Re: [Asterisk-Users] SCO/Linux concerns

On Wednesday 30 July 2003 07:07 pm, Ajit M Kallingal wrote: > Hello > Since I am getting a bit concerned about the SCO vs IBM issue, I was > wondering if can I can setup Asterisk on FreeBSD is it supported ? > Are drivers for Digium cards available on FreeBSD ? > > Thanks > Ajit Not really answeri

Re: [Asterisk-Users] 24port or higher fxs

TE410P + Adtran TSU 750 (or many other channel banks for that matter) Jeremy McNamara Kelvin Chua wrote: hi guys, i'm in need of several 24port or higher fxs device which supports sip, aside from mediatrix and audiocodes (cisco's vg248 doesn't support sip), do you have any idea who else manu

[Asterisk-Users] 24port or higher fxs

hi guys,   i'm in need of several 24port or higher fxs device which supports sip, aside from mediatrix and audiocodes (cisco's vg248 doesn't support sip), do you have any idea who else manufactures such device?     

RE: [Asterisk-Users] PHP API for Manager - Plaintext auth needed?

I also dislike plaintext, but the vast majority of users will probably run the PHP script on the * system itself, so plaintext won't really hurt. Hell, I doubt that most will even bother to run the scripts on a secure server. I'd say set the default to md5, but leave plaintext as an option. --

[Asterisk-Users] PHP API for Manager - Plaintext auth needed?

Quick question: My PHP script is now able to connect to the manager port and successfully authenticate using MD5. I would strongly prefer not to do plaintext authentication at all. Would anyone object to plaintext authentication being left out? -- JustThe.net Internet & Multimedia Svcs. [The F

RE: [Asterisk-Users] Best Analog sets for use w/*

TC, Have you used these phones? They seem to be pretty nifty... I'm wading through the documentation to see how well they would integrate. Sincerely, Andy Hester Consero > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of TC > Sent: Thursday, Ju

[Asterisk-Users] one way audio h323 callmanager

there's this one way audio problem using h323 (CVS) with cisco callmanager? has anybody encountered this problem? oh323 works ok though... or is there any workaround for this? thanks

Re: [Asterisk-Users] Best Analog sets for use w/*

>Hi All, > I am considering testing out some analog sets with * for a customer and >thought I would ask what analog phones are in use? The customer would >require the usual business functionality ie hold, conference calling, and >preferably a soft key to vm and line apearances(correct terminology

[Asterisk-Users] Best Analog sets for use w/*

Hi All, I am considering testing out some analog sets with * for a customer and thought I would ask what analog phones are in use? The customer would require the usual business functionality ie hold, conference calling, and preferably a soft key to vm and line apearances(correct terminolog

[Asterisk-Users] Mutex problem in sip?

Title: Mutex problem in sip? Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... c

Re: [Asterisk-Users] AddQueueMember and RemoveQueueMember

Figured out what to do. exten => 900,1,Queue(techsupport|HTt) exten => 900,2,Voicemail2(b111) And DO NOT put "members =>" in the queues.conf Let members come and go via the AddQueueMember and RemoveQueueMember Works perfect. bkw On Thu, 31 Jul 2003, Brian West wrote: > I currently have this:

Re: [Asterisk-Users] SIP calls cause Segmentation Fault

I actually found this same thing, and traced it down to app_dial.c line 190. It doesn't explicitly check for a valid chan before trying to use it and it segfaults when it does a strlen on a chan entity. I simply put a check in that winner was non-zero before comparing it to o->chan: if (winner &&

[Asterisk-Users] Queue and Agents in CVS

Every time I have upgraded via CVS in last few weeks, the queue and agent program doesn't appear to function together. The calls are not getting passed: Agent 308 is logged in and idle. Yet I have a customer that has been on hold for over 8 minutes. Any Ideas?

RE: [Asterisk-Users] Vonage

I just found this link: http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat It suggests that your username is your phone number, and your password is the 10 digit activation number. Steve On Thu, 2003-07-31 at 15:23, Joe Cooke wrote: > I haven't tried it yet, but I believe the fo

RE: [Asterisk-Users] Vonage

I haven't tried it yet, but I believe the following is correct: SIP id: the original 10-digit activation number that you use to initially register your phone - this is *not* your phone number. SIP password: Codec: g723.1 Server IP: packet8.net I would assume that a packet capture would confirm m

RE: [Asterisk-Users] Vonage

On Thu, 2003-07-31 at 10:20, Humberto Atristain wrote: > 8x8 is the only one I know (or packet8) a little less "important" What specific information do I need to get from them in order to get Asterisk to connect directly? I assume I'll need the following: * SIP id * SIP password

[Asterisk-Users] Zaptel cards, working FXS and SIP, no audio?

Greetings again all, With the help of another list member I was able to get the my TDM400P card working properly (and the PRI card loaded too for channels 1-24). >From the 2 FXS ports I can call back and forth, and MSN Messenger works fine as a SIP client. Now I'm trying to get the demo (ext 500)

Re: [Asterisk-Users] SIP Registration

sip show registry is when asterisk registers with some gateway. you want to look at sip show peers or sip show users. regards Martin On Thu, 31 Jul 2003, Steve Woolley wrote: > I am trying to get SIP registrations to work within Asterisk. From my > snom 200 phone (and on my SJPhone soft client)

[Asterisk-Users] SIP Registration

I am trying to get SIP registrations to work within Asterisk. From my snom 200 phone (and on my SJPhone soft client) I can dial via extension. Example: To Dial extension 1110 on my asterisk1 server: I can simply enter SIP:[EMAIL PROTECTED] and the call goes through just like it should. As I unde

Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected - partially solved

Hi, The only 'strange' thing with my system is that it runs over VMWare Workstation, with a WinXP Pro host. I have no errors in the logs or during boot time. BR, Dan - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 31, 2003 10:38

Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected

then... where's the error as the application does what it must do. Tried with several system commands with the same result. With de modification is the source everything works as expected. BR, Dan - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent:

Re: [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected - partially solved

System should return with 0 when it's successfull. You have to have something wrong with your system. Read "man system" Martin On Thu, 31 Jul 2003, Dan wrote: > Hi Martin, > > I have modified the 'app_system.c' file like that and then recompile > asterisk: > > /* Do our thing here */ >

Re: [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected

One thing is sure: the system should return with 0 if it's successful. Read "man system" regards Martin On Thu, 31 Jul 2003, Dan wrote: > Something even more interesting. > I have tried to execute the command 'ls' in the following line: > ... > exten => s,3,System(ls) > ... > > And this is the r

Re: [Asterisk-Users] Manager

On Thursday 31 July 2003 14:05, Greg Renouf wrote: > Perhaps it would make sense to integrate the Manager function > controls into IAX2? From my understanding, IAX2 already has built-in > encryption (MD5 or RSA)- communications would be secured this way. AFAIK, both IAX and IAX2 can use encryptio

Re: [Asterisk-Users] ADSI and SoftKeys

On Thu, 2003-07-31 at 20:59, Armand A. Verstappen wrote: > > > > Has anyone solved the problem on the ADSI phones > > > > that when you hit one of the soft keys, the Number Pad > > > > stops working? > > > > > It relates to not putting the phone back into voice mode when the > > prompts are playin

Re: [Asterisk-Users] Manager

Perhaps it would make sense to integrate the Manager function controls into IAX2? From my understanding, IAX2 already has built-in encryption (MD5 or RSA)- communications would be secured this way. Many of the 'Manager' features could be useful to 'superusers' (e.g. receptionists, local operators

Re: [Asterisk-Users] ADSI and SoftKeys

On Thu, 2003-07-31 at 16:30, Jayson Vantuyl wrote: > On Wed, Jul 30, 2003 at 05:07:50PM +0200, Armand A. Verstappen wrote: > > On Wed, 2003-07-30 at 16:40, John Congdon wrote: > > > Has anyone solved the problem on the ADSI phones > > > that when you hit one of the soft keys, the Number Pad > > > s

[Asterisk-Users] retrieving dialed number when overlap dialing?

I have a number of local users who can dial out on a pri channel using the fantastic new overlap dialing feature. I would like to add a speed dialing feature, such as 1. User picks up and dials out (dial startet with option 'H') 2. User hangs up call with '*' 3. Dialed number is stored in a variab

Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected - partially solved

Hi Martin, I have modified the 'app_system.c' file like that and then recompile asterisk: /* Do our thing here */ res = system((char *)data); // if (res < 0) { // ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data); // res = -1; // }

Re: [Asterisk-Users] Vonage

Right...The config file itself is encrypted so if you capture the downlaod and can crack the RC4 algorithm then you are in. The SIP authentication itself is just an MD5 hash of the password. If it is a short password you can try to brute force your way into cracking it. But if it is a long one (

Re: [Asterisk-Users] Parking calls - why doesn't work?

Hi Wade, I have no extensions defined starting with 7. Do not forget that for 702 to 720 I get the correct message .."there is not any call parked at this extension"... Dan - Original Message - From: "Wade Weppler" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 31, 200

Re: [Asterisk-Users] Vonage

> There is no way for you to know the vonage password associated with your > account. Even if you sniff out the tftp download, its encrypted. Clearly there must be a way to decrypt it back to plaintext, however, since SIP uses a chap-style MD5 scheme, which requires knowing original password at b

Re: [Asterisk-Users] SIP calls cause Segmentation Fault

Yes, find me on #asterisk so I can login. Be sure you're generating cores and running on very latest CVS. Mark On Thu, 31 Jul 2003, Dave Alan Caruana wrote: > I have an asterisk installation at a client, it's quite simple. > Basically it's an asterisk downloaded from CVS about > a week ago, wit

Re: [Asterisk-Users] Vonage

[EMAIL PROTECTED] wrote: You know, I've called them several times and left my telephone number to call back. I've never heard from them. For some reason, I have never received any voicemail from you. You know, many people on the list raved about them. But for a company with a completely usel

[Asterisk-Users] AddQueueMember and RemoveQueueMember

I currently have this: [agentlogin] exten => 800,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM}) exten => 800,2,SoftHangup exten => 801,1,RemoveQueueMember(techsupport|SIP/${CALLERIDNUM}) exten => 801,2,SoftHangup [callqueue] exten => 900,1,Queue(techsupport) exten => 900,2,SoftHangup But I

Re: [Asterisk-Users] Manager

On Thursday 31 July 2003 12:13, Steven Critchfield wrote: > On Thu, 2003-07-31 at 11:24, Steven J. Sobol wrote: > > On Thu, 31 Jul 2003, Dan wrote: > > > Hi Roy, > > > > > > It is not much safer to use SSH to connect to the computer and > > > then 'asterisk -r' to the asterisk console? > > > >

RE: [Asterisk-Users] Parking calls - why doesn't work?

Sounds like your active context might have an existing extension 701 or starting with 701? -wade > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Dan > Sent: Thursday, July 31, 2003 1:32 PM > To: [EMAIL PROTECTED] > Subject: Re: [Aste

RE: [Asterisk-Users] Vonage

Jeremy MacNamara an active list contributor is involved in the nufone project. In dealing with him personally via email, I have found him very quick and a pleasure to deal with. He seems to give people help on the list all the time reguarding various asterisk issues. Being the owner of a sma

Re: [Asterisk-Users] Parking calls - why doesn't work?

Yup. 701 for parking and 702-720 for parked calls. Dan P.S. Tried with default values too... but without success... - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 31, 2003 8:13 PM Subject: Re: [Asterisk-Users] Parking call

Re: [Asterisk-Users] Manager

On Thu, 2003-07-31 at 12:25, Steven J. Sobol wrote: > At 12:13 PM 7/31/2003 -0500, you wrote: > > > >Does anyone see any potential problem with eventually linking in ssl in > >such a way as to allow connections to be made securely? I do not know > >the amount of work this would entail, but this w

Re: [Asterisk-Users] Vonage

Ricardo Villa wrote: There is no way for you to know the vonage password associated with your account. Even if you sniff out the tftp download, its encrypted. Encryption only stops the unmotivated. (is that a word?) Jeremy McNamara ___ Asterisk-U

RE: [Asterisk-Users] Vonage

[EMAIL PROTECTED] wrote on 07/31/2003 12:52:10 PM: > www.nufone.net is entirely Asterisk/IAX. You know, I've called them several times and left my telephone number to call back. I've never heard from them. You know, many people on the list raved about them. But for a company with a complet

Re: [Asterisk-Users] Manager

At 12:13 PM 7/31/2003 -0500, you wrote: Does anyone see any potential problem with eventually linking in ssl in such a way as to allow connections to be made securely? I do not know the amount of work this would entail, but this would potentially allow for our IAX traffic to be encrypted too. If

Re: [Asterisk-Users] Parking calls - why doesn't work?

Did you make sure to change the range also? On Thu, 2003-07-31 at 10:50, Dan wrote: > Hi Steven, > > Nope. I have change it to 701, because of some former issues with double > DTMF digits on Cisco 7960. > This is the exact procedure, but on 701 I hear just the invalid extension > message. > What

Re: [Asterisk-Users] Manager

On Thu, 2003-07-31 at 11:24, Steven J. Sobol wrote: > On Thu, 31 Jul 2003, Dan wrote: > > > Hi Roy, > > > > It is not much safer to use SSH to connect to the computer and then > > 'asterisk -r' to the asterisk console? > > I personally would think so. > > I believe the Manager interface is

Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected

Yes. It works even in the macro (not only oin the shell), but because of that warning, the macro exit. The resulting file of the mix is perfect (tried after that), but still that warning. See my previous mail ... testet with a simple 'ls' command. I don't know what to do...:( BR, Dan - Orig

Re: [Asterisk-Users] 'System' application exit with error even if it performs the job as expected

Something even more interesting. I have tried to execute the command 'ls' in the following line: ... exten => s,3,System(ls) ... And this is the result from the console: -- Executing System("SIP/103-2259", "ls") in new stack adsi.confasterisk.conf iax.conf modem.conf oss.

[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #944 - 3 msgs

On Thu, 2003-07-31 at 10:07, Ricardo Villa wrote: There is no way for you to know the vonage password associated with your account. Even if you sniff out the tftp download, its encrypted. Is there any comparable service that isn't as anal? Or even better, is there any service that uses IAX in

Re: [Asterisk-Users] Vonage

The authentication goes out as an MD5 hash I think. You could try to crack it:) Ricardo http://www.telesip.net - Original Message - From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; "Ricardo Villa" <[EMAIL PROTECTED]> Sent: Thursday, July 31, 2003 11:47 AM Subject

RE: [Asterisk-Users] Vonage

www.nufone.net is entirely Asterisk/IAX. -wade > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Steve Meyers > Sent: Thursday, July 31, 2003 12:13 PM > To: Asterisk List > Subject: Re: [Asterisk-Users] Vonage > > On Thu, 2003-07-31 a

Re: [Asterisk-Users] isdn4linux/Teles16.3

> > is it possible to use a Teles16.3 via isdn4linux for the external phone > > connections (phone provider net)? > > Yes, it is. I tested using an old card I had lying around. I quickly > switched to a Fritz card and chan_capi however. This solved the big > issue I had with echo for me. > Since I

Re: [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected

Try to do the same in shell. Does it work ? Martin On Thu, 31 Jul 2003, Dan wrote: > Hi, > > When I try to run the command wmix to mix two WAV files recorded by the > Monitor application I get the following warning in the console and the macro > exit at that point. > Running the command from a s

Re: [Asterisk-Users] Vonage

How about the SIP traffic, where the actual authentication goes? On Thursday 31 July 2003 18:07, Ricardo Villa wrote: > There is no way for you to know the vonage password associated with your > account. Even if you sniff out the tftp download, its encrypted. > > Ricardo > http://www.telesip.net

RE: [Asterisk-Users] Vonage

On Thu, 2003-07-31 at 10:25, nathan wrote: > Iconnecthere (www.iconnecthere.com) works without any problems here, > even behind NAT. I looked into them, but there are a couple of problems with them. First, they don't seem to have numbers in my area. They have my area code, but only for a city th

Re: [Asterisk-Users] Grandstream Budgettone 100 & 102

These guys charge £79+VAT for the 102, and that includes postage to anywhere in the UK. The $75 doesn't include the VAT tax which has to be paid on top if shipping to places like the uk. Tan (telappliant.com) - Original Message - From: "Skuse, Phil" <[EMAIL PROTECTED]> To: <[EMAIL PROTE

Re: [Asterisk-Users] Vonage

iconnect uses SIP and works fine with * Ricardo http://www.telesip.net - Original Message - From: "Steve Meyers" <[EMAIL PROTECTED]> To: "Asterisk List" <[EMAIL PROTECTED]> Sent: Thursday, July 31, 2003 11:13 AM Subject: Re: [Asterisk-Users] Vonage > On Thu, 2003-07-31 at 10:07, Ricardo

RE: [Asterisk-Users] Vonage

> >On Thu, 2003-07-31 at 10:07, Ricardo Villa wrote: >> There is no way for you to know the vonage password associated with >> your account. Even if you sniff out the tftp download, its encrypted. > >Is there any comparable service that isn't as anal? Or even better, is there any service that us

RE: [Asterisk-Users] RFC2833 problems with X-Lite

It WAS broken ;) We fixed it in 1050, if your dtmf is not working properly please do a packet trace and send it to me. -Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lee Goodman Sent: Thursday, July 31, 2003 7:25 AM To: [EMAIL PROTECTED] Subject: Re:

Re: [Asterisk-Users] Manager

On Thu, 31 Jul 2003, Dan wrote: > Hi Roy, > > It is not much safer to use SSH to connect to the computer and then > 'asterisk -r' to the asterisk console? I personally would think so. I believe the Manager interface is supposed to be an interface through which a remote control-panel type pr

[Asterisk-Users] 'System' application exit with error even if it performs the job as expected

Hi, When I try to run the command wmix to mix two WAV files recorded by the Monitor application I get the following warning in the console and the macro exit at that point. Running the command from a standard system console it works. More, even from this macro it works and produce a valid mixed fi

Re: [Asterisk-Users] Problem with the Internet LineJACK ISA card...

Hi Bruce, I tried it now -- I compiled and loaded the module, but still, "No such device" error is given... :'( Any other suggestions? And I failed to checkout the "nsdk" module from the SF CVS... What could be the problem? The command was (lines are truncated here): cvs -d:pserver:[EMAIL PROTEC

Re: [Asterisk-Users] Manager.pm port

On 31 Jul 2003, Steven Critchfield wrote: > If you are running the manager from the webpage, then I can remotely > understand php manager interface. But if you plan on making a command > line manager app, then please do yourself a favor and just help with the > perl stuff. Remember php is perl -1

RE: [Asterisk-Users] Vonage

8x8 is the only one I know (or packet8) a little less "important" regards Humberto Atristain -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Meyers Sent: Jueves, 31 de Julio de 2003 11:13 a.m. To: Asterisk List Subject: Re: [Asterisk-Users] Vo

Re: [Asterisk-Users] Vonage

On Thu, 2003-07-31 at 10:07, Ricardo Villa wrote: > There is no way for you to know the vonage password associated with your > account. Even if you sniff out the tftp download, its encrypted. Is there any comparable service that isn't as anal? Or even better, is there any service that uses IAX i

Re: [Asterisk-Users] Vonage

There is no way for you to know the vonage password associated with your account. Even if you sniff out the tftp download, its encrypted. Ricardo http://www.telesip.net - Original Message - From: "Steve Meyers" <[EMAIL PROTECTED]> To: "Asterisk List" <[EMAIL PROTECTED]> Sent: Thursday,

RE: [Asterisk-Users] RFC2833 problems with X-Lite

Quoting Lee Goodman: > Hi > Someone on the X-lite support list said that RFC2833 is broken in > X-lite (I > ran into the same problem you did). Do you know of a way to turn > off 2833 on > X-lite? I don't no. However I don't see what the benefit would be - if * is set to do in-band DTMF then the R

Re: [Asterisk-Users] Parking calls - why doesn't work?

Hi Steven, Nope. I have change it to 701, because of some former issues with double DTMF digits on Cisco 7960. This is the exact procedure, but on 701 I hear just the invalid extension message. What else can it be? Checked on 2 different systems, one with a Zaptel device, the other one without. Bo

[Asterisk-Users] Vonage

I know this has probably been rehashed a million times, but please bear with me for a little bit... Vonage claims that I can't use their service without having it go through the ATA 186. I see no reason to do that, when I can have Asterisk simply connect directly. Has anyone been successful in s

Re: [Asterisk-Users] Parking calls - why doesn't work?

Look at your parking config. You probably have 700 as your parking extension. Then you will hear what extension the call was parked to. Then you can dial that extension and pick the call back up. On Thu, 2003-07-31 at 07:28, Dan wrote: > Hi, > > I have configured everything as requested (like in

[Asterisk-Users] SIP calls cause Segmentation Fault

I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediatel

Re: [Asterisk-Users] Congestion

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 31 July 2003 15:42, Mark Spencer wrote: > > -- Executing Congestion("Zap/120-1", "") in new stack > > ... But the calling phone just keeps ringing. > What sort of message is it? Well, following some of the examples I'm doing a simple t

Re: [Asterisk-Users] ADSI and SoftKeys

On Wed, Jul 30, 2003 at 05:07:50PM +0200, Armand A. Verstappen wrote: > On Wed, 2003-07-30 at 16:40, John Congdon wrote: > > Has anyone solved the problem on the ADSI phones > > that when you hit one of the soft keys, the Number Pad > > stops working? > > No, I haven't. Just confirming that I have

Re: [Asterisk-Users] RFC2833 problems with X-Lite

Hi Someone on the X-lite support list said that RFC2833 is broken in X-lite (I ran into the same problem you did). Do you know of a way to turn off 2833 on X-lite? Thanks Lee Goodman - Original Message - From: "Jamie Neil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 3

Re: [Asterisk-Users] Problem with the Internet LineJACK ISA card...

Hi Andrei Probably the wrong place to ask that particular question, however... What you probably want to do is grab the nixj driver from cvs on openh323.org. It's a much better driver than the one currently shipped in the kernel. While you're at it, get nsdk too, so you can test the card. Th

[Asterisk-Users] Newbie - Looking for pointers

Hi All, I've been lurking on the list hoping to absorb all the knowledge flitting past while the bits and pieces of my new * server arrive. Well, most of the bits and pieces are here, and I've got the Digium hardware installed and (I think) loaded properly (RedHat 9, Compaq 1850R, T100P, TDM400P 2

Re: [Asterisk-Users] Help with ON-Hold, and call-transfer.

Again, it would be helpful to see the output from "sip debug". I would place this in the bug tracker. Mark On Thu, 31 Jul 2003 [EMAIL PROTECTED] wrote: > Hi, > > OK firstly some background before I ask my questions... Sorry that is a > little extensive, but it saves the standard "What is your s

Re: [Asterisk-Users] RTP codec 13 received - Cisco incompatibility?

Probably needs some more information. I would consider placing a detailed bug report in the bug tracker including the output of "sip debug" with a call going through. Mark On Thu, 31 Jul 2003, Cerrajetto wrote: > Hello, > > In our SIP network, Asterisk is the central PBX, and it routes calls to

Re: [Asterisk-Users] Congestion

What sort of message is it? Mak On Thu, 31 Jul 2003, Tais M. Hansen wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hi, > > When congestion doesn't do anything on a zaptel channel, would it be because > the tones in the zaptel zonedata.c isn't correctly defined? > > -- Nobody pi

Re: [Asterisk-Users] (no subject)

Jeg kan ikke annet enn å si meg enig. Det her er veldig interessant. Eller hva sier dere andre? Særlig den vri-en med å blande bokstavene på nettopp DEN måten ... genialt On Thursday 31 July 2003 15:32, Andrey Katkov wrote: > ðÏÞÅÍÕ ÂÙ É ÎÅÔ? ÷ÏÐÒÏÓ ÔÏÌØËÏ × ÔÏÍ, ËÁË Õ ÔÅÂÑ ÂÕÄÅÔ ÓÏÅÄÉÎÑÔØÓÑ > Ð

[Asterisk-Users] (no subject)

ðÏÞÅÍÕ ÂÙ É ÎÅÔ? ÷ÏÐÒÏÓ ÔÏÌØËÏ × ÔÏÍ, ËÁË Õ ÔÅÂÑ ÂÕÄÅÔ ÓÏÅÄÉÎÑÔØÓÑ ÐÁÎÁÓÏÎÉË Ó ÁÓÔÅÒÉÓËÏÍ. Date: Wed, 30 Jul 2003 20:06:17 +0400 From: Pavel Zheltouhov <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk,ata186 and Panasonic TD1232 Reply-To: [EMAIL PROTECTED] I have Pan

Re: [Asterisk-Users] Manager

AFAICS, the manager is better for automating stuff - easier to parse etc On Thursday 31 July 2003 14:52, Dan wrote: > Hi Roy, > > It is not much safer to use SSH to connect to the computer and then > 'asterisk -r' to the asterisk console? > What else can be done better with the Manager interfa

[Asterisk-Users] Problem with the Internet LineJACK ISA card...

Hi, I'm having problems with setting up my ISA LineJack card on Linux machine... I've done everithing according to documentation available, by compiling the new ixj driver (v1.2.1), loading it, adding device node /dev/phone0 with major number 100 and minor number 0, adding aliases into the /etc/mo

[Asterisk-Users] Sound Quality.

I've been using asterisk for a while, only for dialout from a SIP client over a PRI -> PSTN, this works great. Now I have a need to also dialin to asterisk over the PRI/TDM, I've been testing by creating an extension, and essentially playing back a recording on that extension. If I access the exten

Re: [Asterisk-Users] Manager

Hi Roy, It is not much safer to use SSH to connect to the computer and then 'asterisk -r' to the asterisk console? What else can be done better with the Manager interface than with a regular SSH connection? Thanks, Dan - Original Message - From: "Roy Sigurd Karlsbakk" <[EMAIL PROTEC

[Asterisk-Users] Parking calls - why doesn't work?

Hi, I have configured everything as requested (like in Andy's documentation): - the file parking.conf (parking at '701', parked calls between 702-720) - include the parkedcalls context in my phones context When I try to park a call to 701, I get the message "I'm sorry.. this is not a valid extensi

Re: [Asterisk-Users] Manager

se attached file On Thursday 31 July 2003 14:25, Rattana BIV wrote: > Hi, > > > What can I do with Manager ? > > Is there some docs about it ? > > Rattana -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 2254 5070 (work) +47 9801 3356 (mobile) Computers

[Asterisk-Users] Manager

Hi,     What can I do with Manager ?   Is there some docs about it ?   Rattana

RE: [Asterisk-Users] RFC2833 problems with X-Lite

Quoting myself ;) > Hi, > > I've managed to get X-Lite (v2 build 1050) working pretty well with *, but > am having problems with the DTMF signalling. In case anyone is interested, the settings I use to get X-Lite build 1050 to talk to * are: System Settings > SIP Proxy ---

RE: [Asterisk-Users] MGCP behind NAT

> From: "Humberto Atristain" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] MGCP behind NAT > Date: Wed, 30 Jul 2003 19:07:40 -0500 > Reply-To: [EMAIL PROTECTED] > > My trouble is that the MGCP devices lost the connection with the > asterisk > > > My gateways are ASKE

Re: [Asterisk-Users] Some stats

> > I try to make some statistics about call on Asterisk. Is there > > something who makes it ? > > I will be interesting to have the time of a call and a list of current > > calls. > > Current calls can be found either from the CLI, or from the manager > interface. Completed calls are all in the >

[Asterisk-Users] RFC2833 problems with X-Lite

Hi, I've managed to get X-Lite (v2 build 1050) working pretty well with *, but am having problems with the DTMF signalling. I've used inband signalling with no problems on the uncompressed codecs (G711), but obviously this doesn't work with the compressed ones (GSM). However when I try to use RF

RE: [Asterisk-Users] RTP codec 13 received - Ciscoincompatibilit y?

.. poking head above parapet, venturing correction .. RTP payload type 13 is "comfort noise" viz whereas payload type 19 is "reserved". Maybe Cisco is right ;-) I believe * has a partial implementation of comfort noise but that it's not complet

[Asterisk-Users] unsubscribe

On Wed, 2003-07-23 at 13:55, Michael Manousos wrote: > Hi, > > Is it possible to use 2 B channels simultaneously > with either I4L or CAPI drivers? > We use AVM A1 (Fritz) PCMCIA with I4L driver and > AVM B1 PCMCIA with CAPI driver. > > Thanks, > Michael. > > > > __

[Asterisk-Users] unsubsribe

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RE: [Asterisk-Users] Grandstream Budgettone 100 & 102

We bought two 100's for $75 each, and IIRC they charged an extra $100 or so for shipping to the UK (which seemed a little excessive to me - I asked our finance people to look into it). -Original Message- From: Reed Wade [mailto:[EMAIL PROTECTED] Sent: 31 July 2003 06:08 To: [EMAIL PROTECT

RE: [Asterisk-Users] RTP codec 13 received - Cisco incompatibility?

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but

[Asterisk-Users] Help with ON-Hold, and call-transfer.

Hi, OK firstly some background before I ask my questions... Sorry that is a little extensive, but it saves the standard "What is your setup response" :) I am using the cvs checkout for CVS-07/30/03-11:43:15 (cvs co -D "a fortnight ago") - I tried the one this morning CVS-07/29/03-13:13:06, but it

[Asterisk-Users] RTP codec 13 received - Cisco incompatibility?

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it

[Asterisk-Users] Congestion

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, When congestion doesn't do anything on a zaptel channel, would it be because the tones in the zaptel zonedata.c isn't correctly defined? -- Nobody picked up in 15000 ms -- Executing Congestion("Zap/120-1", "") in new stack ... But the c

Re: [Asterisk-Users] Manager.pm port

At 00:41 31-7-2003 -0500, you wrote: On Wed, 2003-07-30 at 21:59, Steven J. Sobol wrote: > For anyone that cares... > > I am porting James Golovich's Manager.pm over to PHP. I plan on also > doing some documentation which will cover both the Perl and PHP APIs, > which will be almost identical (at l

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