That's the extension incoming calls will ring to --
If you use the example below, and sip calls come into the incoming
context, it would go to 99612 instead of s extension. This is great
if you have multiple DIDs and want to handle them differently.
-Original Message-
From: Nitesh
Hi,
I need to have it so that if someone is on their sip phone that any
other attempts to contact that phone will result in a transfer to
voicemail.
Someone mentioned there might be a setting like numofcalls = 1 in the
sip.conf so that only 1 call would every be sent to the sip phone but I
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af James Bean
Sendt: 19. februar 2005 08:14
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Snom phone hint exten question
Hi,
I am sorry to
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorben Jensen
Sent: Saturday, 19 February 2005 6:13 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: SV: [Asterisk-Users] Snom phone hint exten question
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorben Jensen
Sent: Saturday, 19 February 2005 6:13 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: SV: [Asterisk-Users] Snom phone hint exten question
--On February 18, 2005 11:16:11 -0600 [EMAIL PROTECTED] wrote:
On Fri, Feb 18, 2005 at 02:18:37PM +0100, Kurt Bauer wrote:
I just read thru the changelog.txt of the current CVS version and what
catched my eye was the following line: 'Adding Q.SIG switchtype option
to chan_zap' .
But there is no
Unfortunately that did not work, I hard rebooted the snom phone, the bt102
and the asterisk server, the light just stays off, and I tested the LED on
the button as well just to make sure its working
I also added a hint to the outgoing context so when they make an outgoing
call, still no
For anyone playing around with IAXy(S100i) devices, I am making the
following available:
Windows IAXy Provision v1.00
Tony, thanks for this, it was sorely needed! Especially useful when
travelling to an office that has MS only boxes for example.
Have you set the function key on the SNOM to 'Destination' and typed '691'
in the number?
I am sorry, I meant that you have to type 'bt-karen' in the number.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorben Jensen
Sent: Saturday, 19 February 2005 8:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Snom phone hint exten question
Unfortunately
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorben Jensen
Sent: Saturday, 19 February 2005 8:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Snom phone hint exten question
Unfortunately
On 11:33, Sat 19 Feb 05, Stefan Gofferje wrote:
Michiel van Baak schrieb:
Hi,
I read a lot about US providers that can terminate a PSTN
number for you and offer IAX or SIP connectivity.
Does anyone know such a company in The Netherlands ?
I read about Unet. Anyone with experience with them
Also instead of putting a whole bunch of hints in, how might I go about
putting a cluster of SIP extensions in the hint off the PSTN situation?
Could you also maybe throw me a couple of hints what the
exten = 691,1,Macro(stdexten,SIP/bt-karen)
Macro portion I have seen in some examples
snip
[global]
PSTNLine=Zap/g1
AnalogPhone=Zap/g2
[pstn]
exten = s,hint,SIP/bt-karen
exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
exten = s,3,Hangup
;exten = s,5,VoiceMail(u690)
[internal]
exten = i,1,Playback(invalid)
exten =
No its setup in the snom as 691 not bt-karen I will test that now.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorben Jensen
Sent: Saturday, 19 February 2005 8:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorben Jensen
Sent: Saturday, 19 February 2005 8:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Snom phone hint exten question
Have you
I'm trying to configure a 100xp fxo card for the first time but am not able to
get the channel type ZAP recognised
app_dial.c:743 dial_exec: Unable to create channel of type 'Zap'
WHen starting asterisk with -vvvgc i see
[chan_zap.so] = (Zapata Telephony w/PRI)
== Parsing
Hi,
I've just get a 3COM 3102 but is not configured to use SIP protocol. I've
read that I need an NCP PBX from 3com to upgrade to SIP firmware ? Does it's
true ? I must try to upgrade this =)
If someone can help me...
Thanks.
--
Joel
- Original Message -
From: James Bean [EMAIL
I'm trying to configure a 100xp fxo card for the first time but am not able to
get the channel type ZAP recognised
app_dial.c:743 dial_exec: Unable to create channel of type 'Zap'
WHen starting asterisk with -vvvgc i see
[chan_zap.so] = (Zapata Telephony w/PRI)
== Parsing
On Sat, 19 Feb 2005, Joel Vandal wrote:
I've just get a 3COM 3102 but is not configured to use SIP protocol. I've
read that I need an NCP PBX from 3com to upgrade to SIP firmware ? Does it's
true ? I must try to upgrade this =)
On an earlier thread on asterisk-users it sounded like the 3com
Thanks everyone for your feedback, especially Mark. I now have the ALL
the files I need. My order still stands for the $8.00 product from CISCO
but the CP7960 dealer sent me all the files.
Now I will move on to completeing the setup of the TFTP server. Thanks
again
-Original
quote who=beonice
Robert, thank you very much for that informative
write-up. Of course, I now have more questions. The
first is really basic. I thought extension meant
something the caller dials _after_ reaching asterisk.
How come incoming DIDs have to be handled as if they
are extensions?
Some news.
It is not caused by transmission lines, conectors or anything like that.
The telco tecnician just came here and analyzed the circuit and he got no erros!
He sugested me to loop my PRI port in the balum attached in my asterisk box.
And Surprise...
I got the same errors!
The
This is happens because of imperfect HDLC code. I am having the same in
my logs, but quite rare
and on spans which is idle. Therefore this is not an issue with PRIs itself.
I may be wrong, but telco technician checked my PRIs as well, and didn't
find any flaws.
I can tell you more. IT happens
Yes It must.
I have tried to make channels not continuous 1, 11, etc and it doesnt work
[]s
Alex G Robertson wrote:
Hi.
Does anybody know if channels in various spans (in TE410, for example)
must be
contiguous, this way.
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
I had a simular problem receiveing EM Wink dtmf signals from a MITEL sx200
digital. I had a 10% or so problem with astrisk line timing out before all
the digits were dialed.
I tried all the time outs and even messed with the timing of my MITEL
switch.
It turned out that the problems was
Is there a way to make Asterisk wait 3 rings to answer the phone, and
only answer if someone in the house does not?
I just got an idea, perhaps I could set it to answer immediately, and
play a ringing sound, and ring the house phones, then if someone
answers they answer, and if they do not, then
Hi List,
I am a newbie, just came to know about asterisk a few days back while
installing suse 9.2.
I have a question for which I am sorry to say,but I havntread through
all the archives, but AFAIK i didnt get the answer in the archives.
My question is like this.
I am in Nebraska, US.
I have
I read a lot about US providers that can terminate a PSTN
number for you and offer IAX or SIP connectivity.
Does anyone know such a company in The Netherlands ?
I read about Unet. Anyone with experience with them ?
Any information is welcome.
Michiel,
check out www.budgetphone.nl /
Hi,
First i would like to thank the kind people of the list who have
answered my previuos mail, but i am still stuck as asterisk still
crashes upon startup, i have read the install article at
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
and i have search the asterisk archives,
If you are new to VoIP then by all means get phones that can be
controlled via a web page from the phone.
I will say that SNOM has done a great job with their web interface to
the phones. I would praise others, but all I have seen are the Sipura
and the Cisco ATA, both not real intuitive web
Roger Schreiter wrote:
But when dialing a number, I get:
Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to
create/find channel
After I installed my Digium g729 license, I'm trying to place a call
from my Cisco 7960 and I'm receiving the same error:
Feb 19 09:47:06
Hi List!
any body use www.simpletelecom.com?
I subscribe to www.simpletelecom.com for A-Z termination and paid
US$15.00 and US$70.00 via credit card in two days, but my account has
US$15.00 only. I checked my credit card from the bank and they said me
the payment already paid to merchant.
I've
Anyone using the UIP200 with *? I am having
difficulty getting the phone to register and *stay*
registered for more than 4 seconds.
The * console shows the UIP200 registering and records
the user agent. The UIP200 displays station name and
time on its LCD, then 4 seconds pass and it shows #1
Hi,
I'd like to establish way to exchange data between two remote Asterisk
server. Something like call over IAX and send some structured data.
Any advice ?
Regards,
Rob.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
It looks like it's breaking at the iax.conf file. Have
you set up your iax.conf with the registration info
your service provider gave you? It should look
something like this:
register = iaxid:[EMAIL PROTECTED]
So, in my case, I have a line that says
register = myid:[EMAIL PROTECTED]
where myid,
Thanks, Robert. Yes, I _finally_ figured out why I
need multiple extension contexts. I'm now one happy
camper.
Thanks again,
Maya
--- Robert Hajime Lanning
[EMAIL PROTECTED] wrote:
quote who=beonice
Robert, thank you very much for that informative
write-up. Of course, I now have more
- Original Message -
From: Peter Svensson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February 19, 2005 4:36 AM
Subject: Re: [Asterisk-Users] Which PRI card for EuroISDN ?
On Fri, 18 Feb 2005, Robert
Hmmm,
Let's use the single step approach here.
I saw only a few of the prior posts for this problem, so please bear
with me as we start from square one. Also I do not know how the server
is being used and if you can do all these things.
Remember that the hair loss and dental fracturing
Hi Maya,
I dont have a voicepulse account, do i explicitly need one, from the
comments in the iax.conf file i got the impression that if u dont have a
voicepulse account, u should then comment out some of the
sections/context in iax.conf.Am I right or wrong in trying to run
asterisk with out a
I am in Nebraska, US.
I have broadband cable connection at my home. And I have friend and
family in other country.
Using asterisk and some hardware is it possible for me to call to
landlines to other countries. whiout the need to go through or take
any service from say Vonage or any other
I'd like to establish way to exchange data between two remote Asterisk
server. Something like call over IAX and send some structured data.
Any advice ?
I don't know if this could be done thru an IAX call.
What you could do is something like this :
- have a php script on one server that POST
EM is analogue, not digital...
Not true. You can have EM signaling on a T1
CAS interface. Likewise T1 CAS interfaces can also be setup for FXS
and FXO signaling. This is just how the robbed bits
communicate. With EM wink before a remote switch sends a call
to a local switch it send a
Folks,
Here is my problem. I am brand new to Asterisk and infact PBX world,
trying to make IP phone to IP phone talk. Asterisk config files start
form setting up the extentions. I have no clue about the jargon used
in the config files. Is there a place that talks about vr_ready,
agents,
Robert Burcham wrote:
I have seen no responses to my earlier post:
http://lists.digium.com/pipermail/asterisk-users/2005-February/089944.html
and my problem persists. Would someone please share
their configs and firmware versions?
I sent you an email (off-list) the other day with configs
Come to think of it, why don't soft-phones have web interfaces?
If you have a web accessible phone please tell me about it, off-line
too, I need to increase my inventory of phone models.
The snom sofphone they just released as a web interface, just like the real one.
In fact, I think it's a
I have let's say a reception that is comprised of 2 zap extensions and a mobile
phone to dial using ISDN through Capi.
I want to have a delay before starting dialing the mobile phone so that it
rings only when the call has been unanswered for say 25 seconds.
I tried to use
Hi Race,
OK let me start answering ur questions one by one,
1. I am running RedHat 9.0, i just installed it 3 days ago,
2. I am using gcc version 3.2.2 20030222 (Red Hat Linux 3.2.2-5)
3. I am not using NTP.
4. First i i issued make clean then i deleted all the asterisk
directories
If you're going to promote your own company, that is fine, but you
should do it in the -biz list.
If you're going to try hard to sound like you are an objective third
party, next time try to remember to set your email address properly.
At least this kind of astroturfing brings some sample
[EMAIL PROTECTED] wrote:
There is any way to do it or the code has to be modified ?
Not yet, no, but I'm working on this enhancement.
For now you can just use two Dial() statements, one after the other. The
downside to this is that there is a very short moment where no outbound
call will be
Hi List,
I have a Multitech H323 Gateway MVP400 box with 1
phone on port FXO 1.
I have Asterisk ruuning in Fedora Core 3. Both are in
the same network.
But I can't figure what I have to do in Asterisk to
make that box work. What files I have to configure?
Can anyone help me? I will really
The feature that would be most useful to me is some sort
of rating feature on things like reliability, quality, etc...
where individual users could rate each provider. I'm not
for sure how best to handle something like this, but my
biggest problem currently is trying to decide which providers
Digium tech support recommends going with a t1 card and a
channel bank. This is by far the simplest, cheapest and cleanest
solution that I know of.
Jon.
On Friday 18 February 2005 09:21 am, Jim Van Meggelen wrote:
Folks,
In light of all the troubles people report when running more than one
Hi Race,
Seems like question number five did the trick, I just installed asterisk
on another machine with different hardware specs and asterisk is up and
running, what a beauty!!. Man thank u all for the help, I appreciate it.
:-)
Cheers
Edward Stack Buffer Banfa
RADFORMS RESEARCH LABS
JOS,
On Sat, 19 Feb 2005 21:56:44 +0600, Madhawa [EMAIL PROTECTED] wrote:
Hi List!
any body use www.simpletelecom.com?
so anyone here has experience with them? are they a SCAM?
What is it with all the IS THIS A SCAM?! emails lately? I'm almost
starting to wonder if its a single entity with some
Also anyone who says their bank told them the money is gone on a
credit card purchase has to have something wrong with them - just
dispute the charges and problem solved.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leif
Madsen - Independent Asterisk
hi
is the wiki down again?
roy
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Message: 1
Date: Sat, 19 Feb 2005 16:20:31 +0100 (CET)
From: Remco Barende [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dutch VOIP-PSTN provider
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type:
Hi!
if simpletelecom.com is one of best voip service provider, why they
didn't reply me.
I'm waiting for answer(more than 48hrs):(
anyone know contact phone#?
Thanks
/Madhawa
On Sat, 19 Feb 2005 13:15:30 -0500, dean collins [EMAIL PROTECTED] wrote:
Also anyone who says their bank told them
On February 19, 2005 10:56 am, Madhawa wrote:
Hi List!
any body use www.simpletelecom.com?
so anyone here has experience with them? are they a SCAM?
This is -biz material. Spew your is this a scam bullshit there.
-A.
___
Asterisk-Users mailing
Jon,
I've just added the ability to leave feedback on vendors. Having scores
for vendors on voice quality, customer service, etc, is on the wishlist,
though there are quite a few things higher.
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699
is the wiki down again?
It looks like it
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Yes. There are lots of messages in the mailing list archives
regarding this problem, some of them even include things to try. You
didn't see these messages when you searched the mailing list archives?
Alex G Robertson wrote:
Some news.
It is not caused by transmission lines, conectors or
Yes - no go from here
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af Roy Sigurd Karlsbakk
Sendt: 19. februar 2005 19:14
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] wiki down?
hi
is
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af Roy Sigurd Karlsbakk
Sendt: 19. februar 2005 19:14
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] wiki down?
hi
is the wiki down again?
roy
Ain't there any
[EMAIL PROTECTED] wrote:
I have let's say a reception that is comprised of 2 zap extensions and a mobile
phone to dial using ISDN through Capi.
I want to have a delay before starting dialing the mobile phone so that it
rings only when the call has been unanswered for say 25 seconds.
I tried to
Trevor Peirce wrote:
exten = 555,1,Dial(Capi/210699Local/555cell)
exten = 555cell,1,Wait(25)
exten = 555cell,2,Dial(Capi/693555)
I have never done this but I imagine that would solve your problem.
Pretty imaginative! Yes, that should work.
___
I just installed [EMAIL PROTECTED] to see how it works and seems there is a
problem with sounds... I dont hear any announcements or recordings... sounds
are on /var/lib/asterisk/sounds and the logs show this:
-- Created MeetMe conference 1023 for conference '8200'
-- Playing
Well,
I have no idea where the wiki is hosted, but if the wiki needs to be moved
to a more stable location, our hosting facility in Israel is as stable as
you can get. We have 2 circuit running in, BGP4 and an uplink of 4Mbps. I'm
confident it should be enough, no?
Nir S
- Original
Hi all.
I am going to do a simple voting application for a radiostation.
The idea is to have listeners call in to vote on songs.
What I want to do is to take a phonenumer for each song and present the
result on a simple webpage.
Eg.
To vote on song number one, call 555-
To vote on
Hi,
The link for SuSE 9.x startup script is down too !!!
(http://www.leals.com/~mm/asterisk) !
Can someone have this script ?
Can you send me it or post it somewhere ?
Regards,
Fred
Le Samedi 19 Février 2005 19:13, Roy Sigurd Karlsbakk a écrit :
hi
is the wiki down again?
roy
Hi,
with Asterisk 1.0.1 and bri-stuff-0.1.0-RC4a, and two calls already
established on the ISDN BRI, the third call causes scratches in already
running calls and an answer of an unexisting channel:
Ring on unconfigured channel 0/0 span 2
with Asterisk 1.0.5 and bristuff-0.2.0-RC5, this bug
Yes, but he can buy overseas VOIP temination a heck of a lot cheaper than
just calling overseas from Nebraska.
He may also be able to start overseas DIDs that route to his box here in
the states.
Rakesh, if this is what you have in mind, let us know and we'll point you
in the right direction.
Hi,
I'd like to terminate IAX call on PRI interface. What conditions should be
met to be able to send arbitrary caller numbers to called party, so he can
call back to supplied ISDN number (that is different for every IAX user) -
not through PRI interface, but plain ISDN call !!
Thanks in
Wiki is back up.
Between comment SPAM storms, over eager robots ignoring
robots.txt, and mysql issues, it has been an interesting week.
Jim
James H. Thompson[EMAIL PROTECTED]
[EMAIL PROTECTED]
- Original Message -
From:
Roy Sigurd
Karlsbakk
To: Asterisk Users Mailing
Or, he could just sign both ends up with FWD and not have to mess
with this *.
Yes, but he can buy overseas VOIP temination a heck of a lot cheaper than
just calling overseas from Nebraska.
He may also be able to start overseas DIDs that route to his box here in
I have several DIDs (working well) with LiveVoip and I just signed up for
some outbound minutes. Unfortunately they did not send connection
instructions.
I tried:
exten =
_1NXXNXX,2,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN}|60|s)
but I get the error
Feb 19 15:14:09 WARNING[21453]:
[snip]
It's probably made by the same Chinese company that made the cheap GE
and Sanyo cordless phones I am using. Whenver your Sipura (I assume)
sends the refresh message to turn VMWI off, the phone will display this
message for 60 seconds.
Change your refresh to something like
I'd like to terminate IAX call on PRI interface. What conditions should be
met to be able to send arbitrary caller numbers to called party, so he can
call back to supplied ISDN number (that is different for every IAX user) -
not through PRI interface, but plain ISDN call !!
That one
How does he get his offshore relatives into FWD? Nobody said that they have
broadband. Just telephones.
/edg
--On Saturday, February 19, 2005 3:15 PM -0600 Rich Adamson
[EMAIL PROTECTED] wrote:
Or, he could just sign both ends up with FWD and not have to mess
with this *.
Rob,
From a technical point of view, this is no problem. Asterisk can set
any callerid you like, and the PRI can transport it.
The issue is with the provider of the PRI. They may or may not accept
arbitrary callerids on outbound calls. For example, in the UK many
providers will only accept
I received this email link from Macromedia today
http://www.macromedia.com/newsletters/edge/february2005/index.html?sectionIndex=3trackingid=AWQX
They are now selling Breeze video conferencing server by the
minute as an asp, this service is powered by the macromedia communications
server
You have to wait till you get an email from them saying your account
is setup. I had the same problem where my DID was setup before my
outgoing account.
On Sat, 19 Feb 2005 13:16:15 -0800, Ed Greenberg [EMAIL PROTECTED] wrote:
I have several DIDs (working well) with LiveVoip and I just signed
I got a message saying that my account was set up, but did not get
programming instructions.
Were the strings I posted correct?
/edg
--On Saturday, February 19, 2005 4:54 PM -0500 Brian Dingman
[EMAIL PROTECTED] wrote:
You have to wait till you get an email from them saying your account
is
Hi,
I have brought up and asterisk server with a digium T100p. I am
terminating the T1 to a DMS500.
I can make domestic calls fine but when I try to dial international it
gives me an voice error of 77-3 The # you dialed is incorrect please check
the area code and try again. The calls are
Just after some peoples impressions if they have used this phone.
It has 10 function buttons which I am hoping can be individually
programmed for destination to accept hints from asterisk.
Any input would be very much appreciated.
James
___
I installed X100P from DigitNetworks. The system found
the Wildcard X101P and i was able to modprobe zaptel
and wcfxo and ztcfg. then, after i compile asterisk, i
am able to make third party voip calls dialing from
the asterisk cl and using a regular headset. but when
i try to dial out from zap
Okay,
A couple of things could be happening so let's run through a list. Your
questions are a little vague so I shall make my answer also vague.
1. Codec.
Are you allowing for and does the phone support the codec that
the sounds are in? (I.e. do you have a G.729 license for your
Good work Edward.
Sometimes it is not you but the machine. Probably a device driver that
was not kosher.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edward
Banfa
Sent: Saturday, February 19, 2005 1:00 PM
To: Asterisk Users Mailing List -
Hi Race..
In this case, the asterisk|home comes preconfigured with some stuff
different than the asterisk tar file.
I check and the phone supports all mentioned codecs, I also made a test by
using the phone and sjphone to do a live test directly, conversation was
successful using gsm, ulaw and
I have been using simpletelecom.com for over 2 months now to make
outboudn long distance calls, I didn't have any problems what so ever
with them.
To send callerid this is how I do it:
exten = 81NXXNXX,1,SetCallerID(MY NAME 1235551234)
exten =
I don't know - they look kinda lame. I mean, why is their SIP server
seemingly better-routed than their IAX server? In my case, their IAX server
is almost 20ms further away than the SIP one -- seems odd to me.
Think I'll stick with Nufone - very well routed, and only ~15ms away. :)
-d
At 06:37
Grasshopper,
You have your first clue, the live test works.
Do you understand how SIP works? During the INVITE sequence the Asterisk
and the phone trade RTP CODEC information. RTP is the protocol that
actually carries the sounds, SIP only does the handshaking for the call.
A CODEC is what the
This is a very good place to start Race. So if I understand you correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw
might help so I can listen to the prompts right?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Correct.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds
This is a very good place to start Race.
Thx Race!!! I'll try that and post the results in case somebody else has the
same problem.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial
His Luis,
I have the same setup, the only difference is that I am using Quintum
gateway... is there anybody responded with your concern? I am taking too
much time with this project, in case somebody give you a hint, I wonder if I
could have a copy on it for my reference too.
Your kindness is
On Sun, 2005-02-20 at 08:38 +1000, James Bean wrote:
Just after some peoples impressions if they have used this phone.
It has 10 function buttons which I am hoping can be individually
programmed for destination to accept hints from asterisk.
What do you mean by this?
I'm not sure I
Or I can host it.
I have a few colo servers at 1 and 1 with half of terrabyte of monthly
traffic on it. Multiple connections to Tier-1 providers, 12 Gbit total
bandwidth.
I would host it with pleasure.
Nir Simionovich wrote:
Well,
I have no idea where the wiki is hosted, but if the wiki needs
Easily. AGI script + DB.
[EMAIL PROTECTED] wrote:
Hi all.
I am going to do a simple voting application for a radiostation.
The idea is to have listeners call in to vote on songs.
What I want to do is to take a phonenumer for each song and present the
result on a simple webpage.
Eg.
To vote on
1 - 100 of 121 matches
Mail list logo
