[asterisk-users] Req-Installation process for app_dtmftotext.c

Hi, I was looking for a way to pass alphanumeric variables to asterisk via the keypad, found this application app_dtmftotext.c , And I already tried with 'spandsp' application for this. But I am getting errors.[I followed the instructions at http://www.soft-switch.org/installing-spandsp.html] s

Re: [asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?

On Tue, 15 May 2007, Vincent Delporte wrote: > Hello, > > In case there are other users of the AsteriskWin32 port... > > I haven't really used the AGI feature of Asterisk to run an application > from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, > it's also possible t

Re: [asterisk-users] queue_exec: Unable to join queue

lenz wrote: > Is the queue "enidan" configured at all in queues.conf? and how is it > defined? > l. Sorry, I should have added that: from queues.conf: [enidan] strategy = ringall ;announce = enidan-queue member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PRO

Re: [asterisk-users] The purpose of DUNDi

dave cantera wrote: > remco, et al, > could I use dundi where I could use an area code to determine the > connecting server or dial string? just like we would use 88XXX to dial > a 3 digit extension on another server at location 88? or dial 84XXX for > a 3 digit extension on a server located at 8

Re: [asterisk-users] Re: TC400B load problem

thanks Matthew, I'll try to call Digium. On 5/14/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote: On May 14, 2007, at 4:53 AM, Arun Kumar wrote: > Im trying to install my TC400B trans coder card when I do: > > modprobe wctc4xxp > > tail -f /var/log/messages says: > > May 13 14:56:36 pbx2 ker

Re: [asterisk-users] Simultaneous Capacity

On Mon, 14 May 2007, Kapil Dhawan wrote: > Just a quick brief > > I have a requirement of running 10 PRI's (300 Channels). I still have to > decide on hardware and cards. Can you suggest some. As per my understanding it > will be tough to go beyond 150. I didn't test exactly this yet, but from my

[asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?

Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe

[asterisk-users] How to write data to astdb?

Hello, I'm trying to fill CID data into the astdb using AsteriskWin32's asterisk.exe, to no avail: The batch file stops after the first line, and just waits: rem c:\cygroot\mystuff>import.bat rem rem c:\cygroot\mystuff>C:\cygroot\bin\asterisk.exe -rx

Re: [asterisk-users] Web based call control

On Mon, 14 May 2007, Jordan Novak wrote: > Does anyone know if it is possible to use a manager command to answer > an incoming call and not consider it answered unitl it is received. > Here is an example, I am deivering a call in the dialplan to a home > telephone number. I don't want his voicema

Re: [asterisk-users] OT (semi) E60 problem

On May 14, 2007, at 12:34 PM, Tim Panton wrote: On 14 May 2007, at 17:50, Martin Joseph wrote: Hello again gurus. I have been using Asterisk with great results going on a couple of years now. My primary box is running asterisk 1.42 built from a tar ball on Mac OSX 10.4.9. I have a v

[asterisk-users] `PATH_MAX' undeclared here (not in a function) in asterisk!

hello, asteriskers: I compiled asterisk under arm-linux. i am using asterisk 1.4.2. i can run ./configure and menuselect with embedded modules. but running make comes out errors: ranlib libmxml.a make[3]: Leaving directory `/usr/src/asterisk-1.4.2/menuselect/mxml' cc -Wall -o menuselect.o -g -c

[asterisk-users] Web based call control

Does anyone know if it is possible to use a manager command to answer an incoming call and not consider it answered unitl it is received. Here is an example, I am deivering a call in the dialplan to a home telephone number. I don't want his voicemail to answer and I have no idea how long it will

Re: [asterisk-users] Asterisk High-Capacity Stability

I'm curious what kind of configuration/features/modules you could recommend for my setup. Can you explain further what you mean by OpenSER to Asterisk? Thanks Much, kn0x On 5/14/07, EdPimentl <[EMAIL PROTECTED]> wrote: Actually, OpenSER is just the you will need to scale Asterisk. We have per

RE: [asterisk-users] Proper AGI use with MySQL

We are not very strong perl programmers and we've been using the Asterisk::AGI library. Is there any sample you can point us that shows how to do this? Thanks On Mon, May 14, 2007 9:05 pm, Michelle Dupuis <[EMAIL PROTECTED]> said: > How about forking the process when the AGI launches, and pass

Re: [asterisk-users] Blind Transfer - Who transferred the call?

Lee Jenkins wrote: Hi all, Is there a way to tell which extension transferred a call in a blind transfer? Sorry if it's a basic question, but I haven't seen an answer. ${CALLERID(num)} still holds the outside party caller id (which it should), but I'd like to the extension number of the ex

RE: [asterisk-users] Proper AGI use with MySQL

How about forking the process when the AGI launches, and pass the PID back to Asterisk in a variable. When the call ends (caught at the "h"), call another AGI script to kill/stop that pid. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]

[asterisk-users] Blind Transfer - Who transferred the call?

Hi all, Is there a way to tell which extension transferred a call in a blind transfer? Sorry if it's a basic question, but I haven't seen an answer. ${CALLERID(num)} still holds the outside party caller id (which it should), but I'd like to the extension number of the extension that transf

[asterisk-users] Asterisk Now

Can someone tell me what is included in this distro? Does it have voicemail, meetme, panel, and IVR? Thanks, Wiley E. Siler Director of Information Technology 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED]

Re: [asterisk-users] ast_yyerror - Help

On Mon, 2007-05-14 at 14:52 -0500, Rob Schall wrote: > Hey all, > > We're starting to see "all circuits are busy" and a few dropped calls. > When these happen, in the messages log, I see the following error. > > May 14 14:42:13 WARNING[5604] ast_expr2.fl: ast_yyerror(): syntax error: > syntax err

Re: [asterisk-users] Re: dialplan: execute on hangup

thx a lot Tony, I didn't know about using the h-extension (I'm new to Asterisk)! this way it works: ... exten => s,n,Voicemail(${Enter},u) exten => s,n,AGI(foneboxx.php|${Enter}) exten => h,1,DeadAGI(foneboxx.php|${Enter}) greetings, michael On 5/14/07, Tony Mountifield <[EMAIL PROTECTED]> w

[asterisk-users] Re: dialplan: execute on hangup

In article <[EMAIL PROTECTED]>, Michael Kamleitner <[EMAIL PROTECTED]> wrote: > > thx Tony, but DeadAGI doesn't seem to fit my needs... the way I understand > its functioniality ( > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI), DeadAGI > is ensureing that an executed AGI-scri

RE: [asterisk-users] Proper AGI use with MySQL

Sorry, just to make sure this is clear, in #2 below, when I said "We would like for the AGI script to stay running for the life of the call...", I also meant after the call is transfered to the customer service queue. This is so because we need to note that the call ended (update callend = NOW()

[asterisk-users] Proper AGI use with MySQL

Hi, We have a "simple" AGI script that provides some IVR functionality. It connects to a MySQL database in order to create a call record and capture IVR data. During the IVR process, we need to store the time the call started, so basically we INSERT a new MySQL row with callstart = NOW(), uniqu

Re: [asterisk-users] The purpose of DUNDi

remco, et al, could I use dundi where I could use an area code to determine the connecting server or dial string? just like we would use 88XXX to dial a 3 digit extension on another server at location 88? or dial 84XXX for a 3 digit extension on a server located at 84?... thanks, daveC Rem

Re: [asterisk-users] Asterisk High-Capacity Stability

Actually, OpenSER is just the you will need to scale Asterisk. We have perform a number of OpenSER to Asterisk implementation for 50k plus users -E On 5/14/07, Atlanticnynex <[EMAIL PROTECTED]> wrote: Thanks for all the input guys. This is what I had originally expected. Does anyone have an

Re: [asterisk-users] zaptel huge irq problem

You didn't even read the thread before replying. And for what it is worth, we at Digium are very anxious to solve any sort of IRQ problems that you (or others) might have. Matthew Fredrickson On May 14, 2007, at 1:43 PM, Salvatore Giudice wrote: Try switching to a Sangoma card. You won’t hav

Re: [asterisk-users] queue_exec: Unable to join queue

Is the queue "enidan" configured at all in queues.conf? and how is it defined? l. In data Mon, 14 May 2007 13:56:25 +0200, Per Jessen <[EMAIL PROTECTED]> ha scritto: I have a queue defined which I use like this: exten = _X.(reception),n,Ringing exten = _X.,n,Queue(enidan,t,,,20) exten =

Re: [asterisk-users] Asterisk High-Capacity Stability

Thanks for all the input guys. This is what I had originally expected. Does anyone have any recommendations for other software configurations? I've thought about using OpenSER + rtpproxy(or media proxy), but it seems that OpenSER is not designed to do this sort of thing and would require some tric

[asterisk-users] Junghanns DuoBRI Card HELP !

We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We keep on getting the error "layer 1 deactivated (F3)"! The card sees no ISDN device connected to it, neither in NT or TE modes alike. We then contacted Junghanns, who told us that there is no driver for the card for 1.4 and

[asterisk-users] ast_yyerror - Help

Hey all, We're starting to see "all circuits are busy" and a few dropped calls. When these happen, in the messages log, I see the following error. May 14 14:42:13 WARNING[5604] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or TOK_LP or

[asterisk-users] Junghanns DuoBRI Card HELP !

I really need help. We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We keep on getting the error "layer 1 deactivated (F3)"! We then contacted Junghanns, who told us that there is no driver for the card for 1.4 and that we should try 1.2. We tried 1.2 with their driver b

[asterisk-users] Junghanns DuoBRI Card HELP !

I really need help. We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We keep on getting the error "layer 1 deactivated (F3)"! We then contacted Junghanns, who told us that there is no driver for the card for 1.4 and that we should try 1.2. We tried 1.2 with their driver b

[asterisk-users] Some problems with mysql CDR

Hello, We have finally upgraded to Asterisk 1.4, however we've run into two issues that weren't occurring before the upgrade. Issue #1: We're an outgoing call center and need to record all calls. We use the uniqueid field in the CDR to match with the recording, which we labeled with {UNIQUEID

Re: [asterisk-users] OT (semi) E60 problem

On 14 May 2007, at 17:50, Martin Joseph wrote: Hello again gurus. I have been using Asterisk with great results going on a couple of years now. My primary box is running asterisk 1.42 built from a tar ball on Mac OSX 10.4.9. I have a very odd issue that I cannot seem to nail down, whic

Re: [asterisk-users] Re: Remote extensions not working on provider's wireless Internet connection

Actually now I am getting so many other weird problems. First of all, choppy sound on the receiving end on the test server. I don't understand why all of a sudden voice will go choppy, when bandwidth and Internet upload and download speeds are good. On the production server, it registers but won'

Re: [asterisk-users] Asterisk High-Capacity Stability

On May 14, 2007, at 1:29 PM, Zoa wrote: Several people do use it for handling > 50k minutes a day. (I'm one of them). Yes, you need to know what you are doing, and have a nice design, but it is possible.Our code is only slightly altered. (mainly for billing purposes). That's great if y

Re: [asterisk-users] Simultaneous Capacity

I have a requirement of running 10 PRI's (300 Channels). I still have to decide on hardware and cards. Can you suggest some. As per my understanding it will be tough to go beyond 150. Alex Balashov wrote: > On Mon, 14 May 2007, Kapil Dhawan said something to this effect: > >> I want to try Asteri

RE: [asterisk-users] zaptel huge irq problem

Try switching to a Sangoma card. You won’t have anymore IRQ issues once you abandon Digium hardware. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 891

Re: [asterisk-users] Dry Copper Pair

Stephen i disagree. growing up in new work city i can say its quite easy to get away with it in the city. where i live now in new jersey (population of around 6) i wouldnt be able to pull that off. >>> The world is a big place, and I suppose there's room for all kinds. >>> In th

[asterisk-users] IAX2 peer unreachable in one direction - NAT problem?

The situation is one of my asterisk servers is behind a NAT firewall and one is not. Both servers have multiple IAX peers. The NAT firewall has port 4569 mapped through to the asterisk server behind. But, the natted server is almost permanently unreachable from this non-natted server, even though,

Re: [asterisk-users] Asterisk High-Capacity Stability

Daryl Jurbala wrote: There is some light IVR type usage for reporting account balances and the like. With anything more than 80 or 90 calls on the box, the IVR prompts start to break up. Ben through replacing hardware, more memory, different Asterisk builds, etc. Zoa wrote: Several people d

Re: [asterisk-users] Asterisk and unicall + mfcr2 signalling

try using testcall with 255 as debug level and report back results in order to be able to help you. http://www.moythreads.com/unicall/mfcr2-asterisk-unicall-0.2-english.pdf On 5/14/07, Joca Loco <[EMAIL PROTECTED]> wrote: Hi, I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210

Re: [asterisk-users] zaptel huge irq problem

Stephen Bosch wrote: >> # cat /proc/interrupts >> CPU0 CPU1 >> 1: 1626 0Phys-irq i8042 >> 6: 3 0Phys-irq floppy >> 8: 0 0Phys-irq rtc >> 9: 0 0Phys-irq acpi >> 14:

[asterisk-users] Areski CDR

Hi folks, I was wondering what happened to Areski CDR viewer that came before with Freepbx. I've noticed that the live-CD contains Areski but the repositories don't have it. Will you fix that? or shall I install Areski from sources? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Tel

Re: [asterisk-users] How to bring MoH volume down

Alex Balashov wrote: > On Mon, 14 May 2007, Stephen Bosch said something to this effect: > >> Is there a way to do it for voice mail messages? I have a user who has >> trouble hearing the voice messages, saying they are too quiet. > > From a cursory glance at the voicemail settings, I can't see

Re: [asterisk-users] zaptel huge irq problem

François Delawarde wrote: > Thanks Michael, > > I've already been through all that unfortunately, and I have a SATA > drive, so no UDMA mode 2 as far as I know. I'm currently trying > everything again anyway, but i doubt it will work if nothing worked the > first time. > > Anyone would know of is

Re: [asterisk-users] DTMF not recognizing *

On Mon, 14 May 2007, Rob Schall said something to this effect: The problem is with having a "send to voicemail" option. Right now, a user can press "*5053" and they will be sent directly to that user's voicemail box, rather than their phone. But when you press "2*5053", it appears the * is ignor

Re: [asterisk-users] Double DTMF digits

On Sun, 2007-05-13 at 20:54 +0300, Dovid B wrote: > I am actually getting DTMF over SIP when people call in to a clients system > that is running a2billing. They are using RFC2833. > If you are using a Cisco router anywhere in the loop, there is a known bug that causes rfc2833 and inband signall

Re: [asterisk-users] Dry Copper Pair

Stephen Bosch wrote: >Per Jessen wrote: > > >>Jon Pounder wrote: >> >> >> >>>Quoting Stephen Bosch <[EMAIL PROTECTED]>: >>> >>> >>> C F wrote: >Stephen i disagree. growing up in new work city i can say its quite >easy to get away with it in the city. where

Re: [asterisk-users] 'Invalid characters in name' with asterisk-gui

This belongs in the asterisk-gui mailing list. However, I will see what I can do. -bkruse FYI. It is just a javascript pattern matching function, its super easy to change. Tom Lobato wrote: Hi all! Is there a way to asterisk-gui to allow underline (as such cpd_tom) in Names? It

Re: [asterisk-users] RTP Mixer

Perfect Josh...but if i am running an application which has a capability of showing number or participants depending upon CC value, that doesn't work. Secondly, Asterisk can show on CLI about current "talking" users where it is maintaining talking status but not sending it down the line to be u

Re: [asterisk-users] How to bring MoH volume down

Is there a way to do it for voice mail messages? I have a user who has trouble hearing the voice messages, saying they are too quiet. Just one user? Sounds like a user problem... however, with that said, you can try increasing your zaptel volumes. ___

Re: [asterisk-users] Asterisk High-Capacity Stability

Several people do use it for handling > 50k minutes a day. (I'm one of them). Yes, you need to know what you are doing, and have a nice design, but it is possible.Our code is only slightly altered. (mainly for billing purposes). Zoa Daryl Jurbala wrote: On May 12, 2007, at 4:11 PM, Atlanti

Re: [asterisk-users] Simultaneous Capacity

Just a quick brief I have a requirement of running 10 PRI's (300 Channels). I still have to decide on hardware and cards. Can you suggest some. As per my understanding it will be tough to go beyond 150. Alex Balashov wrote: On Mon, 14 May 2007, Kapil Dhawan said something to this effect: I

Re: [asterisk-users] How to bring MoH volume down

On Mon, 14 May 2007, Stephen Bosch said something to this effect: Is there a way to do it for voice mail messages? I have a user who has trouble hearing the voice messages, saying they are too quiet. From a cursory glance at the voicemail settings, I can't see a way. The voicemail messages a

[asterisk-users] DTMF not recognizing *

With our current setup, we have an older avaya system which is linked with our asterisk system via a em wink connection. When you press "2" on the avaya network, it will jump to our asterisk box and then sends DTMF digits. Asterisk listens for those numbers and then responses as soon as it has a ma

Re: [asterisk-users] zaptel huge irq problem

Hi, Francois: François Delawarde wrote: > Hello, > > I had noticed strange crackling sound on my phone calls going through my > zaptel device (TDM400P), so i decided to check on possible timer issue, > and found lots of issues on forums concerning the sensibility of zaptel > with IRQs, and tried

Re: [asterisk-users] Re: TC400B load problem

On May 14, 2007, at 4:53 AM, Arun Kumar wrote: Im trying to install my TC400B trans coder card  when  I do: modprobe wctc4xxp tail -f /var/log/messages  says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) May 13 14:5

[asterisk-users] OT (semi) E60 problem

Hello again gurus. I have been using Asterisk with great results going on a couple of years now. My primary box is running asterisk 1.42 built from a tar ball on Mac OSX 10.4.9. I have a very odd issue that I cannot seem to nail down, which is related to my Nokia E60 SIP phone. I use

Re: [asterisk-users] Re: CITEL gateway does it work well?

Steven wrote: > The Citel Handset Gateways were the best option for our scenario. > > The cost per port for the number of buttons on our NEC DTerm/E phones > was about half. > Also, no network reengineering. I've noticed that all the people who have good things to say about them are using East A

Re: [asterisk-users] zaptel huge irq problem

Thanks Michael, I've already been through all that unfortunately, and I have a SATA drive, so no UDMA mode 2 as far as I know. I'm currently trying everything again anyway, but i doubt it will work if nothing worked the first time. Anyone would know of issues with XEN or SMP (or both) kernel

Re: [asterisk-users] How to bring MoH volume down

Alex Balashov wrote: > > Zeeshan, > > On Mon, 14 May 2007, Zeeshan Zakaria said something to this effect: > >> MoH volume is uncomfortably high and I want to bring it down. Its >> mpg123. How can I do it? > > There are some settings in musiconhold.conf that may yield the desired > effect: >

Re: [asterisk-users] RTP Mixer

Kapil Dhawan wrote: I was reading an article on RTP Mixer so started studying about the mixing done by Asterisk in MeetMe. Read that CC should contain the no of participants ifupto 15 and CSRC should come, but not getting any by asterisk. I'll just leave it at this so we can all move on w

Re: [asterisk-users] Dry Copper Pair

Per Jessen wrote: > Jon Pounder wrote: > >> Quoting Stephen Bosch <[EMAIL PROTECTED]>: >> >>> C F wrote: Stephen i disagree. growing up in new work city i can say its quite easy to get away with it in the city. where i live now in new jersey (population of around 6) i wouldnt be

RE: [asterisk-users] Call to Skype network

Here's my instructions...based off Tim Hunt's great script...needs cleanup but the gist is hear to get someone going...you may think I'm reboot happy as there's more than a couple here but past experience found that reloads didn't do it...reboot seem to get things going...probably something simple.

Re: [asterisk-users] Simultaneous Capacity

On Mon, 14 May 2007, Kapil Dhawan said something to this effect: I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is it feasible. In truth, it is very unlikely. How are you planning to pick up the PRIs, anyway? 3 quad-span T1 cards? -- Alex Balashov <[EMAIL PROTE

[asterisk-users] Difference between making a call and Originate

When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=>"SIP/[EMAIL PROTECTED]", 'Context'=>'mycontext', 'Exten'=>'899', 'Priority'=>1, 'Callerid'=>'whatever')); It creates a screech sound when the fir

Re: [asterisk-users] How obtain the slot position when a call is parked?

On Monday 14 May 2007 10:41 am, [EMAIL PROTECTED] wrote: > I want to ask you if asterisk, when I use the command park(), gives me for > example a variable that contains the slot position where it parks the call > or if it only tells me (audio) in the channel this position number? In > other words,

[asterisk-users] Simultaneous Capacity

Hi List I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is it feasible. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

Re: [asterisk-users] RTP Mixer

I was reading an article on RTP Mixer so started studying about the mixing done by Asterisk in MeetMe. Read that CC should contain the no of participants ifupto 15 and CSRC should come, but not getting any by asterisk. Tzafrir Cohen wrote: On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil Dha

RE: [asterisk-users] zaptel huge irq problem

François, I too had a similar problem and found the information on this page helpful: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting What ended up working for me was changing the UDMA to mode 2 for the hard drive. Once I did that, this card has worked perfectly for me. Mich

RE: [asterisk-users] OT ? Number portability, land line to Cell

I think Joe's analysis is unreasonably negative regarding the landline companies' willingness to port. The link he provides, http://www.fcc.gov/cgb/consumerfacts/numbport.html, reflects my experience. A couple cautions, however: Landline companies may take two to three weeks to actually complete

[asterisk-users] How is Context Determined when Transferring a Call?

When trasferring a call, how is the context determined? When using a zap device, and the DTMF code for blind or attended transfer is entered, does the tranfer originate at the context the zap device is set to be in, or does it originate from where the outside call being transferred originated in,

Re: [asterisk-users] Asterisk High-Capacity Stability

On Mon, 14 May 2007, Daryl Jurbala said something to this effect: That being said, I don't meant to trash Asterisk at all. It's a fantastic feature server, and a great PBX, both of which things I use it for very successfully. Agreed. And, it's worth pointing out, that's what Asterisk is i

[asterisk-users] ChanSpy

Hi Guys, Does anyone know if is it possible to put one channel in two different spygroups? Thanks! Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digi

Re: [asterisk-users] OT ? Number portability, land line to Cell

> Having had various issues with local vendor (begins with "V"). am looking to > move to all wireless. Anyone know if current vendor can refuse to port the > current land line numbers to a wireless provider? > > >From what I've read, the Fed's seem to say "no", they cannot refuse, or > >impede

Re: [asterisk-users] How to bring MoH volume down

Please provide us with your config in musiconhold.conf so I/we can see how you are streaming. There may be a way to lower the volume, but it depends on how you are performing the streaming. On 5/14/07, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote: Here the problem is that it is streaming audio f

[asterisk-users] How obtain the slot position when a call is parked?

Hi, I want to ask you if asterisk, when I use the command park(), gives me for example a variable that contains the slot position where it parks the call or if it only tells me (audio) in the channel this position number? In other words, is there a way to obtain and use the value of the slot positi

Re: [asterisk-users] Asterisk High-Capacity Stability

On May 12, 2007, at 4:11 PM, Atlanticnynex wrote: Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of c

Re: [asterisk-users] OT ? Number portability, land line to Cell

Joe acquisto wrote: Having had various issues with local vendor (begins with "V"). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? >From what I've read, the Fed's seem to say "no", they cannot refuse, o

Re: [asterisk-users] Re: Remote extensions not working on provider's wireless Internet connection

Hi Zeeshan, On 5/13/07, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote: I've solved this problem. It was very easy (only if I knew how to do it before). I changed the UDP ports, i.e. 1. In sip.conf, bindport=5070 2. In my IP Phone server settings, www.myserver.com:5070 Now it seems to be working g

[asterisk-users] zaptel huge irq problem

Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and a

Re: [asterisk-users] OT ? Number portability, land line to Cell

I was going to port a number here in Ohio and Verizon said it would cost $90 to do so as they can charge what it cost them. Bob R Joe acquisto wrote: Having had various issues with local vendor (begins with "V"). am looking to move to all wireless. Anyone know if current vendor can refuse to p

Re: [asterisk-users] OT ? Number portability, land line to Cell

Quoting Joe acquisto <[EMAIL PROTECTED]>: Having had various issues with local vendor (begins with "V"). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? From what I've read, the Fed's seem to say "

Re: [asterisk-users] OT ? Number portability, land line to Cell

On Mon, 14 May 2007, Joe acquisto said something to this effect: Having had various issues with local vendor (begins with "V"). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? LNP does provide for th

Re: [asterisk-users] How to bring MoH volume down

Here the problem is that it is streaming audio from the Internet and I can't lower its volume. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mail

[asterisk-users] Codename Pineapple - Chan_sip3 - what's the status?

Friends, I have gotten a few questions lately on the status on the Codename Pineapple project, the project that hopefully will produce a more stable and SIP compliant SIP stack for Asterisk. Due to lack of funding, it's postponed until further notice. I have a few sponsors, but not enough

[asterisk-users] OT ? Number portability, land line to Cell

Having had various issues with local vendor (begins with "V"). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? >From what I've read, the Fed's seem to say "no", they cannot refuse, or impede >this. joe

[asterisk-users] Re: CITEL gateway does it work well?

The Citel Handset Gateways were the best option for our scenario. The cost per port for the number of buttons on our NEC DTerm/E phones was about half. Also, no network reengineering. We connected new 66 blocks to the Citel units. And just cutover from the old to the new. When you configure the

[asterisk-users] Play a file on a channel from the Manager API

> Is there any way to play a file on a channel from the Manager API > (other than from Originate)? This question was asked by someone else on the ast-dev list and the only advice given was that Redirect was the solution. I find myself with the same problem now but I don't understand the response

[asterisk-users] Asterisk and unicall + mfcr2 signalling

Hi, I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P card. I have one E1 with MFCR2 Signaling. I compiled asterisk + libunicall, and I can make calls over E1, but can't receive. Here the CLI when I make a call: -- Executing [EMAIL PROTECTED]:1] Dial("SIP/23-081cbc40", "Un

[asterisk-users] Re: Mobile Number to Mobile carrier mapping

Not now that they have intoduced number portability. The phone companies have to keep huge databases to keep track of which carrier to send the call to. -- -- Steven http://www.glimasoutheast.org "Ritesh Agrawal" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] Hi Folks, I

[asterisk-users] Re: RE: Digital Phones

We use the Handset Gateways from Citel. They convert SIP to Digital Handsets, so there is no hardware to add to the server and you can still use your 2-wire phone lines. -- -- Steven http://www.glimasoutheast.org "bilal ghayyad" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] H

Re: [asterisk-users] How to bring MoH volume down

Zeeshan, On Mon, 14 May 2007, Zeeshan Zakaria said something to this effect: MoH volume is uncomfortably high and I want to bring it down. Its mpg123. How can I do it? There are some settings in musiconhold.conf that may yield the desired effect: [default] mode=mp3 directory=/var/lib/aste

Re: [asterisk-users] How to bring MoH volume down

Remix your wav/mp3 files with a lower volume :) On 5/14/07, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote: Hi, MoH volume is uncomfortably high and I want to bring it down. Its mpg123. How can I do it? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation

[asterisk-users] Problem with queue

Asterisk 1.2.17 I am starting to have problem with one of my queue. Everytime when I try to login an agent with AgentCallBackLogin(), it will play periodic announcement for the queue during this function call. Also when this agent answer the call, during the conversation, the agent also hear t

RE: [asterisk-users] Sudden appearance of SIP/2.0 401 Unauthorized

Did you have the IP specified in sip.conf? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Yaakov Menken > Sent: May 13, 2007 10:43 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Sudden appearance of SIP/2.0 401

Re: [asterisk-users] RTP Mixer

On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil Dhawan wrote: > Hi > > Can somebody brief me the working of RTP mixer from MeetMe perspective. (RTP? in MeetMe? I don't really know this well but AFAIK RTP doesn't get mixed directly in MeetMe. MeetMe is a generic Asterisk channels mixer) Aparantl

RE: [asterisk-users] Call to Skype network

Hi There, Good guide on setting up chanskype on trixbox http://www.geek-pages.com/articles/asterisk/setting_up_trixbox/asterisk_to_use_skype.html also: http://www.chanskype.com/ working on my trixbox 2.0 :) Best Regards, Com os melhores cumprimentos, Hugo Picão Link Consulting - Redes&Segu

RE: [asterisk-users] Dundi and unknown remote peers

Hmm, I tried this, but I get the following notice: NOTICE[27486]: pbx_dundi.c:4695 set_config: Ignoring invalid EID entry '*' Do you perhaps know for any other option? Thanks, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Fri

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