[Asterisk-Users] yet another question on DID trunks

or ISDN PRI (24 or 23 channels).   From what I understand of the Digium cards, DID signalling is not supported.   Hope that helps a bit -- David Schlossman  

Re: [Asterisk-Users] USA dial plan

  NA - Foreign NPA Toll Calls    1+10D   This says you currently dial local calls within 310 as 7 digits, but the plan will change to require 1+10 digits which is currently permitted.   Hope this helps David Schlossman [EMAIL PROTECTED]  

RE: [Asterisk-Users] RFC3389 messages with ATA 186

I setup a ATA-186 with no problems at all by following the instructions from John Todd’s excellent article at http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt   Hope that helps - David Schlossman [EMAIL PROTECTED]  

[Asterisk-Users] Asterisk SIP + Grandstream 100 phone

hi .. i've just converted myself back to a newbie by trying to experiment with some new stuff .. I have connected two grandstream Budgettone 100 phones to my asterisk, and trying to experiment with them .. I am trying to get into the asterisk sample basically .. when I dial 1000 asterisk receiv

[Asterisk-Users] Call Transfer, Budgettone 100

hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the

Re: [Asterisk-Users] CT1 and callerid / DNIS

t;   The service you might be referring to is Dialed Number Identification Service (DNIS) that is put on T1's for inbound 800 and 900 lines.  This is an inband delivery of the last 4-digits of a dialed number (800/900) that is passed into the PBX from the SP for callcenter or other routing.  Does Asterisk support this?   - David Schlossman ([EMAIL PROTECTED])  

[Asterisk-Users] {Scanned}

Hello All, I loaded [EMAIL PROTECTED] I have one X100P card. I try to dail out but get rejected. Any help... Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please contact

Re: [Asterisk-Users] Newbe Can't dial local numbers. {Scanned}

; ;exten => 8500,1,VoicemailMain ;exten => 8500,2,Hangup ; ; Or a conference room (you'll need to edit meetme.conf to enable this room) ; ;exten => 8600,1,Meetme,1234 ; ; Or playing an announcement to the called party, as soon it answers ; ;exten = 8700,1,Dial(${MARK},30,A(/path

[Asterisk-Users] No such extension {Scanned}

Hello All, I'm trying to dial out with no luck. I'm using [EMAIL PROTECTED] defaults. I have one X100P card and SJPhone. *CLI> dial 96985628 No such extension '96985628' in context 'default' Here is my exten [trunklocal] ; ; Local seven-digit dialing accessed through trunk interface ; exten =>

Re: [Asterisk-Users] No such extension {Scanned}

Well I guess I need to fix or create a channel now. Asterisk Ready. *CLI> dial [EMAIL PROTECTED] Jan 9 10:28:06 NOTICE[10750]: app_dial.c:743 dial_exec: Unable to create channel of type 'Zap' No luck when I dial [EMAIL PROTECTED] David > David wrote: >> Hello All, I

[Asterisk-Users] Route incoming call on 4 X100P to different Ext. {Scanned}

exten => s,1,Dial(SIP/300,10) So what is "s" . Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please contact [EMAIL PROTECTED] if you have question

[Asterisk-Users] test {Scanned}

test -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please contact [EMAIL PROTECTED] if you have questions about this email. ___ Asterisk-Users ma

[Asterisk-Users] error 488

Hello,   Can anybody help me with this issue?     -- Called 999302     -- Got SIP response 488 "Not Acceptable Here" back from 202.125.154.12   == No one is available to answer at this time     Why am I getting error 488. I’m using Sipura SPA-2000   Thanks  

[Asterisk-Users] error 488

Hello,   Can anybody help me with this issue?     -- Called 999302     -- Got SIP response 488 "Not Acceptable Here" back from 202.125.154.12   == No one is available to answer at this time     Why am I getting error 488. I’m using Sipura SPA-2000   Thanks  

RE: [Asterisk-Users] IAXTEL errors !

Christopher,   Any idea what causing “Max retries exceeded…” to happen?   Regards,   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Dobbs Sent: Saturday, January 19, 2002 8:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subje

Re: [Asterisk-Users] SIP USB Phone?

Hi,Adi, We provide the USB phone you wanted, it can access Asterisk natively. It can support Skype,X-Lite,X-PRO,eyeBeam,StanaPhone,SJphone,Net2Phone,Firefly and MSN too. To get more information about that, contact with me offline or goto our website please. Regards. David at

Re: [Asterisk-Users] I need Help everyone I just bough my Xten Eyebeam

I think your remote peer should use the Eyebeam and enabled the video too. Regards. David - Original Message - From: "Ing. Ignacio Ortega A." <[EMAIL PROTECTED]> To: Sent: Thursday, January 27, 2005 10:07 AM Subject: [Asterisk-Users] I need Help everyone I just bough

Re: [Asterisk-Users] Processing incoming calls with multiple contextstover PRI

Hi,Jason,     The TDM400P card failed to get the Callee number or DID, so the * don't know how to route the call. There are something difference between the analog line and the PRI line.       Regards.       David     http://www.iaxtalk.com   - Original Me

Re: [Asterisk-Users] Group Extension

Hi,Edgar, Config the agents.conf correctly and it will do what you want. For more information, search it in the wiki please. Regards. David http://www.iaxtalk.com - Original Message - From: "Edgar de Leon" <[EMAIL PROTECTED]> To: Sent: Mo

[Asterisk-Users] Where to download the soxmix please?

Hi,     Where could I download the soxmix please? I want to mix two .gsm files into one.     Regards.       David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] G729

Title: Untitled Document Hi All,   I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000 (latest Firmware) to use G729. In sip.conf I have set disallow=all, allow=g729   If Line1 is in use by an agent, then Line2 won't work and vice versa (Inbound Calls Only).  I have 40 license

[Asterisk-Users] Trying to do very simple Zaptel Config. NO LUCK!

Hi,I am trying to do the world's most simple install.I have a Wildcard TDM400P with 3 ports: 1 FXS on port1 and 2 FXOs on ports 3 and 4. (i'm not using port 3for now, put want it for expansion purposes)I simply (to start with) am looking to have the FXSphone ring when an FX0 port is dialed.  I woul

Re: [Asterisk-Users] Trying to do very simple Zaptel Config. NO LUCK!

nnels 3 and > 4, so when a call > is detected to arrive to FXO ports, will get to > incoming context and > will ring the receptionis. > > I have no experience with FXS ports, but try what i > have just tell you > and post how is going so far. > > best regards > &

[Asterisk-Users] Sip.conf problems

ype=friend, incoming calls doesn't works. If the type is set to another value (for example peer) incoming calls works fine, but outgoing calls doesn't works. What can I do? Thanks David ___ Asterisk-Users mailing list Asterisk-Users@lists.di

[Asterisk-Users] Setup Zoom V3 Router + VoIP register with Asterisk

Has anyone configured the Zoom V3 to connect with Asterisk? I bought one at Fry in San Diego for $99 bucks. I can't get it to register with Asterisk.. Thanks, David http://www.zoom.com/products/voip_products.html ___ Asterisk-Users mailing

[Asterisk-Users] Replace Adtran 608 With Asterisk

Hello All, I'm new to VoIP. I have a friend that has an Adtran 608 with 6 lines over a T-1. He likes my Asterisk box. Could I replace the Adtran 608 with an Asterisk box??? Any ideas on an interface card?? Thanks, David ___ Asterisk-Users mailing

[Asterisk-Users] VoiceMail Config Questions

age based on the caller’s profile? If yes, how? - How (if at all) can I configure the voicemail to send the emails via an external SMTP server?   Thanks.   David Do you Yahoo!? Plan great trips with Yahoo! Travel: Now over 17,000 guides!___ Asterisk-User

[Asterisk-Users] DID/T1

So DIDs are sharing available channels. In particular for ISDNs are DIDs sharing available channels? -- David Kwok CISSP,(ISC)2 61282315751 ext 1002 FWD#/IAXTEL# : 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

[Asterisk-Users] Voice Quality

Hello, I have setup two * servers and they are communicating using IAX. I'm passing calls from SRV A (internet connection T1) to SRV B (internet connection: 512). For some reasons I have an issue with the quality. The voice is a bit scratchy. I have tried iLBC and SPEEX, but it didn't make any di

RE: [Asterisk-Users] Voice Quality

Thanks Sean, I can't really use ULAW, bcz I will have more than 20 calls at the same time, and the entire path is a single codec (iLBC) You have mentioned something about IAX timing. How can set this value? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]

RE: [Asterisk-Users] Voice Quality

s the best way to configure * if there are some packet losts ? Thanks David Senior Network Administrator Call Center Development Services (t) 514.731.5046 ext. 226 (f) 514-731.5834 (m) 514.814.0203 (e) [EMAIL PROTECTED] (w) www.ccds.ca -Original Message- From: [EMAIL PROTECTE

RE: [Asterisk-Users] Voice Quality

-Commercial Discussion Subject: Re: [Asterisk-Users] Voice Quality David: Bandwidth may be an issue; however, do you have any timing devices installed? Digium's hardware (or any generic knockoffs) will provide this. There are also some other ways, such as ztdummy or a usb controller (haven&#

RE: [Asterisk-Users] Voice Quality

Thanks for your reply... I was told to disable the jitter if using trunk=yes in iax.conf.. Have you guys had any experince with having jiiterbuffer=yes and trunk=yes? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev Sent: Wedne

[Asterisk-Users] my_zt_write

Title: Untitled Document Hello Guys,   Any idea what this means:   WARNING[2138]: chan_zap.c:4409 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 - audio may have been lost   Thanks  

[Asterisk-Users] Where to contrib the sound files ?

Hello,every one,       I have recorded the voice files with mandarin (China). Where should I contrib the files ?       Regards.       David at iaxtalk.com   ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Update Asterisk

Hello, I have a version of asterisk running on my server for more than 1 year. I wanna update it to the latest version without over-writing any of the config files. How can I do this? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium

RE: [Asterisk-Users] *@Home .6 adding a outside number to a group {Scanned}

Thanks I didn't see it. Sound like [EMAIL PROTECTED] isn't well liked on this list.. Thanks for your help, David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: Monday, March 21, 2005 7:39 PM To: Asterisk Users Mailing

RE: [Asterisk-Users] *@Home .6 adding a outside number to agroup{Scanned} {Scanned}

Thanks, I don't play with web pages to much. It has a lot of great stuff for a newbe like me. Thanks, David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Tuesday, March 22, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Comme

[Asterisk-Users] Where to start. {Scanned}

Hello All, Yep I'm a newbe. I'm just started to play with asterisk. What I have Redhat Fedora Core 2 (New install) 3 X100P cards. I installed zaptel-1.0.3 libpri-1.0.3 asterisk-1.0.3 Where should I start?? -- Thanks, David -- This message has been scanned for viruses and dangero

Re: [Asterisk-Users] te410p and Telstra Onramp 10

On Friday 13 August 2004 22:06, Craig Guy wrote: > Hi, > > > Is an onramp 10 what is referred to as a 'channel bank'? A channel bank is a device that would take the onramp 10 in one side a present 10 separate PSTN lines out the other. -- Best

[Asterisk-Users] Asterisk server keeps crashing

e pbx works just as intended, but the crashes are making the system unusable. I am pulling my hair out with this problem and my SO wants me to give up the project. Any and all help will be greatly appreciated! Thanks, David ___ Asterisk-Users mailing

Re: [Asterisk-Users] TDM400P lockups (FXO)

It sounds like my lockups may be related since my TDM422b card has the FXS FXS FXO FXO configuration and doesn't have an FXO in position 1 either. My card is identified in software as Rev E/F and has the wire jumper on the back. David Richard Scobie said: > > > Maciej Kietlinski

Re: [Asterisk-Users] Leading '0's and what do 'pri_dialplan', 'pridialplan' and 'prilocaldialplan' in zapata.conf do?

Nick, I too battled a similar problem with my TDM400p. I solved it by putting the following in the channel descriptions in zapata.conf: stripmsd=0 Clearly this is not the default which I think should be obvious... David Nick Barnes said: > > Hi all, > > I've been batting

[Asterisk-Users] IAXy intermittent sound problem

anged to 4" all further sound stops. The machine seems to be stalling, but I have noload on both oss and alsa modules (they seem to be the culprit of all googled problems and I don't need a console). David Sep 11 20:00:12 VERBOSE[671762]: -- Goto (intern-post,18887452654,1) Sep 11 20:

Re: [Asterisk-Users] IAXy intermittent sound problem

I gather from the lack of response that no one has had a similar problem or knows how to troubleshoot the problem. The "Ooh, voice format changed to 4" is a mystery to me since everything I find with that message has a coder format where I have a 4. David David said: > I

Re: [Asterisk-Users] Leading '0's and what do 'pri_dialplan', 'pridialplan' and 'prilocaldialplan' in zapata.conf do?

calls. David Tim Robinson said: > Nick - > > Put > > nationalprefix=0 > internationalprefix=00 > > in your zapata.conf file! > > Magic! > Rgds > Tim > > Nick Barnes wrote: > >>Hi all, >> >>I've been batting my head against a

Re: [Asterisk-Users] IAXy intermittent sound problem

some other file, and ironically, the file "missing" is missing). Since I can't compile the cvs libiax, I am back to using the debian libiax0 and libiax-dev. And since I can't get the CVS asterisk to run, I am back to RC2 and the problems listed in my last email. Please let me

Re: [asterisk-users] IP Phone support SIP and IAX

> > Hi All; > > Anyone can advise for a good IP Phone that has the > ability to support SIP firmware and IAX firmware? > Ofcourse, SIP there is a lot, but we need also the > ability to use IAX (as it is good for NAT). > > Any advise. > Regards > Bilal > > > I am using an atcom at-530 http://www.

Re: [asterisk-users] audio disappeared after ztdummy install

ronald ramos wrote: > Hi, > > For now i just turned off acpi. and it works now. > just dont know what's the connection of that though > :-) > > i will try to do the things you guys suggested also > when i get the chance, thanks for you help! > > regards, > nhadie > > > --- Tzafrir Cohen <[EMAI

Re: [asterisk-users] IAX IP Phone

Joseph wrote: > On 04/05/08 05:16, bilal ghayyad wrote: >> Hi All; >> >> Till now I am not able to find a good IAX IP Phone or >> Gateway that can be used with good quality. >> >> Anyone can advise for good one? >> >> Regards >> Bilal > > I've not seen IAX phone so your best option will be IAXy a

[asterisk-users] MySQL/IVR Integration

e and retrieves alphanumeric data, plays/announces it to the user and deletes the row from the database: The SQL queries would look something like: SELECT user, pwd FROM codes WHERE dialed = '111'; DELETE FROM codes WHERE user=$user AND pwd=$pwd;

[asterisk-users] Addons

tory". Is there any way to bypass/ignore the fact that MySQL is installed separately and enable the installation of the addons? Thanks, David Got a little couch potato? Check out fun summer

Re: [asterisk-users] problem with my softphone

> Hello, when with my client X-lite try to register in the server that > say me, > Registration error:501 Not implemented. Google is your friend; http://www.google.com/search?hl=en&q=asterisk+register+x-lite&btnG=Google+Search&aq=f&oq= ___ -- Bandwidth

Re: [asterisk-users] Sendmail for Voicemail

MAIL /etc/asterisk/voicemail.conf [default] 1000 => ,David Abbott,[EMAIL PROTECTED] -- Powered by Gentoo GNU/LINUX http://www.linuxcrazy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

Re: [asterisk-users] IAX2 client for "eee pc 1000"

Joseph wrote: > What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux > software)? > > I'll eventually replace this crippled Linux with something better but I don't > time to play around with it as most divers and modules are still too new and > not fully available in all distros

Re: [asterisk-users] IAX2 client for "eee pc 1000"

Joseph wrote: > > It keeps complaining about /lib/tls/libc.so.6 'GLIBC_2.4' not found. > > How do you install this library on EEE pc Xandros? (I know Xandros is Debian > based) but this is eee pc. > > You should ask on another list but this should get you started; http://forum.eeeuser.com/viewt

[asterisk-users] 2008 Post Count

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On the Python Tutor mailing list Kent Johnson uses a script to find the top posters for the year. If this or something like it has been posted, sorry for the noise; 2008 Steve Totaro 796 Tzafrir Cohen 749 Tilghman Lesher 496 Alex Balashov 354 Oliv

Re: [asterisk-users] 2008 Post Count

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Totaro wrote: | I would venture to guess that I would be in the top three (if not 1st) | for the last five or more years. Would it be very hard to run the | same script for years gone by? It would be interesting to see, | especially when Marks

Re: [asterisk-users] voicemail not sending emails

Roberto Milani wrote: > > > >Roberto - I noticed in your original email you had the lines > something like > > > >mailcmd=/opt/local/bin/msmtp -t ; --from blah > > AND > >serveremail=from=blah > > > >In mailcmd everything after the ; will be ignored as a comment > >In serveremail - well

Re: [asterisk-users] voicemail not sending emails

david wrote: > Roberto Milani wrote: > >> > >> >Roberto - I noticed in your original email you had the lines >> something like >> > >> >mailcmd=/opt/local/bin/msmtp -t ; --from blah >> > AND >> >serveremail=from=blah

Re: [asterisk-users] Voice only works from one way.

> Yes, both Asterisk and Cisco are behind Nat. My asterisk box is behind a dsl modem and router. All traffic is bridged from the modem to the router. Here are the settings on the router; http://dwabbott.com/pictures/port_forward.png http://dwabbott.com/pictures/range_forward.png The asterisk box i

Re: [asterisk-users] mpg123 problem

fateme fatah wrote: > Hi: > I want to install mpg123-0.59r on my asterisk server.I downloaded it > in /usr/src then untared it and I typed these command : > #cd /usr/src/mpg123-0.59r > #make linux > after run make linux ,I saw 2 errors in terminal: > make CC=gcc LDFLAGS= \ > OBJECTS='decode_i386.o

Re: [asterisk-users] voicemail didn't send voice message to my email

Have you configured and tested sendmail? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or up

Re: [asterisk-users] Need Help Regarding Asterisk

Preetish Kakkar wrote: > But how would my calls be transferred to extension phones from > asterisk server. Would i need to connect those phones to Digium card > as well. What i mean is would digium card have a main extension where > i would connect main pstn line and other 3 port where i would c

Re: [asterisk-users] VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...

' 'phonetic/z_p' european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM [default] => ,David Abbott,x...@.net Thats all I have in there, asterisk will use my SMTP client without me doing anything. I am using asterisk 1.4 - -david - -- Powere

Re: [asterisk-users] VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 jonas kellens wrote: | David, | | what is your SMTP-client then ? | | Did you change the mailcommand 'mailcmd' in voicemail.conf ?? Or is it | still /usr/sbin/sendmail ?? I don't have mailcmd in voicemail.conf, I was under the imp

Re: [asterisk-users] Nobody picked up in 20000 ms

Joseph wrote: > On 06/21/09 14:04, Joseph wrote: >> When I call internal extension from PSTN line everything is working >> correctly phones are ringing they way they should but internally when I try >> to dial two >> extensions on one sipura unit and my Digium IAXY unit rings only once and >> ca

Re: [asterisk-users] Recordin call in asterisk

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Bayardo Sanchez wrote: | I need help need recording all call for my pbx but i am a novato in | asterisk my confi for record is: | | exten=>_NX,n,Set(CALLFILENAME=CLIENTE-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}

Re: [asterisk-users] IAX IP Phone

bilal ghayyad wrote: > Hi All; > > Anyone knows an IAX IP Phone works fine and tested? > > Does polycom support IAX IP Phone? > > Regards > Bilal > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-us

Re: [asterisk-users] IAX IP Phone

bilal ghayyad wrote: > Dear David; > > At what price u get it? > > Did u test it with IAX and SIP? Are u sure it is good? As really I did not > deal with chinese phone until now and I found it fine. > > Regards > Bilal > > > --- On Mon, 1/19/09, David wrote: &

Re: [asterisk-users] soft phone

Try iaxLite or sipLite - Original Message - From: David fire To: bilmar...@yahoo.com ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 26, 2009 7:43 AM Subject: Re: [asterisk-users] soft phone there isnt any free soft phone wich support G729

Re: [asterisk-users] Autodialler query

Hi Sriram, > the customer should be billed a premium rate ex, Rs.9 per minute.. Will be billed by you or by telecomm company? Regards David - Original Message - From: Sriram To: asterisk-users@lists.digium.com Sent: Thursday, February 05, 2009 1:46 PM Subject: [aster

[asterisk-users] Problem redirecting user running a Dynamic feature

a dynamic feature, the redirect fails. Additionally, if I do it backwards and the calling user transfers the called user using #3, it works perfectly without any problems. What have I done wrong ? Is there a better way to implement a custom transfer feature? Thanks, David __

[asterisk-users] Video phone crashing meetme on asterisk 1.4.

Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command

[asterisk-users] Video phone crashing meetme on asterisk 1.4.

And finally : == Spawn extension (default, 8500, 3) exited non-zero on 'SIP/system.117-b6c02408' Is it the phone or meetme that is not working properly? Why would meetme accept video if it does not work? How can I tell meetme to never do video and still allow it between peers?

Re: [asterisk-users] Provisioning GXP 2000

password is. David Michiel van Baak wrote: > On 13:45, Thu 26 Mar 09, Lutgring, Sam wrote: > >> My preferred method is to use my own TFTP server. This makes changes to >> accounts/phones very fast and easy. The whole process takes me about 5 >> minutes to depl

Re: [asterisk-users] Building a System.

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John F. Ervin wrote: | So, people have recommended building a system from scratch, start with a | CentOS base and installing asterisk and all of the other utilities. | I've only used Trixbox for my business system. I'm wondering what | surprises I'd r

Re: [Asterisk-Users] USB handset wanted

rted in Windows > environment. I use an Eutetcics IPP200 USB handset with linux usb audio drivers and kiax for software. http://www.eutecticsinc.com/news/news.html It works ok but it depends on the audio drivers. I thought any USB handset would work with linux sub audio drivers but that was just an as

[Asterisk-Users] DNID on IAX2 trunks?

erisk.  I know you can pass info INTO AGI, but can you pass the info back OUT of AGI into the Asterisk extensions.conf dialplan?Many thanks. David Yahoo! FareChase - Search multiple travel sites in one click. ___ --Bandwidth and Coloc

[Asterisk-Users] Fax2Mail

to email addresses.   Thank you in advance.   David Yahoo! Music Unlimited - Access over 1 million songs. Try it free.___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[asterisk-users] "No Mailbox" Prompt

omeone reaches an extension that doesn't have an active mailbox? Something like: exten => _123105.,2,Playback(no-box,noanswer) Thanks. David. Have a burning question? Go to www.Answers.yahoo.com a

[Asterisk-Users] Dial Macro timeout fails

meout never occurs, I never see MACRO_RESULT set, and the call is connected even though it shouldn't be until the caller presses 1. Any help (or explanation about why this doesn't work) will be greatly appreciated. I have been pulling my hair out trying to get this to work. Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Dial Macro timeout fails

To add to the mystery, if the cell phone answers and presses "1" as requested, the logs don't register priority 1,1 being executed. It is as if the macro has prematurely aborted. David David said: > I just downloaded, compiled and installed Asterisk 1.2.9.1. I did this &

Re: [Asterisk-Users] Dial Macro timeout fails

Thanks for the response! I used your template to write a similar one for us and it works great. I wonder if there is a bug in the macro timeout code. David whois wes said: > This may sound stupid, but I had a similar issue that I solved by > placing an Answer at the beginning of what wo

[asterisk-users] Pager Voicemail Message

Hello,In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system.Is there a way to manipulate this message, as well?Thanks,David___

[asterisk-users] H.263 Video Messages

Hello,I'm trying to set the Asterisk to leave a video message to the mailbox, but there is some compatibility problem, although h263 is identified as the matching codec, as you can see in the debug messages below:Capabilities: us - 0x80100 (g729|h263), peer - audio=0x43f (g723|gsm|ulaw|alaw|g726|ad

Re: [asterisk-users] Pager Voicemail Message

ECTED]>To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, October 30, 2006 2:15:03 AMSubject: Re: [asterisk-users] Pager Voicemail Message Yes. It should be in that same file. Poke around. - Original Message - From: David To: asterisk-users

Re: [asterisk-users] Pager Voicemail Message

terisk Users Mailing List - Non-Commercial Discussion Sent: Monday, October 30, 2006 8:06:22 AMSubject: Re: [asterisk-users] Pager Voicemail Message On 10/29/06, David <[EMAIL PROTECTED]> wrote:I looked. There's nothing there.I even did a search under /etc/asterisk for files containing &

[asterisk-users] DTMF not being sent ( RFC2833 )

wit's end and do not know where to go from here. I would really appreciate it if someone could give me some pointers on where to go next, what additionnal debugging steps I should perform. I would also really appreciate if someone could propose a solution. Please help! David Never give u

Re: [asterisk-users] DTMF not being sent ( RFC2833 )

ug occurs when one asterisk calls the second asterisk which bridges to a DAHDI channel. My next step is too compare the SIP signalling between the two calls. Maybe something is different. What I find really weird is that the DTMF is incorrectly sent from the first asterisk only when the

Re: [asterisk-users] DTMF not being sent ( RFC2833 )

Id and SDP version. Everything else is identical. So the problem appears to be caused in the RTP and not in the SIP. So something about the RTP packets coming from the DAHDI channel on asterisk-pri makes asterisk server send invalid DTMF. David On 11-04-24 11:42 AM, David wrote: I did more te

[asterisk-users] DTMF incorrectly sent ( RFC2833 or SIPInfo )

HDI/1-1 [Apr 24 12:50:22] DTMF[2845]: channel.c:2802 __ast_read: DTMF end '#' received on DAHDI/1-1, duration 80 ms [Apr 24 12:50:22] DTMF[2845]: channel.c:2858 __ast_read: DTMF end passthrough '#' on DAHDI/1-1 [Apr 24 12:50:22] DTMF[2845]: channel.c:2874

[asterisk-users] AGI WAIT FOR DIGIT - key press BEFORE command

ser enters a digit between the "200 result=" and the next "WAIT FOR DIGIT"? Will the next WAIT FOR DIGIT catch the digit? Is the digit lost? How can I insure I don't lose the digit ? I am calling one digit at a time because I wan

Re: [asterisk-users] asterisk practices

wait for closing hours. David On 2011-04-27 13:34, vip killa wrote: I just completed building a feature rich asterisk voicemail system using perl, php, and mysql. My only concern is that the system i built will not be able to handle the call volume needed. Let me start by explaining my

Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

r link and will give you lots of distortions on your VoIP. David On 2011-04-28 11:25, Bruce B wrote: Hi everyone, How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines s

Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

holas wrote: *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David *Sent:* Thursday, April 28, 2011 10:32 AM *To:* asterisk-users@lists.digium.com *Subject

[asterisk-users] chan_dahdi.c, dtmfmute, rtp.c

real dtmf as echo. I tried both mg2 and oslec echo canceller, I saw no difference between the two. What is the next step in debugging this issue ? David -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.

Re: [asterisk-users] Asterisk for Razberry Pi

I've created some images. I currently don't have a free Raspberry Pi so I have not updated any images for a little while. A how to on building your own. www.klaverstyn.com.au/david/wiki/index.php?title=Asterisk_for_Raspberry_Pi A how to on writing a pre-compiled i

RE: [Asterisk-Users] (newbie) Hardware sizing question

and click the "I'm feeling lucky" button. It will take you into the tiki. Good luck, and I hope you find what you need, or at least what questions you need to ask. > Thanks > Javed David Gomillion PS. You may want to use plain text for email instead of HTML when posting t

RE: [Asterisk-Users] 911 and lawsuits and redundancy

t on sale for $59... From just a cursory look, it seems to be a fully-featured Ultra2 RAID controller, which claims to work with Linux. Just wondering if anyone used one of these before I take the time to order it. Thanks, David ___ Asterisk-Users ma

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