Re: [Asterisk-Users] Help Configuring TDM04B

Did you change the kernel driver? The FXO card uses wcfxo driver and the TDM400P uses the wcfxs (older versions of zaptel) or wctdm (newer versions of zaptel) driver. So, if you replace the FXO card with a TDM card, you need to update the zapata/zaptel config for the extra channels (which you indic

Re: RE: [Asterisk-Users] Any suggestions for an IP phone?

If the phone is behind a firewall, make sure that port 69 is open so that it can reach the TFTP server. Regards, Derek - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, July 12, 2005 10:08 PM Subject: Re: RE: [Aste

[Asterisk-Users] Problem with modem and asterisk

I have a modem and my digium card in my PC. The problem is when i try to establish an incoming call, the modem responds first before the digium card. Is there any way to allow asterisk to get the call first before my modem? ___ Asterisk-Users mailing list

RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?

you could also use the existing phone lines from your Comdial. That way you don't need any new phone lines or a VOIP provider. What do you have connected to it now? --- Roland Zagler <[EMAIL PROTECTED]> wrote: > Hi Newbie, > > I wonder how you could find the mailing list but NOT > the wiki > at

Re: [Asterisk-Users] Help Configuring TDM04B

Hi: I am sorry for the false report, I still have the same problem. You said you had the same problem. What did you do in your case? Regards; Chawki --- [EMAIL PROTECTED] wrote: > I ha the same problem, could be irq issue if you > have zaptel.conf and > zapata.conf configured properly. > modp

Re: RE: [Asterisk-Users] Any suggestions for an IP phone?

Polycom does not support Asterisk. Thsi does not mean phones do not work with it. Rudolf P.S. I am having troubles setting up Polycom 300 with tftp server. By some reason phones always say "can not contact boot server". Phones are set to use tftp and correct boot server IP is set via dhcp. I w

Re: [Asterisk-Users] Any suggestions for an IP phone?

That's just a marketing thing. Just because they are "not supported" under Asterisk doesn't mean they don't work under Asterisk. It just means don't call them (voipsupply or Polycom) if you have problems getting them to work under Asterisk. Otherwise myself and many others on this list wouldn't

Re: [Asterisk-Users] help needed-call recording

I think you're looking at this the wrong way. Take a look at automon in features.conf. Play the "for-quality-purposes" disclaimer/misleader on all incoming calls to these extensions and use the "w" option on Dial(). What you've done below won't record an existing call. Mark Swapna Gupta wrot

[Asterisk-Users] SpanDSP+astfax with multiple fax pages

I've searched the list and could not find any solutions even though this question has been brought up. I'm not able to fax out multiple pages of tiff, where only the first page is received but not the other pages at the receivers end. I have no problem with receiving multi pages at my side. I'm us

RE: [Asterisk-Users] Any suggestions for an IP phone?

According to voipsupply.com http://www.voipsupply.com/product_info.php?cPath=95_112&products_id=817 --Please Note: Polycom phones are not supported under Asterisk Open Source PBX. Polycom certified platform partners include Path Navigator, Broadsoft, Interactive Intelligence, Sphere, Sylantro,

Re: [Asterisk-Users] Help Configuring TDM04B

Hi: I appreciate every user responded to my post. It is working now. I installed the X101P on the next slot beside the TDM04B to see if it still working. Then ztcfg reported no error and I was able to compile asterisk. I know it is not the X101P that fixed, but this is what I did and it is working

Re: [Asterisk-Users] Help Configuring TDM04B

Hi: I appreciate every user responded to my post. It is working now. I installed the X101P on the next slot beside the TDM04B to see if it still working. Then ztcfg reported no error and I was able to compile asterisk. I know it is not the X101P that fixed, but this is what I did and it is working

[Asterisk-Users] personal voicemail , and call transfer --- howto

Could someone please help me to understand how to a) customize voicemail . so that I can say "Hi this is Allan, not in right now . " . I have read that one can press 0 to get to the voicemail menu..but I cant figure at what point to press 0 to get to this menu. b) also what key s

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 12, Issue 79

Could someone please help me to understand how to a) customize voicemail . so that I can say "Hi this is Allan, not in right now . " . I have read that one can press 0 to get to the voicemail menu..but I cant figure at what point to press 0 to get to this menu. b) also what key s

[Asterisk-Users] Festival problems

I have installed festival a while ago and it could say "Mary had a little lamb" When I changed the text, it kept silence. Changed back the text and it worked. Now it does not say anything anymore!!! festival.conf: [general] usecache=yes cachedir=/var/cache/asterisk/festival/ festivalcommand=

Re: [Asterisk-Users] NO calling tone

Can you show me the example, i am newbie.NOt sure whether the code i modified is correct or not.. my code as below.. exten => 671042,1,Dial(${PHONES1},20,Ttmr) Cullin J. Wible wrote: Add the "r" parameter to the end of the Dial() statement. -Original Message- From: [EMAIL PROTECTE

[Asterisk-Users] Skip Announcement Confirmation in MeetMe

Anyone know how to bypass the CONFIRMATION of the user announcement recording in MeetMe? While I like the "please say your name" to announce a user into a conference, I find it confusing and time consuming to make the user to press 1 to accept a recording they haven't even previewed. I'm not

RE: [Asterisk-Users] NO calling tone

Add the "r" parameter to the end of the Dial() statement. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Wong Sent: Tuesday, July 12, 2005 10:38 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] NO calling tone Hi, When I make a cal

RE: [Asterisk-Users] Any suggestions for an IP phone?

We just purchased 4 of the Polycom SoundPoint 301's. We are very happy with them so far. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 12, 2005 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Disc

Re: [Asterisk-Users] ASTPP

I'm sorry, the forum was down. It is back. Lets move this to http://www.aleph-com.net/astpp/forum/ if that is okay with you. Darren Wiebe [EMAIL PROTECTED] Carlos Chavez wrote: Does anyone have experience setting up ASTPP? I have an Asterisk server in my office that I also give acc

[Asterisk-Users] NO calling tone

Hi, When I make a call by using sip phone or softphone, there is no calling sound, how do I get the calling sound ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIB

Re: [Asterisk-Users] Unable to call certain 800 numbers through Teliax

Cullin J. Wible wrote: We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Air

Re: [Asterisk-Users] Any suggestions for an IP phone?

I love the spa-3000 and related devices. I know they're not IP phones, but they turn any phone you want into one without sacrificing the features of either. Jon. On Tuesday 12 July 2005 11:16 am, Alexandre Leclerc wrote: > Hi all, > > We are in the process of selection IP Phones to work with our

Re: [Asterisk-Users] Last CVS -> High Load

> Rich Adamson wrote: > > June 24th was good here, but we're running a very basic config with no > > queues, etc. Yesterdays head does have an unknown failure/lockup that > > happens infrequently, so stay away from it for now. > > I believe all these problems are corrected now, the major MOH probl

RE: [Asterisk-Users] Cisco 79XX Jitter Stats Question

Actually I don't think it would matter if reinvite was on or not. The asterisk server in this case would simply be passing the RTP stream from and to the LD provider to the phone. Seeing the asterisk server is not generating the stream, the jitter will be end to end, because asterisk is simply ac

Re: [Asterisk-Users] Polycom 600 phone

We use the 500's and they are great. It has 3 lines, which is plenty. The resolution on the 600 screen is better, but unless you are using graphics on the LCD, it isn't worth the extra $$. We have a couple of 300's and the screen is ugly. It is small and hard to read. I would stick with the

Re: [Asterisk-Users] Cisco 79XX Jitter Stats Question

It depends on whether you have re-invite enabled. It is measuring the jitter of the rtp stream, so if re-invite is on, then it is measuring to the far end. If re-invite is off, then it is measuring to *. Matthew Boehm wrote: When on a call, you can press the middle round button and bring up s

Re: [Asterisk-Users] Unable to call certain 800 numbers through Teliax

> We are unable to call certain 800 numbers through Teliax but I thought I > would post this here and see if anyone else had the same problem with > either Teliax or other carriers. > > The 800 numbers causing problems pick-up the call right away and are in the > US - American Airlines (800433

Re: [Asterisk-Users] Last CVS -> High Load

> Rich Adamson wrote: > > June 24th was good here, but we're running a very basic config with no > > queues, etc. Yesterdays head does have an unknown failure/lockup that > > happens infrequently, so stay away from it for now. > > I believe all these problems are corrected now, the major MOH probl

[Asterisk-Users] AgentCallbackLogin Question

I'm using ver. 1.0.7 here are a couple of lines from my extensions.conf file: exten => x,1,AgentCallbackLogin(${CALLERIDNUM}|[EMAIL PROTECTED]) exten => x,2,Hangup I'm looking for a way to capture the Agent ID after login, to keep track which agent is associated in a certain call. __

[Asterisk-Users] Unable to call certain 800 numbers through Teliax

We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and

[Asterisk-Users] Little doubt on Asterisk and EyeBeam

Hi, Recently I been using EyeBeam with Asterisk. I can make calls with video, I'm using two PCs with EyeBeam but I noted that I can't enable only one client to stream video, I mean if I start streaming video on one client the other one doesn't receive any video until it starts streaming vide

[Asterisk-Users] Manger-command Getvar?

Hi, I'm trying to use the manager cmd Getvar. Unfortunately I always get (null) as variable content. I'm using asterisk 1.0.7 When calling a non existant channel, I get an appropriate result. This is what I tried and got: Action: Getvar Channel: SIP/01234567-5242 Variable: CALLERID Response:

Re: [Asterisk-Users] Polycom 600 phone

Just because the phone has the extra "lines" doesn't mean you are required to use them. Each "line" can handle 2 calls. The 600 has a working XML microbrowser, which the 50x does not. The Polycom 501 (not sure if the 500 is the same) doesn't have a place on the phone to plug power in. It gets

RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?

Hi Newbie, I wonder how you could find the mailing list but NOT the wiki at http://www.voip-info.org/tiki-index.php?page=Asterisk, that documents a huge area of how to use Asterisk in many scenarios. First of all, take a week or two to read the wiki and to set up a testing environment and try to

Re: [Asterisk-Users] Any suggestions for an IP phone?

Polycom SP300 is a pretty good phone. Rudolf > Alexandre Leclerc <[EMAIL PROTECTED]> wrote: > > Hi all, > > We are in the process of selection IP Phones to work with our *new* > Asterisk PBX. > > We want to buy 4 for something less than 1000$ but with a nice set of > features to work with ou

Re: [Asterisk-Users] Last CVS -> High Load

Rich Adamson wrote: June 24th was good here, but we're running a very basic config with no queues, etc. Yesterdays head does have an unknown failure/lockup that happens infrequently, so stay away from it for now. I believe all these problems are corrected now, the major MOH problem patch was r

Re: [Asterisk-Users] DELL 2800 : PCI Parity error

I had an occasional PCI parity error on a TE405p on an HP DL320. Turned out to be grease or some similar substance on the edge connector of the PCI riser to the mainboard in the server presumably from the manufacturing process. Had been bugging me for months until I finally tracked it down. Crai

Re: [Asterisk-Users] Having Trouble Creating an IVR

On Tue, Jul 12, 2005 at 02:23:39PM -0400, Tim P wrote: > I have asterisk 1.0.5 installed via apt on a debian system. It's a > custom distrobution called Voyage Linux that runs from a flash card > and I have a hard drive installed with mysql installed on it as well > as apache. /me wonders if a

[Asterisk-Users] Compile failure on Mac OS X Tiger

I seem to be having a problem compiling on MacOS X Tiger. The errors are as follows: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -DASTERISK_VERSION=\"1.0.9\" -DINSTALL_PREFIX=\"\" -DASTETCDIR

Re: [Asterisk-Users] Polycom 600 phone

Chris Gamble wrote: From their website, the key difference between the polycom 500 and 600 phones is the number of "lines" they support. What does this mean in terms of asterisk? Do I have to have a seperate extension for each of these lines or ? Also, slightly off-topic, how does the 500 POE "

Re: [Asterisk-Users] Having Trouble Creating an IVR

Maybe you are using a Sipura ATA? they use *XX to acticate/deactivate advanced functions (DND,Call Return Code:Blind Transfer Code,Call Back Act Code,DND Act Code) or Block ANC Act Code:*77 2005/7/12, Tim P <[EMAIL PROTECTED]>: > I have asterisk 1.0.5 installed via apt on a debian system. It's a

Re: [Asterisk-Users] how to debug perl agi

On Tue, Jul 12, 2005 at 05:42:37AM -0700, Kamran Ahmad wrote: > hello > > i am trying to develop perl application for asterisk > with radius accounting how can i debug that weather > callback is working when call is stoped. > > how can i check this > > syslog('info', 'hello Asterisk!'); That sh

RE: [Asterisk-Users] Last CVS -> High Load

Thank you! My setup is pretty basic too, just need solid IAX and SIP, some transcoding, voicemail and MOH. Finally jumping into realtime :) > -Original Message- > From: Rich Adamson [mailto:[EMAIL PROTECTED] > Sent: Tuesday, July 12, 2005 5:52 PM > To: Asterisk Users Mailing List - Non-

Re: [Asterisk-Users] How to avoid collect calls on ISDN BRI trunks?

If you get ANI2 then you should be able to determine that it is a collect call and just do busy or congestion. In any case you can ask your provider to not allow collect calls. On 7/12/05, Armin Schindler <[EMAIL PROTECTED]> wrote: > On Tue, 12 Jul 2005, Dhennys Pestana wrote: > > Hello, Armin! >

Re: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?

inline On 7/12/2005, "Ed Pastore" <[EMAIL PROTECTED]> wrote: >Hi, folks. I am planning on implementing Asterisk in 2006, and need >to budget for it now, so I need to know what I'll need to get. My >company has about 50 users, and is currently languishing on a very >old Comdial PBX. All of our cl

Re: [Asterisk-Users] Queues and busy agents problem

Asterisk doesn't have any magic way of knowing whether a SIP phone is busy or not; the "SIP way" is that you just call the phone (ie send an INVITE) and the phone then decides by itself what to do. It might send back a "Busy here" response - in which case Asterisk gets the message and tries someo

[Asterisk-Users] TDM400P FXO callprogress doesn't detect remote answer

Location = US asterisk/zaptel from CVS. Updated last week some time. Currently rebuilding with todays checkout. I have 2 fxo channels hooked up to outside standard Bell South phone lines. If I configure as so [channels] context=pstn group = 1 signalling = fxs_ks callprogress = yes channel =>

[Asterisk-Users] Polycom 600 phone

>From their website, the key difference between the polycom 500 and 600 phones >is the number of "lines" they support. What does this mean in terms of >asterisk? Do I have to have a seperate extension for each of these lines or ? Also, slightly off-topic, how does the 500 POE "optional" cable w

Re: [Asterisk-Users] Digium Wildcard TE110P IRQ problem

Accursio: Thanks for the tip, It worked for me too!!! Cheers, JR > This worked for me: > > before compile bristuff edit the file > wcte1xxp.c > > near line 1526 initialize the array pci_device_id t1xxp_pci_tbl[] > this way: > > static struct pci_device_id t1xxp_pci_tbl[] = { >{ 0xe159, 0x

[Asterisk-Users] H323 email address

on net-meeting first use you put your name and email on the config wizard. on net-meeting calls to asterisk, asterisk put the name on ${CALLER_ID} var, Is posible to obtain the emial address from a netmeeting H323 call?. -- David Romero## _

RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?

Hello and welcome... Most of what you want to know is available on the wiki located here... http://voip-info.org/tiki-index.php Just scroll down to the "All Things Voip" section. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Pastore

[Asterisk-Users] SNOM 360 and parking

OK, last showstopper that I just can't puzzle my way through - parking calls with the snom phones. I get the two phones connected, I hit transfer on one, the other phone goes to MOH and the first phone gives me DT, so I dial 700 and hit the OK button. Call transferred, the SNOM hangs up befor

Re: ***SPAM*** Re: [Asterisk-Users] Last CVS -> High Load

Steve Clark wrote: if (class->pseudofd > -1) { /* Pause some amount of time */ res = read(class->pseudofd, buf, sizeof(buf)); } else { This patch has been reverted until the problem can be solved. ___ Asterisk-

Re: [Asterisk-Users] Pushing new firmware to Snom 190

On Tuesday 12 Jul 2005 19:02, Patrick Friedel wrote: > Bob Goddard wrote: > >There are 2 problems here, the first is if you click on "memory" and > >the connection count is not 0, then you will be unable to reboot the > >phone, all you can do then is power cycle it. > > > >Secondly, to update the p

Re: [Asterisk-Users] Odd MOH problem...

There don't appear to be any running after asterisk dies, while asterisk is alive I have this: voip:/etc/asterisk# ps aux | grep mpg root 15785 0.0 0.2 2316 1032 pts/1S+ 16:55 0:00 /bin/sh /usr/local/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 -z [a whole load of the right f

RE: [Asterisk-Users] Cisco 7940/7960 interdigit timeout

I think the file you want to edit is the dialplan.xml. I don't remember the syntax off the top of my head, but I'm sure it is documented on the Cisco web site. B. J. -Original Message- From: Roland Zagler [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 12, 2005 11:22 To: asterisk-user

[Asterisk-Users] IAX2 ping confusion and unreachable soft phones

I've turned on debug in a (IAXComm based) soft phone. I see the phone sending pings to *. I see * getting the pings. For some reason, with iax2 show debug, I never see any response on the console from *. However, the phone shows a response with INVAL. Seems like an odd response to a ping request

[Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?

Hi, folks. I am planning on implementing Asterisk in 2006, and need to budget for it now, so I need to know what I'll need to get. My company has about 50 users, and is currently languishing on a very old Comdial PBX. All of our client computers are Macs; our servers are mostly OS X, with a

Re: [Asterisk-Users] Last CVS -> High Load

June 24th was good here, but we're running a very basic config with no queues, etc. Yesterdays head does have an unknown failure/lockup that happens infrequently, so stay away from it for now. > On that note... What's the latest pseudo-stable HEAD anyone is running > with

[Asterisk-Users] Cisco 79XX Jitter Stats Question

When on a call, you can press the middle round button and bring up some RTP statistics. Can anyone confirm my theory that the AvgJtr and MaxJtr are between this phone and the far end? Or is this jitter reading only between this phone and asterisk? I'm guessing its the foremost, because when

Re: [Asterisk-Users] Odd MOH problem...

When you restarted Asterisk, did you kill the mpg123 processes? -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer. This is a personal e-mail and as such, the opinions expressed are my own. On Jul 12, 2005, at 11:52, Patric

[Asterisk-Users] help needed-call recording

Hi,   I am trying to change the dialplan to enable call recording (incoming and outgoing calls) on the “click of a button”. Is it possible? All the documentation I found so far, enable recording for ‘all calls’ to an extension.   Does this code look ok? Currently Recording “on” only f

RE: [Asterisk-Users] Pushing new firmware to Snom 190 <--solved

Found out I was doing it right, but the outbound firewalls that the Snom's sits behind has a captive portal that intercepted the HTTP request. Adding the Snom's MAC address to the firewall's captive portal worked correctly. A shortcut for any future Snom users that want to update firmware remotely:

Re: [Asterisk-Users] bristuff patches and realtime mysql

On Tue, Jul 12, 2005 at 05:07:20PM +0200, Christoph wrote: > On Tue, 2005-07-12 at 09:42 -0500, Matthew Boehm wrote: > > How do you expect me to possibly fix my module if you don't supply any > > compile errors? I don't use BRI so you will need to provide me alot of > > info. Which bri package di

Re: [Asterisk-Users] Last CVS -> High Load

Hi, Didn´t check the high CPU load but I my Asterisk´s SIP channel stop responding after upgrade to yesterday CVS. I upgraded to today´s CVS and MoH stopped working :( On 7/12/05, Gentian Bajraktari <[EMAIL PROTECTED]> wrote: > > Yes I have experienced the same on my test machine. It has b

Re: [Asterisk-Users] Last CVS -> High Load

On that note... What's the latest pseudo-stable HEAD anyone is running with SIP? > -Original Message- > From: Steve Clark [mailto:[EMAIL PROTECTED] > Sent: Tuesday, July 12, 2005 12:36 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: ***SPAM*** Re: [Asterisk

Re: [Asterisk-Users] Asterisk and Dell SC420 Server

Works for me and Pulver sells em. http://voipstore.pulver.com/product_info.php?products_id=57 - Original Message - From: <[EMAIL PROTECTED]> To: Sent: Tuesday, July 12, 2005 10:01 AM Subject: [Asterisk-Users] Asterisk and Dell SC420 Server > Has anyone attempted or have had any issues

RE: [Asterisk-Users] Cisco 7940/7960 interdigit timeout

Thanks, i added dialplan_template: "dialplan" to SIPDefault.cnf and the lines you sent to dialplan.xml in TFTP-directory and it works! Thanks again, Roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Wellsted Sent: Tuesday, July 12, 2005 7:08

[Asterisk-Users] Cisco SIP Frimware for 7940/7960 v7.5

Hello list, is there anyone out there that could grab the new SIP firmware 7.5 for the 7940/7960 from Cisco's Site and mail it to me ([EMAIL PROTECTED])? i already ordered a support contract but did not get my access data yet! Thanks, Roland ___ Aster

Re: [Asterisk-Users] Howto get streaming mp3 at an extension?

> stream => /var/lib/asterisk/stream,http://sourcepfstream.com:8001/ > > Then add an externsion number to extensions.conf that uses the stream > variable to play the hold music. > > There's quite a bit about this in the wiki. The stream I'm trying to listen to has a filename in the URL ... http:

RE: [Asterisk-Users] Zaptel won't compile under Fedora Core 4

Which version of Zaptel are you using?  I am using version 1.09 and having issues,  I did not have issues with 1.08. -Original Message-From: Eric Bullen [mailto:[EMAIL PROTECTED]Sent: Tuesday, July 12, 2005 12:36 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion

[Asterisk-Users] Re: Asterisk and Dell SC420 Server

Hi Jyran - Has anyone attempted or have had any issues getting Asterisk and Digium cards working on a Dell SC420 server? Yes. One of our offices has an SC420 running Tao Linux and Asterisk (CVS HEAD), using a Digium TDM400 (TDM04B). It works pretty well, for the most part. Just so yo

[Asterisk-Users] Having Trouble Creating an IVR

I have asterisk 1.0.5 installed via apt on a debian system. It's a custom distrobution called Voyage Linux that runs from a flash card and I have a hard drive installed with mysql installed on it as well as apache. I have been though the AMP install guide (asterisk management portal) and in the i

[Asterisk-Users] Returning values from macro inside Dial command

I'd like to set some variables inside the macro called from Dial(). As far as I can tell, though, if you just do a SetVar inside the macro, it's set in a different context and isn't available to the code that calls the macro. Am I missing something? What's the right way to return data from the m

Re: [Asterisk-Users] Pushing new firmware to Snom 190

Bob Goddard wrote: There are 2 problems here, the first is if you click on "memory" and the connection count is not 0, then you will be unable to reboot the phone, all you can do then is power cycle it. Secondly, to update the phone, you have to create 2 files, the first is entered into the "Se

[Asterisk-Users] ASTPP

Does anyone have experience setting up ASTPP? I have an Asterisk server in my office that I also give access to some friends and family that live outside Mexico so they can make local calls. I want to keep track of the costs and I only need to use ASTPP to rate the calls, not for calling

[Asterisk-Users] Odd MOH problem...

So I decided, for the formal asterisk rollout, to change over to less commercially-infringing MOH than the prior admin had thrown on the server. (plus: it was blown out and nasty sounding over the phones. Ew.) I changed the files in /var/lib/asterisk/mohmp3 to something else (can't dig up th

Re: [Asterisk-Users] Looking for a consutant in France about Asterisk.

Hi all the list, Bonsoir,   It could be nice to give a correct email adress to contact you, instead of an "over quota" one... If you want a direct reply out of this list, please, give us one other.   Best Regards, a bientot, Francois BERGERET, France. [EMAIL PROTECTED] - Original Messa

Re: [Asterisk-Users] Zaptel won't compile under Fedora Core 4

Eric Bullen wrote: I hope someone can offer me some help with this. Basically, the current CVS version of Zaptel will not compile under Fedora Core 4. I have closely followed the directions in http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 using the versions given in the

Re: [Asterisk-Users] How to avoid collect calls on ISDN BRI trunks?

That's exactaly the idea! Make it work like E1 trunks with R2 signalling (Brazil) and do 'double-answering'. If it's not possible to do with CAPI channels, I'll have a lot of trouble to change all my AVM cards for HFC (or Intel537) ones, so I can use ZAP... :/ -Dhennys - Original Message

RE: [Asterisk-Users] Asterisk and Dell SC420 Server

[EMAIL PROTECTED] wrote: > Has anyone attempted or have had any issues getting Asterisk and > Digium cards working on a Dell SC420 server? I have it up and running on 2 SC420's with TE110P cards. Haven't had any problems as of yet (knock on wood). Dell SC420 TE110P 2x40 GB SATA drive in a RAID 1

Re: [Asterisk-Users] Any suggestions for an IP phone?

You can go with the GXP-2000 from Grandstream. It is shaping up as a very good phone. It usually retails for around $114 USD but you can find it a bit cheaper if you really look around. Two questions: Are the buttons rubbery, and is the display dim? I don't have the opportunity to

Re: [Asterisk-Users] Help Configuring TDM04B

Is it a revision I board? - Original Message - From: "chawki hammoud" <[EMAIL PROTECTED]> To: Sent: Tuesday, July 12, 2005 9:44 AM Subject: [Asterisk-Users] Help Configuring TDM04B Hi: I had an fxo card from Digitnetworks and it was working fine on my Asterisk box. I then replaced

Re: [Asterisk-Users] PRI problem

On Tue, Jul 12, 2005 at 10:59:42PM +0800, matt001 wrote: > currently we are able to use our USA sip phone to conenct into the E1 box, > but still unable to dial out to chinese phone numbers. They said from their > ISDN switch console, it shows D channel not connected to the voip server yet. > >

Re: [Asterisk-Users] Help Configuring TDM04B

Chawki - If you don't find help on the listserv, you can purchase 1 (60) minute Tech Support package from us for $79.95 and one of our in-house engineers will assist you. Does not matter that you purchased your Digium board from another reseller. (60) Minute Engineering Support Package http://

Re: [Asterisk-Users] How to avoid collect calls on ISDN BRI trunks?

On Tue, 12 Jul 2005, Dhennys Pestana wrote: > Hello, Armin! > Flash() should issue a flash for a few milliseconds, just like flash button on > some analog phones. > As far as I know (based on Google and AsteriskBrasil.org search), it works > only > on ZAP channels for now. > I found out Flash() ru

Re: [Asterisk-Users] Help Configuring TDM04B

I ha the same problem, could be irq issue if you have zaptel.conf and zapata.conf configured properly. modprobe ??? had been loaded? check dmesg check irq's cat /proc/interrupts lspci -vv look for the card information JR > Hi: > > I had an fxo card from Digitnetworks and it was > working fine on m

Re: [Asterisk-Users] How to avoid collect calls on ISDN BRI trunks?

Hello, Armin! Flash() should issue a flash for a few milliseconds, just like flash button on some analog phones. As far as I know (based on Google and AsteriskBrasil.org search), it works only on ZAP channels for now. I found out Flash() runs smoothly on analog channels when receiving calls with FX

RE: [Asterisk-Users] Any suggestions for an IP phone?

try the cisco 7940 with sip firmware: tons of features and easy to install see http://www.voip-info.org/tiki-index.php?page=cisco%2079xx regards, roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexandre Leclerc Sent: Tuesday, July 12, 2005 6:16

Re: ***SPAM*** Re: [Asterisk-Users] Last CVS -> High Load

Gentian Bajraktari wrote: Yes I have experienced the same on my test machine. It has been like this for 3 weeks of CVS Head. Someone must have a look at that, I think is the SIP channel. - Original Message - *From:* Thierry Wehr *To:* asterisk

Re: [Asterisk-Users] Zaptel won't compile under Fedora Core 4

On 7/11/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: On Tue, Jul 12, 2005 at 11:04:16AM +1000, Gonzalo Servat wrote:> On 7/12/05, Eric Bullen <[EMAIL PROTECTED]> wrote:> > I hope someone can offer me some help with this. Basically, the current CVS > > version of Zaptel will not compile under Fedora

Re: [Asterisk-Users] Help: TE100P connecting to non PRI, ISDN interfaces

On Tue, 2005-07-12 at 20:00 +0800, ian sison (mailing list) wrote: > Hello, i've googled and can't find a definite answer, so here goes: > > I have purchased the Digium TE100P, and am setting up the connection, > however the > telco i'm supposed to work with does not support PRI/ISDN E1 > connecti

[Asterisk-Users] FYI: BT Caller ID.

Hi, Just incase anyone hasn't heard, BT are now offering a service called BT Privacy which includes caller ID for free. So if you are currently a residential customer who is paying for caller id, you may as well save a bit of money and check out BT's web site! Thanks, Darren.

Re: [Asterisk-Users] Re: DTMF not sending properly via IAX

On 12 Jul 2005, at 15:05, Tony Mountifield wrote:In article <[EMAIL PROTECTED]>,Mark Edwards <[EMAIL PROTECTED]> wrote: I'd be interested to understand where you read about the 'sequence numbers' issue. This sounds like it might relate to the problem I am experiencing. It's either that, or the DTMF

[Asterisk-Users] Network Configuration Question for Asterisk Server

I am looking for some advice on how to best setup the Asterisk Server on the network for access from remote SIP phones. The following is my current configuration: SBC DSL (5 Static IP Available) ---> Netgear Firewall/Router ---> Asterisk Server & Local SIP phones on LAN. The Netgear router is cur

[Asterisk-Users] Asterisk and Dell SC420 Server

Has anyone attempted or have had any issues getting Asterisk and Digium cards working on a Dell SC420 server? Thank you, Jyran Glucky Advisory Programmer BlueWare, Inc. Strategic HealthWare Solutions 3060 W. 13th Street Cadillac, MI 49601 Phone: (231) 779-0224 ext. 111 Fax: 231-779-1002 mailto:

Re: [Asterisk-Users] Any suggestions for an IP phone?

On Tue, 2005-07-12 at 12:16 -0400, Alexandre Leclerc wrote: > Hi all, > > We are in the process of selection IP Phones to work with our *new* > Asterisk PBX. > > We want to buy 4 for something less than 1000$ but with a nice set of > features to work with our mail box, lines, good sound quality,

RE: [Asterisk-Users] Problem with DTFM and complex international setup

I don't think so.   Your problem seems to do with your not being able to use an IAX client to transmit DTMF tones properly somehow.   I am using a normal phone to connected to FWD which then connects to an Asterisk server using IAX protocol. The point is that between the phone and the far Aste

Re: [Asterisk-Users] Cisco 7940/7960 interdigit timeout

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tue, 12 Jul 2005, Roland Zagler wrote: Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in

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