Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

Olivier <[EMAIL PROTECTED]> writes: > To get a polite "go to hell !" in return ? ;-) I think the vendors will be nicer than that. Asterisk has a good bit of the VoIP PBX market. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digi

Re: [asterisk-users] How to check if a SIP phone isforwardedwithoutringing it ?

"Steve Langstaff" <[EMAIL PROTECTED]> writes: > I agree that sending an OPTION message from the Asterisk server could > well have a low processing load. > > The original poster wanted to use OPTIONS sent FROM the Asterisk server > to query the phone state, so I don't think your concerns about rece

Re: [asterisk-users] Asterisk 1.4 and ISDN-BRI support

stoffell schrieb: Has anyone been able to get ISDN-BRI support to work reliably on Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro, kernel, modules, versions, config files). Maybe your best bet is using bristuff, the bristuff-0.4.0 series are tests for asterisk 1.4,

Re: [asterisk-users] Asterisk 1.4 and ISDN-BRI support

On Thu, Jan 10, 2008 at 07:27:12AM +0100, stoffell wrote: > > Has anyone been able to get ISDN-BRI support to work reliably on > > Asterisk 1.4? If so, I'd love to know how you did it (hardware, > > distro, kernel, modules, versions, config files). > > Maybe your best bet is using bristuff, the br

Re: [asterisk-users] HFC-S zap channels always busy

On Sat, Jan 05, 2008 at 04:27:25PM +0100, Patrick wrote: > > On Sat, 2008-01-05 at 01:18 +0100, Jaap Winius wrote: > > Quoting Michiel van Baak <[EMAIL PROTECTED]>: > > > > >> I don't know about NL but in the UK, multiple ISDN2e lines have to be > > >> configured as bri_cpe_ptp not bri_cpe_ptmp.

Re: [asterisk-users] Which IP Phone is really the best?

On Aug 31, 2007 7:11 PM, William Herrera <[EMAIL PROTECTED]> wrote: > Out of all the IP Phones out there, which one is the best and why? My experiences are with Polycom (ip500) and the Linksys/Cisco SPA94?. I like both but they are different. The best suggestion on this thread was to pick 4 and sh

Re: [asterisk-users] Which IP Phone is really the best?

I'm in Europe (yeah, that does matter when choosing a good phone!) .. Some of my (and my customers') favorites: - Polycom (pretty much all of them) - Thomson ST2030 - Siemens Gigaset C450 IP dect (for wireless phones) cheers, stoffell --- http://www.electromarket.be ___

Re: [asterisk-users] Asterisk 1.4 and ISDN-BRI support

> Has anyone been able to get ISDN-BRI support to work reliably on > Asterisk 1.4? If so, I'd love to know how you did it (hardware, > distro, kernel, modules, versions, config files). Maybe your best bet is using bristuff, the bristuff-0.4.0 series are tests for asterisk 1.4, I haven't tested the

[asterisk-users] forward call intended for another domain

I'm new here. For a registered SIP 'friend' that dials an address not handled by my server (say [EMAIL PROTECTED]), how do I get Asterisk to forward that call to ekiga.net? Thanks. Randall ___ -- Bandwidth and Colocation Provided by http://www.api-

Re: [asterisk-users] IEEE 802.1x capable sip phones

I called Cisco and they are so far the only vendor that offers it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Moskowitz Sent: Wednesday, January 09, 2008 11:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [aster

Re: [asterisk-users] IEEE 802.1x capable sip phones

Jeronimo Romero wrote: > > Does anyone know if sip phones from any of the major IP phone vendors > support 802.1x authentication? Any feedback would be greatly appreciated. > This is so unlikely. I worked on 802.1X and 802.11i. There is just too much overhead there. No way to meet the ITU 50ms

Re: [asterisk-users] OT: Traffic Shaping

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Erik Anderson wrote: > On Jan 9, 2008 9:40 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: >> Heh yeah that's what I was thinking of doing. What's the traffic >> shaping like? Can I specify max bandwidth etc or use hfsc shaping? > > DD-WRT will do both

Re: [asterisk-users] IEEE 802.1x capable sip phones

Im pretty sure the Cisco Unified IP Phones 7900 Series phones support this, Dont quote me on it but its worth checking out Kev Jeronimo Romero wrote: > > Does anyone know if sip phones from any of the major IP phone vendors > support 802.1x authentication? Any feedback would be greatly appreci

[asterisk-users] IEEE 802.1x capable sip phones

Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. == Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.

Re: [asterisk-users] OT: Traffic Shaping

On Jan 9, 2008 9:40 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: > > Heh yeah that's what I was thinking of doing. What's the traffic > shaping like? Can I specify max bandwidth etc or use hfsc shaping? DD-WRT will do both HTB and HFSC shaping, though I've only ever used HTB. Here's the dd-wrt w

Re: [asterisk-users] Asterisk 1.4 and ISDN-BRI support

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jaap Winius wrote: > Hi list, > > Has anyone been able to get ISDN-BRI support to work reliably on > Asterisk 1.4? If so, I'd love to know how you did it (hardware, > distro, kernel, modules, versions, config files). > > I've tried to get it to w

Re: [asterisk-users] OT: Traffic Shaping

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Erik Anderson wrote: > On Jan 9, 2008 8:33 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> Hi, >> >> Does anyone know of a cheap (very cheap) dual port traffic shaping box >> (i.e. sub $100) that ca

Re: [asterisk-users] Polycom 550 IP SoundStation Fuzzy Voice Quality

Mike Coakley wrote: > I'm setting up a new Asterisk system on a Dell server and I'm getting > "fuzzy" voice between the Polycom IP SoundStation 550 and the Asterisk > Probably related to this bug: http://bugs.digium.com/view.php?id=11243 Doug -- Ben Franklin quote: "Those who would gi

[asterisk-users] Asterisk 1.4 and ISDN-BRI support

Hi list, Has anyone been able to get ISDN-BRI support to work reliably on Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro, kernel, modules, versions, config files). I've tried to get it to work on a Debian etch system with an HFC-PCI card and the zaptel package (v1.4.

Re: [asterisk-users] OT: Traffic Shaping

On Jan 9, 2008 8:33 PM, Matt Riddell <[EMAIL PROTECTED]> wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hi, > > Does anyone know of a cheap (very cheap) dual port traffic shaping box > (i.e. sub $100) that can be configured for IAX/SIP? Pick up a Linksys WRT54GL and install dd-wrt on

Re: [asterisk-users] Two lines for outgoing calls

Dominik, apart from the good responses, please get rid of the 't' in the options of dial or you will be allowing the called party to transfer the call while you are paying. Regards, Jonathan GF On Dec 26, 2007 3:32 PM, Dominik Zalewski <[EMAIL PROTECTED]> wrote: > Dear All, > > I'm using Ast

[asterisk-users] OT: Traffic Shaping

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Does anyone know of a cheap (very cheap) dual port traffic shaping box (i.e. sub $100) that can be configured for IAX/SIP? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Grea

Re: [asterisk-users] x100p wcfxo hangup on outgoing calss

Hi Miguel, i'm in Spain like you. For a normal operational system inmediate should be set to no. Busycount and Busydetect can be improved performance with busypattern. The pattern should be shown in the CLI, just take a look. In Spain we use Kewlstart. If you card allows it try use Julian J. Mene

Re: [asterisk-users] Two Asterisk Boxes Playing Together

Google is your friend. http://www.google.com.au/search?hl=en&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&hs=jR6&q=asterisk+iax+two+servers&btnG=Search&meta= Shane D wrote: > Okay, here's the dal. > > Me and my friend both have asterisk boxes. I want to be able to type > extension 27 on my

Re: [asterisk-users] Two Asterisk Boxes Playing Together

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Shane D wrote: > Okay, here's the dal. > > Me and my friend both have asterisk boxes. I want to be able to type > extension 27 on my end and get his extension 27, and he wants to be > able to type 277 on his end and get my extension . We both have

Re: [asterisk-users] Two Asterisk Boxes Playing Together

You can do it over SIP or IAX2. What you would do is set up a SIP or IAX2 trunk between the two Asterisk boxes (type=peer) and then define extensions in your dial plan such that those extensions are routed over the trunk. In sip.conf, define your trunk. I'm going to use SIP in this example, b

[asterisk-users] Two Asterisk Boxes Playing Together

Okay, here's the dal. Me and my friend both have asterisk boxes. I want to be able to type extension 27 on my end and get his extension 27, and he wants to be able to type 277 on his end and get my extension . We both have FQDN's, and would like to see about doing this either over sip or IAX..

Re: [asterisk-users] Intercom & Paging with Polycoms

You can use app_page. If you call a local channel that uses app_chanisavail first then you should be able to call as many as you need to. You can actualy break it down in groups that way. On Jan 9, 2008 1:28 PM, Rob Schall <[EMAIL PROTECTED]> wrote: > I've been able to page to a specific phone (i

[asterisk-users] FXOTUNE update

Hey all, First of all, some background: Fxotune is a utility that is used to tune the hybrid on FXO modules For all of you with FXO modules out there, fxotune can help you adjust the analog and digital hybrid that is on the FXO interface and tune it so that it maximizes echo return loss. This

Re: [asterisk-users] Busy notification with call limiting byGROUP_COUNT()

On 1/9/08, Don Pobanz <[EMAIL PROTECTED]> wrote: > Peter Galiovsky wrote on Wednesday, January 09, 2008 9:39 AM > > I want the user to be presented as busy if he has at > > least one call active, be it incoming or outgoing. How > > should I set things up to achieve this? > > I have a very similar n

Re: [asterisk-users] Busy notification with call limiting byGROUP_COUNT()

At 01:42 PM 1/9/2008, you wrote: >I have a very similar need. We are using call queues and would like to >have only 1 call presented to our trouble call reps at a time, but would >like to give them the ability to initiate an outgoing call on a >different line (even while on an incoming call). We ar

Re: [asterisk-users] Newbie: confusion with the new FXO/FXS card

Or better yet, download the PDF of the Asterisk: The Future of Telephony (aka the starfish book): Glenn Cobb wrote: > Go here > > www.voip-info.org > > and read alot. Almost everything you need to know (or a link to it) can > be found through there. > > Seriou

Re: [asterisk-users] Limiting number of simultaneous calls in E1line

You can also set a busy tone in asterisk. You can send it to a context that keeps track of how many incoming calls there are and if there are 10 channels in ues then tell the 11th and on that the line is busy. - Original Message - From: Christian Victor To: Asterisk Users Mailing

Re: [asterisk-users] WaitExten and Macros

>> > Spoke to soon, forgot that I used Background, which is doing the > same thing. Any reason macro context is not supported? Just > curious. > Okay, I figured out that I can put the macro context into the Background option, that works. Bit messy in the dial plan, but it will work. Would be

Re: [asterisk-users] Busy notification with call limiting byGROUP_COUNT()

Peter Galiovsky wrote on Wednesday, January 09, 2008 9:39 AM > I want the user to be presented as busy if he has at > least one call active, be it incoming or outgoing. How > should I set things up to achieve this? I have a very similar need. We are using call queues and would like to have only

[asterisk-users] IAXy ringing

When I make calls from my IAXy I don't hear any ringing most of the time. I've tried using the r option on the asterisk dial application to "indicate ringing to the calling party" but that didn't make a difference. Anything else I can try?___ -- Bandwi

Re: [asterisk-users] Linksys SPA-9xx Audio Issues

Distorted and broken noise at the remote end. The odd thing is I can never reproduce the issue but it very constant. I have set the 0.020 setting and I will continue to test with G723. On Jan 8, 2008 7:57 PM, Daniel Cole <[EMAIL PROTECTED]> wrote: > Can you describe the issue more please? Can the

[asterisk-users] Polycom 550 IP SoundStation Fuzzy Voice Quality

I'm setting up a new Asterisk system on a Dell server and I'm getting "fuzzy" voice between the Polycom IP SoundStation 550 and the Asterisk server. I've checked all of my codec settings and both the Asterisk and the Polycom agree on u-Law encoding. I'm using the latest release of the Aster

Re: [asterisk-users] WaitExten and Macros

>> You may just want a "Read" if you know how many numbers you're >> looking for. >> >> Rob >> >> > This worked, thanks!!! Feel silly for not seeing that option, > guess I was being lazy. > > Tony Plack Spoke to soon, forgot that I used Background, which is doing the same thing. Any reason macr

Re: [asterisk-users] [Zaptel] Checking that TDM card works?

On Wed, 09 Jan 2008 08:32:40 -0600, "Darrick Hartman (lists)" <[EMAIL PROTECTED]> wrote: >Uncomment that if you expect it to work. The module should be listed in >ZAPMODS not in rc.modules. [...] Since you have ZAPMODS commented out, > the zaptel init script doesn't know which modules it should b

Re: [asterisk-users] WaitExten and Macros

> You may just want a "Read" if you know how many numbers you're > looking for. > > Rob > > This worked, thanks!!! Feel silly for not seeing that option, guess I was being lazy. Tony Plack ___ -- Bandwidth and Colocation Provided by http://www.api-di

[asterisk-users] Subscriptions, Firewalls, 489 "Bad Event" and Bug 7608

I am running 1.4 svn-r95946 and have an error -- Got SIP response 489 "Bad event" back from 1.2.3.4 The problem is that this error is being generated by Asterisk trying to send a NOTIFY to a phone behind a NAT firewall that does not exist anymore. The phone was physically decommissioned 3

Re: [asterisk-users] WaitExten and Macros

On Wed, 2008-01-09 at 12:09 -0600, Tony Plack wrote: > I am trying to use a WaitExten in a Macro Bad idea... as far as I know, neither Background nor WaitExten work correctly inside of macros. I'd suggest you use the Read() application instead. -- Jared Smith Community Relations Manager Digium,

Re: [asterisk-users] WaitExten and Macros

You may just want a "Read" if you know how many numbers you're looking for. Rob Tony Plack wrote: > I am trying to use a WaitExten in a Macro, and I am finding that the > extension which is pressed ends up in context of the calling context and not > in the Macro. > > How do you do a WaitExten

[asterisk-users] Intercom & Paging with Polycoms

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that s

Re: [asterisk-users] WaitExten and Macros

On Wednesday 09 January 2008 12:09:29 Tony Plack wrote: > I am trying to use a WaitExten in a Macro, and I am finding that the > extension which is pressed ends up in context of the calling context and > not in the Macro. > > How do you do a WaitExten in a Macro? You cannot. -- Tilghman ___

Re: [asterisk-users] How to check if a SIP phone isforwardedwithoutringing it ?

2008/1/9, Steve Langstaff <[EMAIL PROTECTED]>: > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Johansson Olle E > > Sent: 09 January 2008 14:11 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-user

[asterisk-users] WaitExten and Macros

I am trying to use a WaitExten in a Macro, and I am finding that the extension which is pressed ends up in context of the calling context and not in the Macro. How do you do a WaitExten in a Macro? Tony Plack ___ -- Bandwidth and Colocation Provided b

[asterisk-users] WaitExten and Macros

I am trying to use a WaitExten in a Macro, and I am finding that the extension which is pressed ends up in context of the calling context and not in the Macro. How do you do a WaitExten in a Macro? Tony Plack ___ -- Bandwidth and Colocation Provided b

Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

2008/1/9, Benny Amorsen <[EMAIL PROTECTED]>: > > Olivier <[EMAIL PROTECTED]> writes: > > > As using OPTIONS requests main benefit is to non-phone specific, what > > shall we do when most vendors do not comply with RFC ? > > Write polite letters to the vendors? To get a polite "go to hell !" in re

[asterisk-users] Broken calls

Hello Fellows! I have the following problem: When peers are in a conversation the call broken, it not happen every time, but 60% of calls. :( In Asterisk log I got this message: rtp.c: RTCP Read too short. I'm using a TDM2400 with Asterisk 1.4.14 and zaptel 1.4.7 (Debian Etch) Thank yo

Re: [asterisk-users] Zaptel FXS Cards - Station Distance

Kevin P. Fleming wrote: > Tim Nelson wrote: > >> Hello! In the near future, I'll be deploying an asterisk system that will >> have (in addition to many SIP handsets) four FXS handsets. Is there a known >> limitation to the length of cabling that can be used between a >> Digium/Openvox/Sango

Re: [asterisk-users] Zaptel FXS Cards - Station Distance

On Wednesday 09 January 2008 09:15:08 Tim Nelson wrote: > Hello! In the near future, I'll be deploying an asterisk system that will > have (in addition to many SIP handsets) four FXS handsets. Is there a known > limitation to the length of cabling that can be used between a > Digium/Openvox/Sangoma

Re: [asterisk-users] Zaptel FXS Cards - Station Distance

Tim Nelson wrote: > Hello! In the near future, I'll be deploying an asterisk system that will > have (in addition to many SIP handsets) four FXS handsets. Is there a known > limitation to the length of cabling that can be used between a > Digium/Openvox/Sangoma FXS card and the end station? I un

[asterisk-users] Busy notification with call limiting by GROUP_COUNT()

Hello all, I was wondering what will be the "proper" way to manage BUSY state notification in presence once call-limit, incominglimit and all those settings are gone. I'm using GROUP_COUNT for call limiting in Asterisk 1.4.13 but I have no idea how to set up the settings needed for BUSY notifi

[asterisk-users] Zaptel FXS Cards - Station Distance

Hello! In the near future, I'll be deploying an asterisk system that will have (in addition to many SIP handsets) four FXS handsets. Is there a known limitation to the length of cabling that can be used between a Digium/Openvox/Sangoma FXS card and the end station? I understand the card will ne

Re: [asterisk-users] [asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers

Vinicius Fontes wrote: > Hey guys, I don't know if this is the right place to ask this. I was > thinking about reporting a bug, but maybe it's better to sort out if > this is really a bug or just me being lame. > > I want to record *every* call in my Asterisk box, so I use the > MixMonitor()

Re: [asterisk-users] How to check if a SIP phone isforwardedwithoutringing it ?

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Johansson Olle E > Sent: 09 January 2008 14:11 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] How to check if a SIP phone > isforwardedwithoutringing i

Re: [asterisk-users] [Zaptel] Checking that TDM card works?

Vincent wrote: > On Wed, 09 Jan 2008 06:01:32 -0600, "Darrick Hartman (lists)" > <[EMAIL PROTECTED]> wrote: >> But look in your /etc/rc.conf file for the ZAPMODS variable. You should >> have that variable set to: >> >> ZAPMODS="wctdm" > > Yes indeed: > > #ZAPMODS="wctdm" > > Should I add this

Re: [asterisk-users] What's the best ztdummy?

On Wednesday 09 January 2008, Tzafrir Cohen wrote: > As of kernel 2.6.22 you can use high-resolution timers support in the > kernel, which is better. In that case you'll see the source "HRTimer". Is this still an option with kernel 2.6.23? I didn't find that option in my current kernel sources.

Re: [asterisk-users] Newbie: confusion with the new FXO/FXS card

Go here www.voip-info.org and read alot. Almost everything you need to know (or a link to it) can be found through there. Seriously, its your best starting point regards, Glenn > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Vytenis Sab

Re: [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?

9 jan 2008 kl. 10.46 skrev Steve Langstaff: >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of >> Johansson Olle E >> Sent: 09 January 2008 06:50 >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] How to

Re: [asterisk-users] [Zaptel] Checking that TDM card works?

On Wed, 09 Jan 2008 06:01:32 -0600, "Darrick Hartman (lists)" <[EMAIL PROTECTED]> wrote: >But look in your /etc/rc.conf file for the ZAPMODS variable. You should >have that variable set to: > >ZAPMODS="wctdm" Yes indeed: #ZAPMODS="wctdm" Should I add this module here, or in rc.modules? Are we

[asterisk-users] Newbie: confusion with the new FXO/FXS card

Hello everyone, I'm trying to set up a Asterisk server. I have two cards - one is an BeroNet BN2S0 with two ISDN lines (4 channels): http://www.adcomtec.com/webstore/beronet_bn2s0.php?cat=90 and a Rhino R8FXX with one FXO module and two FXS: http://www.voipsupply.com/product_info.ph

Re: [asterisk-users] [Zaptel] Checking that TDM card works?

Vincent wrote: > On Wed, 9 Jan 2008 12:05:34 +0200, Tzafrir Cohen > <[EMAIL PROTECTED]> wrote: >> wcfxo is not needed. >> >> Basically all you need is: >> >> modprobe >> >> This also pulls all of its dependencies (e.g: zaptel) >> >> modprobe wctdm > > Thanks, but on AstLinux, the modules are no

Re: [asterisk-users] Linksys SPA-9xx Audio Issues

> I have found with a number of clients to who we have installed the LinkSys > phones, that when you get the input gains to 6, that the phones have a > tendency to pick up too much background noise. Have you experienced this > at all? We have a number of customers out there with SPA-942s and have

Re: [asterisk-users] [Zaptel] Checking that TDM card works?

On Wed, Jan 09, 2008 at 12:26:46PM +0100, Vincent wrote: > On Wed, 9 Jan 2008 12:05:34 +0200, Tzafrir Cohen > <[EMAIL PROTECTED]> wrote: > >wcfxo is not needed. > > > >Basically all you need is: > > > > modprobe > > > >This also pulls all of its dependencies (e.g: zaptel) > > > > modprobe wctdm

Re: [asterisk-users] [Zaptel] Checking that TDM card works?

On Wed, 9 Jan 2008 12:05:34 +0200, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: >wcfxo is not needed. > >Basically all you need is: > > modprobe > >This also pulls all of its dependencies (e.g: zaptel) > > modprobe wctdm Thanks, but on AstLinux, the modules are not unloaded: === pbx admin # /e

Re: [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?

Olle, Yes, OPTIONS is too heavy for keep-alives and conflicts with its intended usage - capability discovery without actually placing a call. The IETF seems to be finally reaching a conclusion on how to do keep-alives in a lightweight fashion. These are described in the SIP-outbound draft: http:/

Re: [asterisk-users] What's the best ztdummy?

On Tue, Jan 08, 2008 at 11:29:08AM -0800, Steve Edwards wrote: > I have several servers using ztdummy as the timing source, some CentOS > 4.x, some CentOS 5.x, some Asterisk 1.2.x, some Asterisk 1.4.x. > > "zap show status" differs between the servers: > > ZTDUMMY/1 (source: Linux26) 1

Re: [asterisk-users] conferencing help

Hi Tzafrir, cat /proc/zaptel/* Span 1: ZTDUMMY/1 "ZTDUMMY/1 1" Kernel: 2.6.18-5-686 #1 SMP Zaptel: zaptel-1.2.20.1 OS: Debian GNU/Linux 4.0 i downgraded my zaptel from 1.2.22.1 to 1.2.20.1 but still the same. thanks again regards, nhadie Tzafrir Cohen wrote: > On Wed, Jan 09, 2008 at 03:36

Re: [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?

There is something called as answer-mode in SIP. The idea is to allow the UAC to request the UAS to auto-answer the call. At least in theory, this could be used to check the status of the phone without ringing it. This is obviously not an ideal replacement of OPTIONS. Also, this is a new spec so I'

Re: [asterisk-users] [Zaptel] Checking that TDM card works?

On Wed, Jan 09, 2008 at 10:42:52AM +0100, Vincent wrote: > BTW, is there an order when loading modules for a TDM card? The > OpenVox seems to need zaptel, wctdm, and wcfxo, so I just run this: > > # modprobe zaptel > # modprobe wctdm > # modprobe wcfxo wcfxo is not needed. Basically all you nee

Re: [asterisk-users] Set CDR userfield in a realtime dialplan

I answered my own question ... it was a stupid syntax mistake. CDR(userfield) = "Y" <--- No spaces allowed at all here Mea culpa. On Wed, 2008-01-09 at 08:54 +0100, Yves Räber wrote: > Hello, > > I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have > some trouble with the

Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

Olivier <[EMAIL PROTECTED]> writes: > As using OPTIONS requests main benefit is to non-phone specific, what > shall we do when most vendors do not comply with RFC ? Write polite letters to the vendors? /Benny ___ -- Bandwidth and Colocation Provide

[asterisk-users] remote Snom 360: no voice passing thru

Hi, I have an Asterisk PBX 1.2.18 connected to a remote Snom 360 (firmware ver: 6.5.10), outside my LAN. If I call the Snom, I sometimes cannot hear the called party and the called party cannot hear me. When this happens, I get the following message from Asterisk CLI: NOTICE[19164]: chan_sip.c:

Re: [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Johansson Olle E > Sent: 09 January 2008 06:50 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] How to check if a SIP phone > isforwardedwithout ringing

Re: [asterisk-users] [Zaptel] Checking that TDM card works?

On Tue, 08 Jan 2008 13:43:50 -0500, Jared Smith <[EMAIL PROTECTED]> wrote: >I always find that looking at the files that are generated >under /proc/zaptel is very enlightening as far as showing what the >zaptel drivers are seeing. Thanks for the tip. _

Re: [asterisk-users] Help! channel_find_deadlocked: Avoided initial deadlock for ...

FYI, check the changelog for 1.2.14 to 1.2.25 - IIRC, there are a significant number of deadlock-fixing updates. There is at least one related to the code where that error message is displayed. Regards, Steve On 1/9/08, Douglas Garstang <[EMAIL PROTECTED]> wrote: > > Replying to myself. :) > I ju

Re: [asterisk-users] [Zaptel] Checking that TDM card works?

On Tue, 8 Jan 2008 20:29:20 +0200, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: >This change is simply due to different versions of Zaptel. Zaptel >= >1.4.6 prints "to configure" because this message is printed (and has >always been prinetd) before the configuration is actually applied. Good to know :

Re: [asterisk-users] Set CDR userfield in a realtime dialplan

Hello I'm running the same with no-problem. CDR(userfield)=INCOMING We use also a quite nice patch from Matt Riddell http://bugs.digium.com/view.php?id=9424 that allow to have extra userfield. CDR(userfield2)=${CODEC-IN} CDR(userfield3)=${CODEC-OUT} and so on... This is quite good for custom

Re: [asterisk-users] Set CDR userfield in a realtime dialplan

On Wednesday 09 January 2008 09:54:59 Yves Räber wrote: > Hello, > > I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have > some trouble with the CDR userfield that is not changed when using the > SET command in the realtime dialplan. > In my dialplan (extensions.conf, the file) I

Re: [asterisk-users] conferencing help

On Wed, Jan 09, 2008 at 03:36:06PM +0800, Nhadie wrote: > Hi Matt, > > I tried > > /usr/local/src/zaptel-1.2.22.1# ./zttest -v > > and it just freezes at this. > > Opened pseudo zap interface, measuring accuracy... > > no more outputs, when i cancelled this is what i got. > > --- Results aft

Re: [asterisk-users] conferencing help

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: > Hi Matt, > > I tried > > /usr/local/src/zaptel-1.2.22.1# ./zttest -v > > and it just freezes at this. > > Opened pseudo zap interface, measuring accuracy... > > no more outputs, when i cancelled this is what i got. > > --- Result