Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

On Sunday 21 December 2003 00:29, Darren Nickerson wrote: > Folks, > > I can't seem to get DTMF signaling working properly using SJphone > connecting to Asterisk via a SIP connection. Here's an example of a > voicemail session where I entered 1234 for both the username and the > password: > > -

Re: [Asterisk-Users] Level(3) SIP termination services

Kudos John on an excellent set of questions! Not to mention the pointer to isp-clec. Thanks! John Todd wrote: [crossposted to isp-clec and asterisk-users] That's not such a great price at 11 million minutes, in my opinion. Did you ask them if they would speak to Asterisk via SIP? We have o

[Asterisk-Users] SJphone, Asterisk and DTMF tones ...

Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context = ) This

Re: [Asterisk-Users] ivr key press?

On Sat, 2003-12-20 at 21:59, Rich Adamson wrote: > Okay, let me see if I can reword it a little different. > > While the exten => 620,5,Background(npi-greeting) statement is executing, > I'm wanting to key in "3000" (as a valid extension). Should I be able to > do that, or is the response limited

Re: [Asterisk-Users] RxFAX application

Hi sergio On Fri, 19 Dec 2003 14:49:15 +0100 "Sergio Serrano Revuelto" <[EMAIL PROTECTED]> wrote: > Hi all, > I have tested RxFAX application through X100P card. When Fax > arrive i obtain the next trace: > snip > 5 (0.01679,-0.16590) -> 0.02781 > 6 ( -

Re: [Asterisk-Users] Level(3) SIP termination services

[crossposted to isp-clec and asterisk-users] Date: Fri, 19 Dec 2003 21:12:22 -0500 To: [EMAIL PROTECTED] From: John Todd <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Level(3) SIP termination services? Reply-To: [EMAIL PROTECTED] Anyone investigated the new service offerings from Level(3) in the l

Re: [Asterisk-Users] ivr key press?

Do you have the context with exten 3000 included in the same place as exten 620 is? If not then it can in no way work. bkw On Sat, 20 Dec 2003, Rich Adamson wrote: > Hi Steve, > > > On Sat, 2003-12-20 at 21:06, Rich Adamson wrote: > > > I'm testing an ivr implementation (first time) using: > >

Re: [Asterisk-Users] s, h, t, etc, extensions?

Care to expound a bit on that topic for the wiki, with some details as to why? JT At 6:32 PM -0600 12/20/03, Brian West wrote: On a side note.. you can't use exten => h, if you have any hope of getting accurate billing info. Its wise to call ResetCDR(w) in your exten => h, or not use it at all.

Re: [Asterisk-Users] ivr key press?

Hi Steve, > On Sat, 2003-12-20 at 21:06, Rich Adamson wrote: > > I'm testing an ivr implementation (first time) using: > > exten => 620,1,Wait,1 > > exten => 620,2,Answer > > exten => 620,3,DigitTimeout,5 > > exten => 620,4,ResponseTimeout,10 > > exten => 620,5,Background(npi-greeting) ; "Thanks

Re: [Asterisk-Users] ivr key press?

On Sat, 2003-12-20 at 21:06, Rich Adamson wrote: > I'm testing an ivr implementation (first time) using: > exten => 620,1,Wait,1 > exten => 620,2,Answer > exten => 620,3,DigitTimeout,5 > exten => 620,4,ResponseTimeout,10 > exten => 620,5,Background(npi-greeting) ; "Thanks for calling press 1 for"

[Asterisk-Users] ivr key press?

I'm testing an ivr implementation (first time) using: exten => 620,1,Wait,1 exten => 620,2,Answer exten => 620,3,DigitTimeout,5 exten => 620,4,ResponseTimeout,10 exten => 620,5,Background(npi-greeting) ; "Thanks for calling press 1 for" exten => 1,1,Goto(npi-directory,s,1) For initial testing, I

Re: [Asterisk-Users] BYEXTENSION and DBPut

Don't use BYEXTENSION use ${EXTEN} bkw On Sat, 20 Dec 2003, Walt Davis wrote: > Hey I need another pair of eyes on this! > > I would like to add phones numbers to the blacklist from any handset so I > did this: > > exten => _*66XX,1,StripMSD,3 > exten => _XX,2,DBPut,bla

Re: [Asterisk-Users] BYEXTENSION and DBPut

Walt Davis wrote: Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XX,1,StripMSD,3 exten => _XX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XX,3,Hangup However what I get in

[Asterisk-Users] BYEXTENSION and DBPut

Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XX,1,StripMSD,3 exten => _XX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XX,3,Hangup However what I get in the database i

Re: [Asterisk-Users] Level(3) SIP termination services

- Original Message - From: "Darnell Gadberry" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, December 20, 2003 2:53 PM Subject: [Asterisk-Users] Level(3) SIP termination services > John, > > I spoke with Level(3) last week regarding SIP termination. They > quoted $0.01/min

Re: [Asterisk-Users] More beginner questions

--- Jon Creasey <[EMAIL PROTECTED]> wrote: > Using DIAX softphone which seems to be working OK can get to > VM/echotest etc > in the demo context > > Am trying to setup FWD but get the following problems > > Can hear it ringing when dialing FWD no 612 for time. Connects but > no sound > from re

Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine

> > > Another problem I have is that the x100p detects loud noise on the > > > line as ringing. > > > > Use "callprogress=no" in the zapata.conf > > That's unclear as you left my text as well as Roberts text. > > Are you talking about "Another problem I have is that the x100p detects loud > n

Re: [Asterisk-Users] DIAX phone busy

Yes,I often get the same result, but not always. - Original Message - From: "Michael Welter" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, December 20, 2003 3:40 PM Subject: Re: [Asterisk-Users] DIAX phone busy > Yes, I've tried that as well. When I dial "70" from anoth

Re: [Asterisk-Users] ZTMonitor - /dev/dsp problem

ztmonitor 1 -v On Sat, 20 Dec 2003, Daniel Bichara wrote: > Hi, > > I am trying to run ZTMonitor to get debug info from my E100P board but I > got the following message: > > -bash-2.05b# ./ztmonitor 1 > Unable to open /dev/dsp: No such file or directory > Cannot open audio ... > -bash-2.05b# > >

Re: [Asterisk-Users] s, h, t, etc, extensions?

On a side note.. you can't use exten => h, if you have any hope of getting accurate billing info. Its wise to call ResetCDR(w) in your exten => h, or not use it at all. bkw On Sat, 20 Dec 2003, Steven Critchfield wrote: > On Sat, 2003-12-20 at 16:57, Rich Adamson wrote: > > I'm in the process o

Re: [Asterisk-Users] s, h, t, etc, extensions?

On Sat, 2003-12-20 at 16:57, Rich Adamson wrote: > I'm in the process of reworking my dialplan to include an ivr and > other items. I've seen several examples over the last several months > that mention the "s", "h", "t" (and probably others) extensions, but > I don't fully understand what they are

Re: [Asterisk-Users] s, h, t, etc, extensions?

> I'm in the process of reworking my dialplan to include an ivr and > other items. I've seen several examples over the last several months > that mention the "s", "h", "t" (and probably others) extensions, but > I don't fully understand what they are used for. Can someone either > give a short defi

Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine

On Saturday 20 December 2003 05:54 pm, Rich Adamson wrote: > > > That's interesting. My system is/was like that. (I can still simulate it > > > with the emergency phone that is paralleled/split at the incoming point). > > > > > > Pressing # certainly doesn't stop asterisk on my system. > > >

Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine

What I mean, is that if I pick up the analogue phone connected in parallel with the x100p and blow into the handset, The x100p detects it as ringing and rings the phone connected to the s100u. Also, if I phone asterisk from another phone, and hang up when it is ringing, the s100u phone keeps ringi

Re: [Asterisk-Users] DIAX phone busy

Yes, I've tried that as well. When I dial "70" from another extension, I hear ringing but the DIAX doesn't ring. Dan wrote: Hi, Yes, IAX2 is checked. Then change the line to: exten => 70,1,Dial(IAX2/mike,30,tr) BR, Dan ___ Asterisk-Users mailing lis

[Asterisk-Users] Cisco 7912 speed dials

Hi ! We have Cisco 7912 phones, and the doc says that I can "create up to four speed dial buttons on my phone using the Cisco CallManager". Does anyone knows which protocol is used to configure speed dials (Is it documented somewhere) ? Did someone tried to reverse engineer the protocol ? It woul

[Asterisk-Users] s, h, t, etc, extensions?

I'm in the process of reworking my dialplan to include an ivr and other items. I've seen several examples over the last several months that mention the "s", "h", "t" (and probably others) extensions, but I don't fully understand what they are used for. Can someone either give a short definition of

Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine

> > That's interesting. My system is/was like that. (I can still simulate it > > with the emergency phone that is paralleled/split at the incoming point). > > > > Pressing # certainly doesn't stop asterisk on my system. > > It only works if you don't have a # extension, and no i extension. >

Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine

On Sat, Dec 20, 2003 at 03:01:37PM -0500, Martin wrote: > Hello Rob. > > That's interesting. My system is/was like that. (I can still simulate it > with the emergency phone that is paralleled/split at the incoming point). > > Pressing # certainly doesn't stop asterisk on my system. It only wo

Re: [Asterisk-Users] Best SIP PHones to buy ?

- Original Message - From: "Michael T Farnworth" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, December 21, 2003 4:31 AM Subject: Re: [Asterisk-Users] Best SIP PHones to buy ? > We have bought around 30 Grandstream phones, both BT101 and BT102. In > general the phone is rea

Re: [Asterisk-Users] DIAX phone busy

Hi, > Yes, IAX2 is checked. > Then change the line to: exten => 70,1,Dial(IAX2/mike,30,tr) BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Channelized T3 card for linux

Title: Channelized T3 card for linux Regarding some previous posts on T3 Channelization, here is a card that might be of use: http://www.linuxdevices.com/news/NS8878561328.html Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerou

Re: [Asterisk-Users] asterisk on beowulf cluster

On Sat, 2003-12-20 at 14:54, Balaji NJL wrote: > Hi All, > > Can i install * on a beowulf cluster or Is * compatible to clusters. I > am planning to install a 4 node beowulf cluster using few cheap > hardwares. If no one had tried before i can spend some time on > installing and configuring * on

RE: Re: [Asterisk-Users] x100P incoming

> >immediate=no did not work either, so at least to get * not write to CDR, I will have to use command noCDR(). My extension context is as follows [x100pincoming] exten => s,1,Ringing exten => s,2,Wait(40) exten => s,3,NoCDR() It still write the CDR in MySQL, any idea how to get rid of that ?

Re: [Asterisk-Users] DIAX phone busy

Yes, IAX2 is checked. Thanks Dan wrote: Hi, Did you registered using IAX or IAX2? Check if in Registering page you have selected IAX2 or not. Did you check this one too? BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.co

[Asterisk-Users] asterisk on beowulf cluster

Hi All,   Can i install * on a beowulf cluster or Is * compatible to clusters. I am planning to install a 4 node beowulf cluster using few cheap hardwares. If no one had tried before i can spend some time on installing and configuring * on this cluster. Let me know.   thanks, -Balaji Do you

[Asterisk-Users] X101P + TDM400P

I thought I'd share my Asterisk experience, which hasn't exactly been as pleasant as I would like but now seems usable in most ways and more then I expected in other ways. I wanted a home PBX system, that would let me treat different callers different ways depending on CID. I initially bought th

Re: [Asterisk-Users] IVR sample config?

Joe Dennick wrote: Rich, This is in response to your question about and IVR Menu. Below is the dial-plan from the * at SAI. If you dial 1-800-747-9111 and the Extension 2998, you'll be able to hear this one in action. Now documented on http://www.voip-info.org/tiki-index.php?page=Asterisk+tips

Re: [Asterisk-Users] IVR sample config?

Rich Adamson wrote: Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, jus

Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine

On Saturday 20 December 2003 02:13 pm, Robert Murray wrote: > Hi > > I'm also interested in this. I currently have to press # to stop > asterisk when I pick up the phone. Even if I answer before asterisk, > it still answers. > > Cheers > > Rob Hello Rob. That's interesting. My system is/wa

[Asterisk-Users] Level(3) SIP termination services

John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com Date: F

Re: [Asterisk-Users] Asterisk MGCP register

On Sat, 2003-12-20 at 03:22, Senad Jordanovic wrote: > Hi, > > I am trying to figure out if * can register as a client on a remote MGCP > service. Just like SIP and other protocols > Do. Anyone tried this? > No I don't believe it can. The MGCP implementation in Asterisk is a CallAgent not a Use

Re: [Asterisk-Users] can X100P detect phone pick up like an answering machine

Hi I'm also interested in this. I currently have to press # to stop asterisk when I pick up the phone. Even if I answer before asterisk, it still answers. Cheers Rob On Sun, Dec 14, 2003 at 11:00:40PM -0500, Jim Flagg wrote: > If Asterisk is configured as a simple answering machine replacem

RE: [Asterisk-Users] More beginner questions

Once the connection is made between your * and FWD, your * passes it off so its really just a connection between your soft-phone and FWD. So, your soft-phone needs to be properly NATed, and the appropriate ports TO your soft-phone need to be opened. Each SIP phone will attempt to use different po

Re: [Asterisk-Users] x100P incoming

>> Exten => s,1,Wait(1000) This will make * not answering the call, but still you would see notices coming on your screen and also an entry in CDR. >immediate=no Will try this out SW From: Tilghman Lesher <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] x

[Asterisk-Users] More beginner questions

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192

Re: [Asterisk-Users] Best SIP PHones to buy ?

On Sat, 20 Dec 2003, Carlos Arnt wrote: > I wanna buy a Sip Phone, but what is the best and cheap one ? > > I see alot of messages about, grandstream , snow etc etc. We have bought around 30 Grandstream phones, both BT101 and BT102. In general the phone is reasonable, but it does have limitatio

Re: [Asterisk-Users] x100P incoming

On Friday 19 December 2003 08:12, David Gomillion wrote: > > How do I make x100P does not answer incoming calls ? > > The only thing that springs to mind is that you create an incoming > context, and have an extension like: > > Exten => s,1,Wait(1000) > > Dunno if it will work or no

Re: [Asterisk-Users] iconnect 480 unavailable msgs

> > [iconnect] > type=friend > username= > secret= > host=sipauth.deltathree.com > qualify=1000 > callerid="178197026" > allow=g721.1 Hello...looks like you have your codecs messed up. If you have enough bandwidth I recommend you modify it to: [iconnect] type=friend username=xx

RE: [Asterisk-Users] iconnect 480 unavailable msgs

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of vocalvoip Sent: 20 December 2003 16:14 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iconnect 480 unavailable msgs Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a

RE: [Asterisk-Users] IVR sample config?

Rich, This is in response to your question about and IVR Menu. Below is the dial-plan from the * at SAI. If you dial 1-800-747-9111 and the Extension 2998, you'll be able to hear this one in action. The key to creating it is to use Extension 205 (defined below) to record your menu prompts. Th

Re: [Asterisk-Users] iconnect 480 unavailable msgs

Hi, try to put in both users setup, canreinvite = no and what is this codec allow=g721.1 ... :))) Lubo vocalvoip wrote: Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asteri

Re: [Asterisk-Users] DIAX phone busy

Hi, > > Did you registered using IAX or IAX2? > > Check if in Registering page you have selected IAX2 or not. Did you check this one too? BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] iconnect 480 unavailable msgs

Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box are on public ips. The problem is that when i ring anyone in the world it'll ring they'll pickup and i can hear them

RE: [Asterisk-Users] Best SIP PHones to buy ?

Carlos,   It depends how good phone do you need. Grandstream phones are cheapest on market.  Sometimes, when you are dealing with something cheap you might think that it’s garbage and this is why it’s cheap. In case of Grandstream IP Phones, I would say they are normal, they have all basi

RE: [Asterisk-Users] Asterisk Crash

I didn't repeat this question. I read the responses and amended my request. I was simply asking for any suggestions as how to diagnose this problem including the ways to provide additional information to this list to help solve the problem. After reinstalling Linux and Asterisk from scratch with

RE: [Asterisk-Users] IVR sample config?

exten => _X., Goto(ivrmenu,s,1) [ivrmenu] exten => s,1,Ringing exten => s,2,DigitTimeout,30 exten => s,3,Background(something) ; press 1 for sales exten => t,3,Goto(business,0,1) include => business ;map yourl extens here... [business] exten => 0,1,... exten => 1,1, .. From: Ric

Re: [Asterisk-Users] DIAX phone busy

This was a typo--it already is: exten => 70,1,Dial(IAX/mike,30,tr) and I still get a busy... Thoughts? Dan wrote: Hi Michael, - Original Message - From: "Michael Welter" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] DIAX phone busy I've configured the DIAX phone. It registers wit

RE: [Asterisk-Users] IVR sample config?

Rich- Chapter 4 of the (so-called) draft handbook, details what you need to know pretty well. Here's the link: http://www.digium.com/handbook-draft.pdf Regards, Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] URL:

[Asterisk-Users] IVR sample config?

Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes

[Asterisk-Users] Best SIP PHones to buy ?

Hi People,   Can anyone help-me here with a simple question. I wanna buy a Sip Phone, but what is the best and cheap one ?   I see alot of messages about, grandstream , snow etc etc.   So for use with my * system, what sip phone is the best ??   Can him be used behind a nat system etc ? Or with a B

[Asterisk-Users] Chan_h323 & gnugk

Ok, I've managed to get inbound and outbound calling to work with chan_h323 and gnugk. A few questions: 1) if I do a reload in *, chan_h323 loses its registration with gnugk, and will no longer pass calls to it. A second reload will crash *. Is this supposed to be? 2) For a configuration in h3

[Asterisk-Users] Asterisk MGCP register

Hi, I am trying to figure out if * can register as a client on a remote MGCP service. Just like SIP and other protocols Do. Anyone tried this? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-use

AW: [Asterisk-Users] SNOM 200 and * issues

We included presence in the latest builds. This is necessary to enable auto-redial (call completion), at least in the eyes of the SIP hot-shots. In the latest image (2.03c, see http://snom.com/download/share) we included a flag where this feature can be turned off. But as long as it's not causing

[Asterisk-Users] ZTMonitor - /dev/dsp problem

Hi, I am trying to run ZTMonitor to get debug info from my E100P board but I got the following message: -bash-2.05b# ./ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... -bash-2.05b# Thanks, Daniel ___ Asterisk-Users

[Asterisk-Users] Chan_h323 docs

Title: Chan_h323 docs Jeremy, In some posting in the mailing lists, you mentioned that docs for h323 had been submitted but hadn't made it into distribution. Could you post those docs in your download directory?  I'm trying to understand the nuances of your driver, gnugk, and extensions

Re: [Asterisk-Users] Unable to detect process 256 frames

Does the same apply to GSM channels? - Original Message - From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, December 19, 2003 9:34 PM Subject: Re: [Asterisk-Users] Unable to detect process 256 frames > > >WARNING[1232119104]: File dsp.c, Line 1198 (ast_d

Re: [Asterisk-Users] DIAX phone busy

Hi Michael, >- Original Message - >From: "Michael Welter" <[EMAIL PROTECTED]> >Subject: [Asterisk-Users] DIAX phone busy > I've configured the DIAX phone. It registers with the * server, and I > am able to make calls from DIAX. Did you registered using IAX or IAX2? Check if in Registe