-users] Googles 411 services

Hi Brian, Thanks for the reply. I tried searching for your posts, but no luck. Do you have sample code I could see? Or do you just take a +101 result and move to the next in the list?? Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J. Murr

Re: [asterisk-users] Extension not found

On Mon, May 19, 2008 at 11:36 AM, bas karan <[EMAIL PROTECTED]> wrote: > I am new to this concept, Could you explain me little > bit extra please? You will need to put extensions in contexts. The context is a fundamental concept of the dialplan. All extensions are inside a context. In your sip.con

[asterisk-users] - Failed to authenticate user

Hie, I managed to connect two Asterisk box via SIP. My problem is when I login using Realtime SIP, I will get chan_sip.c:8373 check_auth: username mismatch, have <8003000777>, digest has Failed to authenticate user "8003000777" >;tag=as4d11916d when trying to send call to the 2nd server. bu

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