I have made no recent changes to the IAX2 config on my system. Today I
tried a 1800 call and got the below error. Not sure when this started
since only use 800 once in a while. Does anyone know if IAXTEL is
experiencing problems connecting to the 8xx gateway?
7 16:14:54 WARNING[147466]: chan_i
Tim,
It looks interesting.. Are you willing to release the source code?
Robert
Tim Sailer said:
> On Fri, Mar 05, 2004 at 01:29:38PM -0500, Tim Sailer wrote:
>> Since there's not too much out there, I decided to take about 2 hrs and
>> pound something into shape for a simple status for my * serv
Angel Gabriel said:
> I have 5 BT phone lines coming into my office. We use four for
> international calls, and one for local/mobile calls. We have just obtained
> another call carrier, and now we would like to be able to make calls from
> any phone to any carrier, without having to remember what d
Paul Mahler said:
> With record:
>
>
>
> ; Record voice file to /tmp directory
>
> exten => 9000,1,Record(/tmp/asterisk-recording:gsm)
>
> exten => 9000,2,Hangup
>
>
>
> Is there a way to stop recording other than hanging up?
>
>
>
> Thanks!
Press the # key.
Below is from my extensions.conf. It
Comments are inline.
Robert
Jeroen Rikhof said:
> Hello,
>
> Can somebody give me some information about:
>
> 1. How stable Asterisk is?
My experience and from what I have read on the list is that it is very
stable if run on stable hardware and you don't mess with the program code.
If you mess wit
Soren Rathje said:
>>
>> I use ChanIsAvail() to check to see if the phone is connected at the
>> top
>> of the dialplan for that extension. This works for IAX2 and SIP channels
>> but not for MGCP.
>>
>> If you are interested in the actual code I can send it to you from home
>> tonight.
>>
>> Rob
Soren Rathje said:
> - Original Message -
> From: "Olle E. Johansson" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, February 22, 2004 8:52 PM
> Subject: Re: [Asterisk-Users] SIP extension "busy" when not available ??
>
>
>> > Although the current logic does not require a sip
John,
You are now advertising your EMEA company in your signature block. Maybe
I missed an email that explains the EMEA pricing and availability. Could
you please give an update via the list as to the status of your product
availablity, pricing and delivery times in Europe? The ordering procedu
Jim Sneeringer said:
> Whenever an outside number is dialed, Asterisk says "We're sorry. Your
> call
> did can not be completed as dialed. Please check the number and dial again
> or call your attendant to help you." I have tried many configurations,
> but
> let me give the simplest: It fails whe
Matt McIntyre said:
> I am interested in subscribing to a service that will let me dial the
> PSTN in Ireland and am interested in what the community thinks about who
> has the best services available. I would prefer to purchase time in
> blocks of minutes or pay as I go in lieu of having a monthly
[EMAIL PROTECTED] said:
>>
> I like using whisper tones...
>
> recored the file companyname_whisper.gsm and put it in
> /var/lib/asterisk/sounds
>
> Then add the lines to extensions.conf
>
> exten => 0031,1,Dial(SIP/Recp|20|A(companyname_whisper.gsm)r)
>
>
In my implementation of this the file ext
Christian,
Where is a good place to purchase your phones in Germany? I found a
distributor in the UK but maybe just am not looking in the right place for
Germany.
Thanks,
Robert
American Expatriate in Friedrichshafen (Grund oder Entschuldigung für die
englisch)
Christian Stredicke said:
> Sorry,
Tim Sailer said:
> I've looked, poked, and hoped, but I can't seem to make * understand
> the difference between a SIP channel being busy or not being there.
> Both come up as 'busy'. I would expect the unregistered SIP to be seen
> as unavailable. Am I just missing something obvious, again?
>
> Ti
Feedback for the list. I compiled Andy's code. Installation went well
(except for me misspellng something in the dialplan) with no problems.
The Application works great. Will run down Brian's and give it a try too.
Robert
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Asterisk-Users mailing l
Andy,
I would be interested in your Cepstral engine code.
Regards,
Robert
Friedrichshafen, Germany
Andy Powell said:
> lo,
>
> Is there a single central location for code and applications other than
> CVS? I'm talking about code that can't/wont be included in CVS for various
> reasons? Does the wi
Real Player is required. Excellent video/slide presentation.
http://graphics.cs.uni-sb.de/VCORE/recordings.html
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Martin said:
> Hello.
>
> I vaughely remember someone talking about an asterisk implementation at a
> University in germany some months back.
>
> Any other information ?
>
> Regards...Martin
> --
>
http://graphics.cs.uni-sb.de/VoIP/en/index.html
Some of those folks and also from the Uni Stuttgart
Rob Fugina said:
> On Fri, Jan 30, 2004 at 10:48:35PM -0500, John Todd wrote:
>
>
> In the mean time, I've seen references to bug #'s, here on the list and
> in the CVS logs. I've yet to stumble across the bug tracking system,
> though -- can you give me a nudge in the right direction?
>
> Thanx,
I have compiled the zaptel library and zaprtc on a system that gives the
following from "uname -a":
Linux fxx76.mydomain.de 2.4.18-64GB-SMP #1 SMP Wed Mar 27 13:58:12 UTC
2002 i686 unknown
Makefile for zaptel had the following line uncommented:
#
KFLAGS+=-D__SMP__
When doing the "make load"
> - Original Message -
> From: Daniel Bichara
> To: [EMAIL PROTECTED]
> Sent: Saturday, January 24, 2004 4:12 PM
> Subject: Re: [Asterisk-Users] looking for iax termination
>
>
> Hi,
>
> We have termination based on IAX and SIP at Brazil.
>
> Daniel
>
Daniel,
I would be interested in the d
Mike Nash said:
> Hi
>
> I'm trying to configure my Asterisk box to provide a simple sample
> configuration. It's a mandrake 9.1 box, no cards except a sound card.
> The
> config I am trying to achieve is simply one server, with two SIP clients.
>
> Two issues are cropping up - the first, when I s
John Todd said:
>
> Time to dump the Netgear router. That's an unacceptable answer for a
> router vendor to say "Oh, well, for this MAJOR protocol we're going
> to simply corrupt those packets so they're unusable." What!?
>
> JT
> __
OR get an older on
Kannaiyan Natesan said:
> Do they offers, free evening and weekend calls? I get from BT.
> You can get a free 0870 number from http://www.speak2world.com but they
> charge for it.
>
> Kannaiyan
>
Don't think so but sometimes "free" isn't free. Depending on calling
patterns it might actually be low
Kannaiyan Natesan said:
> Have anyone tried to interface BT's Broadband Voice with asterisk?
>
> Kannaiyan
> ___
>
No, and not sure of their rates but http://www.telappliant.com/ has good
rates, voice quality and is easy to interface to Asterisk.
Robert
Info based on how I do it is imbedded below.
Robert
Larry Keyes said:
> I've got two Grandstream phones talking to * and a X100P card going, so
> that
> I can make inbound and outbound calls via the PSTN, and calls from one
> extension to another.
>
> 1. Is there an equivalent to the "more" comman
John Todd said:
>
>
> United States:* +1-800-...
> +1-888-...
> +1-877-...
> +1-866-...
> via: Telesthetic/Local Exchange Carriers of Michigan
>
>
JOhn, Good idea on leaving the code in. I'll do that. Since IAXtel has
8xx dia
John Todd said:
>
> Robert -
>IAX as a protocol is completely dependent on the far-end gateway,
> and not on any specifications you can change. All the gateways at
> the moment only support SIP; none support IAX or IAX2, though
> hopefully that will change since some of them are actually runn
Looks like the list server is really lagging tonight. I found out some
more info so will just post it in a new email with the same subject.
I added: "search => freenum.org" to enum.conf and got a match (SIP
system) when doing the lookup Maybe I overlooked that in the
original instruction
Top posting(sorry) then imbedding the answers to your questions. Otherwise
doesn't make sense.
Thanks for your reply. Sorry it took a while to get the answers. I'm in
Germany and your email came last night just as I was headed to the rack.
Robert
John Todd said:
>>
>>>
>>my sip.conf contains:
>>
John Todd said:
>
> The freenum.org project wants to use your trunks! The freenum.org project
> is an ENUM parallel tree, which has as an eventual goal the distribution
> of ENUM numbering in nations or areas which due to political or other
> issues are not able to get secure, inexpensive, or fun
John Todd said:
>
>...
> Ideas welcome for more text; I may have another timeslot with Allison
> early next week in which there will be some leftover room for
> additional words. Short phrases and meaningful sets of words for
> existing applications are desired; please don't give me words for
> ap
Chris Albertson said:
>
> I'm looking for a service that will accept VOIP calls and
> send them to the PSTN. Or, I should say _another_ service
> that will do this. I don't need the other direction
>
> Currently I'm using IconnectHere and it works, but I get
> complaints of poor audio quality fro
Chandra said:
> i also had the same problem temporarily i solved my problem with both
> outside NAT. u can also do it if both inside NAT. * outside NAT and
> Budgetone behind NAT simply doesn't seem to work. if u ever solve this
> problem please let me know too.
>
> thanks
>
> cm
>
I am able t
John,
Take your discussion off list... It is way off topic. I think you
do yourself more harm than good by responding to these issues on list.
If you want to build confidence in your company then ask your satisfied
customers to reccommend you and give their testimonials regarding your
speedy servic
admin said:
> I work for an interconnect that sells 3com and NEC. When I made this
> project my own and followed through to show my boss, he said, "this is
> going
> to ruin our industry"
>
> If that is the case then so be it. Same with mp3s and the music industry.
> Had they embraced the technol
Philipp von Klitzing said:
> oHi!
>
>> Ladies and Gentlemen, can anyone please help and let me know what is
>> the way to start Asterisk automatically using a cronjob, thanks
>
> http://www.voip-info.org/wiki-Asterisk+administration
>
> Philipp
>
>
Guess maybe I don't leave my system running long
> Morning All,
>
> I have created some virgin forums that I think may relinquish the mailing
> lists from major burdens. Everything is .001 in version and I need help.
>
> I need some advice as far as images and content. I know the project is
> opensource but is content and graphics? If not can
> Hi,
>
> Do the callers in USA dialling from USA Telco lines always have to
> prefix the CITY/AREA code with "1" in order
> To successfully make a call to other USA destinations?
>
>
> I have not been to USA (yet) :)
>
> Ta
> SJ
For comprehensive info by area code (and as pointed out it does
It looks like Mark and others have addressed the development/CVS issues.
We should let their plan be put into effect and give it a chance to work.
Regarding the email list: A number of people have suggested creating more
email lists. I think this is not a good idea because there will be even
more
Philipp,
Good document, my comments are inline with the parts to which they apply.
(and yes, this was a top post, otherwise it wouldn't make sense.)
Robert
> Hi there,
>
> mostly based upon list postings I compiled a couple of administrative
> suggestions on the Wiki page below. I'd be glad to h
Philipp,
Good document, my comments are inline with the parts to which they apply.
(and yes, this was a top post, otherwise it wouldn't make sense.)
Robert
> Hi there,
>
> mostly based upon list postings I compiled a couple of administrative
> suggestions on the Wiki page below. I'd be glad to h
> Sorry 'bout that.
>
> -Original Message-
> From: Kris Edwards [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, January 06, 2004 3:38 AM
> To: '[EMAIL PROTECTED]'
> Subject: Matrix Orbital (usbl LCD or VFD)
>
> This probably isn't practical for anyone other than home users, but I
> would like to
Check http://www.telappliant.com for their VoIP Starter kits or Telephony
Cards sections.
Robert
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello there,
>
> .
> for pointing me at a friendly/knowledgeable UK supplier of such cards.
>
> Any advice would be greatly appreciated: onc
John wrote:
> Hi
>
> This is hard work :) I have read the Asterisk Handbook, BudgeTone User
> Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource
> Pages
> and more.
>
> I am not a linux newbie but am new to Asterisk. I have failed to find any
> docs that explain how to get a very
> Where can I find that Howto? I'm new to Asterisk and am looking for all
> the
> doc I can find.
>
> TIA,
>
> Eric
>
Eric,
You will find at at:
http://members.lycos.co.uk/wipe_out/asterisk/
Robert
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http:
>
> I'm trying to buy a new X100P but
> http://shop.store.yahoo.com/bsdmall/wisifxoin.html
> is failing to check the order
> Anybody knows any other way to purchase it?
>
> Isamar
>
Try http://store.yahoo.com/asteriskpbx/wildcardx100p.html
You won't get the "whopping" 95 cent discount from BSD
> Is that FCC sticker on the back of the phone for real?
>
> A customer could not use his computer while talking on his GS BT102 phone.
> The customer was using a major name wireless keyboard/mouse with his pc.
> The keyboard/mouse stops working if the GS phone is too close.
>
> --
> Bob Knight
> [
> Hallo.
>
> I am living oin Germany and having two ISDN BRI Lines available. Capi
> driver!
>
> I need a Sip Gateway and a H 323 Gateway.
> About H.323, there should be a full implementation of H.450.
>
> Which software is available that gives me a Sip and a H.323 Gateway to
> enter
> my PSTN with
> Hi there,
>
> yesterday I came across the "Vocera Communication Badge" and now I'd like
> to know if anyone here has played with that thing (or even just seen it
> in real life), and if a price tag can be found for this device?
> Too bad they don't use SIP... ;-(
>
> http://www.vocera.com/
> http
> > Still, there seems to be a "you get what you pay for" theme to many of
> today's posts and this clearly applies to support on FWD. Naybe we should
> remove the signature from * that enables FWD to identify * systems :-)
>
That certainly seems the case for today's theme... It is certainly the
r
>
>
> The phone powers up and I can make calls through my Asterisk gateway to
> other endpoints. However the four leds under the keypad are permanently
> illuminated and the backlight slowly flashes on and off. When I pick up
> the handset there is a repeated tone before I get a dial tone.
> I know
>> Message: 11
> From: "Asterisk online forums" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P
> Date: Wed, 24 Dec 2003 11:23:14 -0500
> Reply-To: [EMAIL PROTECTED]
>
> Brian,
>
...
>
> We are looking now to improve GS products and st
From: Brian West <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream Quality Survey :P
Reply-To: [EMAIL PROTECTED]
...
I have 2 of these phones and they work fine for my application. Granted
its not the most intensive use and definatly not the most critical users
> Hi!
>
>> I don't get why people always say dtmfmode=info mine works fine with
>> rfc2833.
>> bkw
>
> Dunno. I tried rfc2833 first, and had exactly the same problem as
> described below with voicemail (but only there). Info then worked just
> fine (as obviously also confirmed by this user here).
>
> On Sat, 2003-12-13 at 16:41, Joe Dennick wrote:
>> I just updated yesterday, but I did a complete rm -Rf for all of the
>> following directories:
>> /usr/src/zaptel
>> /usr/src/zapata
>> /usr/src/libpri
>> /usr/src/asterisk
>>
>> Then I did a new cvs checkout for all four of t
Today I deleted the files in the asterisk, libpri, zaptel directories that
are in /usr/src and did a new CVS checkout (not update). After doing
the "make install"s and starting asterisk the "show version" is the same
as before:
Asterisk CVS-10/09/03-20:33:57 built by [EMAIL PROTECTED] on a i586
> I tried again at runlevel 3 but to no avail.
>
>
> I'm pretty sure I have sufficient horsepower since I'm running on a box
> with
> half gig memory and a speedy CPU.
>
> burak
>
>
I run on a Pentium I /100 Mhz, 32MB RAM with RedHat 9.0 and have no
trouble with voicemail audio or Music On Hold.
> it's a firmware problem on GS, they are working on that but it seems its
> not that simple to make volume higher on the speaker and echo go away,
> anyway 4.26 seems stable for now and with many new features!
>
Miguel,
What are the new Features?
Robert
>> On Fri, Dec 12, 2003 at 01:57:02AM -0500, Brian Capouch wrote:
>>John Brown (CV) wrote:
>> > Hi List,
>> >
>> > Just a quick note that we have cleared all back logs of Grandstream
>> > product. If you have been awaiting shipment, its shipped. Everyone
>> > should be getting tracking number
>
> Yes, I've been having problems as well but had not taken the time to
> diagnose
> the problem. Just did some looking and it appears iaxtel.com has removed
> the iax v1 support. iax2 seems to be working fine.
>
Rich,
That solved the outbound problem.. Thanks for the hint... 800 numbers are
acces
Is anyone other than me having trouble dialing out via IAXTEL? I havn't
changed my config files in weeks but seems that IAXTel calls (800 and FWD)
stopped working in the past week sometime.
Robert
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http:/
>
>
> Hey, surprise! Just discovered it on the web:
>
> http://graphics.cs.uni-sb.de/~rainer/tour.jpg
>
> Mark is going on tour!
>
>
Not sure if this is real info or just a JPG that someone created.
Is Stuttgart a definate date on the 30th? If so, where in Stuttgart??
Robert
Friedrichshafen
> Does anyone have experience using the Netphone SIP phone from Ortena
> Networks (http://www.ortena.com). I contacted them, and they will sell
> me 10 units for 95 euros/unit. At least i -looks- better then the
> Grandstream :-)
>
The phone looks interested and appears to have been on the market
Are you also able to make outgoing calls via Iconnecthere? If so do you
mind posting your config? I tried their 10 minute trial a couple of
months ago but was not able to get a connection.
Thanks,
Robert
> I'm receiving calls on my asterisk server from iconnecthere. My asterisk
> server is be
> Why don't we just add it on the DIgium list server, wouldn't that make
> more sense, to have a single place for all list memberships?
>
> Mark
>
OR even just leave the discussion on asterisk-users... If we create new
lists everytime some people disagree with a topic being on-list then we
will ha
I have a problem where the Background application only seems to work if
one digit is pressed. Extensions with multiple digits just timeout and
asterisk hangs up.
Below is the relevant excerpt from extensions.conf. In this example,
pressing 2 will access the service menu. Then pressing 1 will do
Does anyone know of SIP phone providers (Grandstream in particular) who
are located in Germany (or the EU)
Thanks for any info.
Robert
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Max, That is what worked for me. if you want the MESSAGE button on the GS
to dial the VM then put whatever extension you have defined for VM in the
field "Voice Mail UserID" via the GS Admin Web Interface.
Robert
>
> Hi Folks,
>
> Bit of a newbee here, so please be gentle. :)
>
> I'm trying
> Is anyone running Asterisk under Fedora Core 1
> (http://fedora.redhat.com/)?
> If so, did everything with Asterisk work properly? I'm looking to migrate
> from Red Hat 8.0 to Fedora this week.
>
> Thanks.
>
Interesting question... Since RedHat will in the future have only their
Enterprise versio
>
> I think it is time to start commercial Pro version (not expensive !!!) of
> Asterisk.
> In my company we already made decision to do it, to offer people
> ready-to-go solution. But is is hard to do anykind of such product without
> Digium and Mark's support.
> Mark I think you are very overl
Earlier today someone posted a RFC number related to voice mail.
Unfortunatly I deleted the message so have lost the number and don't see
it yet in Google. Can you please resend that to me?
Thanks,
Robert
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> Cepestral was installed and working within 10 mins of my decision to
> purchase it. It's $30.00 and can be purchased on their web site and
> they give you a download. They have a demp on their website that will
> do text-to-speech and give you a .wav file to download and listen to.
> Download,
> Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net
>
> I've posted my demp weather report Asterisk AGI script at
> http://www.fnords.org/~eric/asterisk/downloads/
>
Eric,
Can you comment on the difference in installation ease for Festival and
Cepstral?
Regards,
Robert
> Hi Robert,
>
> I haven't the HeadSet model but the lan switch model so I can't be of
> any help for you.
>
> Daniel
>
I have the IP10S LAN Switch model too.. Thats why I find it wierd that the
headset setting makes the difference !
Robert
___
A
With some lively IP10S discussions here maybe someone knows about this
issue: I can use the speaker phone ok. However the handset and switch
hook do not seem to work. If I enable "headset" then I can get audio via
the handset but still have to use the speaker phone button to take ot "off
hook".
> Daniel,
>
> the MGCP log you sent shows you sending the digits and asterisk receiving
> them, however after that either nothing happens (infinite digittimeout) or
> you cut the log short. Can you also send some console output with 'mgcp no
> debug' :-) It saves clutter. Maybe a peek at your exten
> On Mon, 2003-11-03 at 16:27, Alastair Maw wrote:
>> On 03/11/03 20:03, Steven Critchfield wrote:
>>
>> > Sounds like you really need a C programmer and get into the guts
>> > of asterisk. Can't get more flexible than having the source code
>> > yourself to do anything you want. You could add your
> Hi ,
>
>
> I even think to avoid using an installer mainly because the installer
> part is bigger that the application himself.
> What do you think?
>
Dan,
I agree that if an installer or registry entries are not needed then it
makes an automated rollout much easier. Also makes it possible
> Besides, even if I didn't have the files ready, I wouldn't use my lovely
> voice for it - I'll go to a recording studio with a professional (talking
> about a production environment) so it's good to know how to do this
> yourself, in case the studio doesn't know how to record them in this
> forma
As far as I know they do only SIP. If your Asterisk box is behind a NAT
firewall then you probably will have problems.
> Hi All,
>
> I have a FWD number and wish to connect it to Asterisk to receive my FWD
> calls.
>
> How I do?
>
> Is it a register in sip.conf or iax.conf?
>
>
> Regards
>
> Dave
> Hi,
>
>> -Original Message-
>> >The portion of extensions.conf is:
>> >exten => 3001,1,Dial(MGCP/aaln1,20)
>>
>> exten => 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20)
>
> Or aaln/1@ should do just fine. However this doesn't explain why there
> is no dialtone on the phone..
>
> Oh, one thou
>>> I've started to write an FAQ for asterisk, available here:
>> http://asterisk.pronto.tv/faq.php
>>
>> Please help me fill it up with the good stuff :)
>
> Why don't you put it here:
> http://www.voip-info.org/tiki-index.php
> and folks can updated/edit online?
>
>
>
Agreed. There is no need
> > Oh, one thought: Did you set your toneconfiguration to Europe or US ? If
> you
> choose custom you need to configure it another way...
>
Florian
The tone config on the phone is set to Europe. Asterisk is USA.. Hmm..
Will change the phone to USA when I get home and see if that makes a
differen
> Citeren rnc Info Lists <[EMAIL PROTECTED]>:
>
>> I have a SwissVoice IP10S but can not seem to get it to have dialtone or
>> dial on *. Calls to or from 3001 don't work.
>
> Were you able to configure the phones through their webinterface ?
>
> You could
I have a SwissVoice IP10S but can not seem to get it to have dialtone or
dial on *. Calls to or from 3001 don't work.
Any ideas are appreciated.
Robert
mgcp.conf is:
[general]
port = 2427
bindaddr = 192.168.0.110
[ip10]
host = 192.168.0.5
context = from-sip
line => aaln/1
The portion of extens
tensions.conf:
>
> exten => _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
>
> This works for me
>
> regards
>
> Miklos
>
>
>
> - Original Message -
> From: "rnc Info Lists" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent:
> Interesting. Someone thinks that a strategic use for * should be off
> this list. Someone thought my FAX modem for * should be off this list.
> However, nobody seems to think a 1000 messages about Grandstream phones
> should be off this list.
>
> Personally I would welcome seeing more of what peo
I have an Asterisk box behind NAT and am trying to connect to Iconnecthere
as was indicated possible earlier. Am getting the following on the
Asterisk console:
-- Executing Dial("SIP/2001-12c8", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
== No one is available to an
> My asterisk server(s) are behind NAT, and I am a customer of Vonage
> (thrice-over), iconnecthere, and Net2Phone.
>
> There are still some rough edges (especially with iconnecthere) but
> overall it is not correct to say that they won't work.
>
> B.
>
Thats great to hear. Can you please share yo
Alexander,
I will be happy to help with the testing but since I am behind NAT am not
sure it will be of much help to you.. I have 2 Grandstream phones and
Asterisk.
Robert
Friedrichshafen, Germany
> Hello All,
>
> We are looking to test interoperability between Asterisk and Nextone
> softswitch.
>...
>> At this moment, Asterisk behind a NAT can't connect to an outside SIP
>> provider. If you put asterisk outside your NAT, your inside clients
>> can connect to Asterisk and Asterisk will be able to connect to your
>> providers.
>
> I suspected this would be the case. The problem is that I ha
> Okay, at the CLi i did a show version and it's still showing the old
> version. What I'm attempting to prevent the overwriting of my already
> established config files and sound files. Any further suggestions?
>
When I did the make on Asterisk the first (and only) time, I had to do
"make sampl
Jarad,
I would be interested in one or 2 of your examples to get an idea of how
to get started.
Thanks,
Robert
Friedrichshafen, Germany
> On Fri, 2003-10-24 at 05:54, WipeOut wrote:
>> First off, can AGI scripts be created using PHP??.. This is where our
>> skills are and since PHP can be run fro
>
>
> My interrest is radio. I'd like to use Asterisk as a N-way audio
> switch between a set of ham radios and to act as a "transcoder" between
> a few of the ham-oriented VOIP systems like IRLP, Echo Lnk, Wires and
> the like.
>
> What got me started was one day I was sailing off the coast of
>
Can anyone please point me toward the source/binary (linux and Win32) for
Gastman??
Robert
> Hi,
>
> The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all
> my machines, no error, no log.
> Although, the CVS version works great on Linux.
>
> Is it anybody who knows how to comp
>...
> Still: When I call my Asterisk box (which has a fixed IP and is located
> within a university network) using X-Lite I get "choppy sound" to say the
> least. In fact I can hear only the first half second of what I am
> supposed to hear followed by permanent silence. Note that this * box has
> Is anyone else having trouble receiving IAXTel calls? I don't know if
> it's my config that's broken or IAXTel that broken. Several people have
> given me their IAXTel numbers and calls to them all fail. I can call
> FWD numbers via IAXTel just fine.
>
> --Eric
>
Eric,
I am having a similar
Ouch.. you hit one of my pet peeves.. See below.. This is not meant to be
a but rather .
Robert
> John Brown (CV) wrote:
>
>>http == hyper text transport protocol
>>
> So are the entries on your hard drive with a .htm or .html extension not
> files? (sorry a little sarcastic I know)
*** Big d
Are you manually updating the mySQL tables or do you have a web app. to do
that?
Robert
> Steve Creel wrote:
>
>>You'll want to #include it. This leaves the burden of the [general] and
>>any static configs on sip.conf but allows the script to blindly write out
>>from the database to sip_additiona
> On Tue, 21 Oct 2003, Low, Adam wrote:
>
>> Maybe I am missing something here but why would it downgrade their
>> network speed to 10mbps, its very rare to find a 100bT switches these
>> days that don't also support 10bT. In a switched ethernet network there
>> would be no performance loss for th
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