On Tue, 2003-12-09 at 22:47, Ralf Illing wrote:
> Hi …
>
>
>
> I already set-up sendmail on another network server thus it would be
> nice to use that one or is sendmail on * server required!?
>
> I had a look in the archive but couldn’t find any information where to
> set the mail server from
Hi,
- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, December 10, 2003 12:54 AM
Subject: Re: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
> - Original Message -
> From: "Andrew Thompson" <[EMAIL PROTECTED
Hi,
- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 11:27 PM
Subject: Re: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
> - Original Message -
> From: "Dan" <[EMAIL PROTECTED]>
> To: "Aste
Depends if you're phone supports it, and you have reinvites etc enables
in *.
-d
At 03:17 PM 12/8/2003, you wrote:
Hi
all,
Has anyone have an idea why, if you capture the files on a Asterisk
network (ex with Ethereal) you always see the communication between the
two sip phones( hard or soft) pas
Just started putting my first * together with a tdm400p and x100p.
Analog phones, xlite and diax I've got working.
Just got Grandstream budgetone-100.
The budgetone registers with * just fine. * accepts the dtmf and dials the
number. The remote phone rings. From there things go south.
The CLI r
If you are in an area code serviced by a SIP provider, you can run
everything over a broadband connection. I would keep one POTS line to
guarantee access to emergency services, though.
While a proprietary system may seem attractive now, wait until it's a couple
of years old and you can't get phon
Hello--
I submitted of extensions.conf that contains my "telemarketer torture"
menus, last week sometime to the mailing list.
I got back a note from the mailing list machinery, stating that it was
too big, and would be subject to approval. No such approval came, I
guess. Either I missed it, or
At 08:45 PM 12/9/2003, you wrote:
Ok - Here is where I am at. I know this topic has been discussed
before, but never a solid answer was set in place. Is anyone aware of
any phones that can put a caller on hold and the caller hear MOH by the
user pressing the hold button. I understand most phones
Hi …
I already set-up sendmail on
another network server thus it would be nice to use that one or is sendmail on * server required!?
I had a look in the archive but couldn’t find any
information where to set the mail server from localhost
to my network server …
Cheers
Ralf
On Tue, 2003-12-09 at 20:48, Michael Rowley wrote:
> Hey guys,
>
> appreciate the input. Here are some thoughts.
>
> ADSI phones are out of the question. This is a business environment, I
> can't worry about my employees not knowing how to forward calls, answer
> calls when away from the mult
Ok - Here is where I am at. I know this topic has been discussed
before, but never a solid answer was set in place. Is anyone aware of
any phones that can put a caller on hold and the caller hear MOH by the
user pressing the hold button. I understand most phones are only muting
the speaker and h
On Tue, 2003-12-09 at 20:34, Matt Lawson wrote:
> I'm using Asterisk to do audio as well as H.263 video over SIP.
> Actually the video works pretty well but I have trouble with the audio.
> I'm wondering if someone can suggest codec/jitter settings or other
> tweaks. The system looks like thi
On Tue, 2003-12-09 at 23:06, Ulexus wrote:
> After having received my brand new SNOM 200 phones and trying to get the
> remote extension monitoring to work, if seems that they use the "SUBSCRIBE"
> and "PUBLISH" SIP methods to do this.
Does Snom 100/105 remote extension monitoring also ?
On Tue, 9 Dec 2003, Steven Critchfield wrote:
> > >> has anyone put 6 of the wildcat X100P cards in one machine?
> > > Read the docs. 2 card maximum sane install.
that 2 cards limit was primarily meant for E400P or T400P, not the X100P
(not sure if it'd be IRQ dependent because both the X100P and
I've been messing around with a "free line" notification where an
extension is dialed for a second when a line becomes available. I
can't seem to get the "h" extension to continue when the local party
hangs up. I've seen references to other people having the same
problem in the list archives, an
On Tue, 2003-12-09 at 19:26, Andrew Kohlsmith wrote:
> > Sounds like a failover should be used.
> > Dial(Sip/inoffice)
> > Dial(IAX2/home_machine)
> > Dial(Zap/g2/${cellnumber})
>
> I was going to pooh-pooh the idea since I don't like the idea of the user
> having to wait for each dial to time ou
On Tue, 2003-12-09 at 19:11, Barton Hodges wrote:
> [EMAIL PROTECTED] wrote:
> > On Tue, 2003-12-09 at 15:18, Michael Rowley wrote:
> >> Hey guys,
> >>
> >> has anyone put 6 of the wildcat X100P cards in one machine?
> >> I am thinking about putting 6 in one machine, what is everyone
> elses
> >>
> I know how to RTFM. I have found no reference yet to how many x100P
> cards I can run in one box. Where did you find that reference? I have
> looked at the manual avaliable from *.org, downloaded the pdf of the
> x100p card tech info, it doesn't say it anywhere.
You won't find it published
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 4:30 PM
Subject: Re: [Asterisk-Users] FXO cards
> On Tue, 2003-12-09 at 15:18, Michael Rowley wrote:
> > Hey guys,
> >
> > has anyone put 6 of the wildcat X100P c
Comments Inline
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Sri
> Sent: Tuesday, December 09, 2003 4:41 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] FXO cards
> This maybe a stupid question. Pardon me.
> I see everyone talking about p
Maybe even just 4 cards. I would like to know where it states that as
well. I have been reading a lot and thought I was ok setting up my first
system with 4 cards.
Thanks,
Dustin
-Original Message-
From: Michael Rowley [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 9:38 PM
To
On 2003-12-09-20:20:12, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
> I have never been able to get DTMF to work through iconnecthere.
> i.e. if you're calling through ich to get voicemail or interact with
> any kind of IVR, be prepared for it not to work. I have never heard
> of anyone else getti
Each card requires it's own IRQ not shared with any other device. That
right there limits the number of cards you can put in a box.
On Tue, 2003-12-09 at 20:37, Michael Rowley wrote:
> Steven,
>
> I know how to RTFM. I have found no reference yet to how many x100P
> cards I can run in one box.
- Original Message -
From: "Michael Rowley" <[EMAIL PROTECTED]>
> So, the docs say no more than 2 x100p cards sane, has anyone done it?
> put 5 or 6 in one box?
I'm using 4 of them, it works.
___
Asterisk-Users mailing list
[EMAIL PROTECT
Hey guys,
appreciate the input. Here are some thoughts.
ADSI phones are out of the question. This is a business environment, I
can't worry about my employees not knowing how to forward calls, answer
calls when away from the multiline phone, and no ADSI phone will handle
multiple lines that I
Steven,
I know how to RTFM. I have found no reference yet to how many x100P
cards I can run in one box. Where did you find that reference? I have
looked at the manual avaliable from *.org, downloaded the pdf of the
x100p card tech info, it doesn't say it anywhere.
'course, I'm just sad as i
Voicepulse Connect $7.99/month Up to 4 calls at a time
Your local telco $30+/month Up to 1 call at a time.
Yes, I do belive you missed something.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Andrew Kohlsmith
> Sent: Tuesday, December 09, 200
I do have 2 X100P for my PSTN ports for inbound calls. The problem is
when both those port are busy, then callers get a busy signal or I have
to muck with caller-waiting from the PSTN line.
What I'd like to do is get a VOIP DID number that is set for my local
calling area. Then configure my PSTN
After having received my brand new SNOM 200 phones and trying to get the
remote extension monitoring to work, if seems that they use the "SUBSCRIBE"
and "PUBLISH" SIP methods to do this.
Further, doing a swift grep of the asterisk code, I don't see anything like
this in Asterisk.
Has anyone he
Nufone, will let you set callerid
On Tue, 9 Dec 2003, Andrew Kohlsmith
wrote:
> > Do you know if I can have my calls show up as any caller ID string that
> > I want?
>
> Generally no; you can set outgoing CID # and the telco will match it up to
> whatever name is registered in their CID datab
- Original Message -
From: "Ariel Batista" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 6:28 PM
Subject: Re: [Asterisk-Users] IpDialog phone issues.
>
> Thank you for explaining this. I just tried it and it did not work! I
> still don't get the digits to b
> Sounds like a failover should be used.
> Dial(Sip/inoffice)
> Dial(IAX2/home_machine)
> Dial(Zap/g2/${cellnumber})
I was going to pooh-pooh the idea since I don't like the idea of the user
having to wait for each dial to time out, but it never occurred to me to
have IAX registration and that *
> You get a SIP router like "SER" all your SIP phones are set to
> use SER as the proxy. SER when it gets a registration will look
> for "good" Asterisk servers and will resend the registration to
> _each_ good Asterisk server.
That would work, but SIP is an evil nasty disgusting little protocol.
I have been playing with Asterisks for about 6 months, and have fallen
in love with it. I have finally taken the plunge and gotten a T100P and
a Zhone Zplex-10B (24FXS) channel bank. I have read everything that I
can find on the mailing list, checked wiki, even searched the web with
google.
> GSM is fairly low bandwidth, but IIRC they are using the GSM FR codec,
> not GSM EFR so sound quality sounds like a mobile phone from the mid
> 90's.
My voice quality with GSM and voicepulse is like a cell call from my
2003-model CDMA phone. It's damned near perfect as far as I'm concerned.
I
> Do you know if I can have my calls show up as any caller ID string that
> I want?
Generally no; you can set outgoing CID # and the telco will match it up to
whatever name is registered in their CID database.
> What about the G.729 codec? From what I've heard it allows you to stuff
> an analog
Hello, I am not familiar with video transmission and didn't know that
Asterisk could also do it. How much bandwith is required for an audio/video
transmission using H.263? Does it require any hardware? Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of M
--- Original Message -
From: "listas iPfone" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 3:16 PM
Subject: Re: [Asterisk-Users] IpDialog phone issues.
> Hi!
>
> I have one ipdialog working well with cvs 10/09 but with latest cvs i
have
> the same problem.
>
>
> You might want to double check that you can find a VOIP provider that
> will give you a DID for your local calling area.
Why do you need to? Get a couple of regular PSTN ports for your inbound
calls.
... unless I'm missing something.
Regards,
Andrew
__
> Non expert reply: Yes it will do what you want. You do
> not need to pay for the DID. "Iconecthere" will let you
> make as many outbound calls as you want for a flat 2.5 cents
> per minute plus a small monthly fee. Higher monthly fees
> buy you prepaid "free" minutes. That $8.00 allows _inb
> I see everyone talking about purchasing the channel bank from ebay.
> 1. As a user who has never used ebay, are these used equipments ?
> 2. Are these reliable in terms of all ports working and all hardware
> intact? 3. Is there a huge price difference between purchasing it from a
> authorized de
> Are the Aastra PTXXX phones ADSI? How compatable are they with asterisk?
> I see some PT450s on eBay for reasonable prices, and I just may be
> tempted to pick one up for fun. Is it worthwhile?
Yes they're ADSI. They work very well with *, to the extent that * has ADSI
support. (meaning *
[EMAIL PROTECTED] wrote:
> On Tue, 2003-12-09 at 15:18, Michael Rowley wrote:
>> Hey guys,
>>
>> has anyone put 6 of the wildcat X100P cards in one machine?
>> I am thinking about putting 6 in one machine, what is everyone
elses
>> experience
>
> Read the docs. 2 card maximum sane install.
Can y
From what I read here:
http://www.globalipsound.com/pdf/gips_iLBC.pdf
iLBC is free and better quality than G.729A, same quality as G.729E and
offers "substantially better quality over congested networks". Its
bandwidth requirements are a little higher (13-15 kbps) but they aren't bad.
Adam Hart
> Adding to my own question, VoicePulse doesn't appear to support G.729.
> Here is their list of supported codecs: GSM, G.711ulaw, G.711alaw,
> ADPCM, ILBC, SPEEX (from
> http://connect.voicepulse.com/specifications.aspx). Does anyone know if
> one of these is as good as G.729?
>
All of thoses bes
> Adding to my own question, VoicePulse doesn't appear to support G.729.
> Here is their list of supported codecs: GSM, G.711ulaw, G.711alaw,
> ADPCM, ILBC, SPEEX (from
> http://connect.voicepulse.com/specifications.aspx). Does anyone know if
> one of these is as good as G.729?
>
All of thoses bes
GSM is fairly low bandwidth, but IIRC they are using the GSM FR codec,
not GSM EFR so sound quality sounds like a mobile phone from the mid
90's.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Carl Youngblood
> Sent: Tuesday, December 09, 2003 7
What about the G.729 codec? From what I've heard it allows you to
stuff an analog call into 8 Kbps. This would give you a theoretical
maximum of 80 simultaneous connections on a 640 Kbps DSL line. I
would expect this to be much lower in practice, say 20 simultaneous
streams, but still, that
Thanks works great
James Schenck
Egraph Design Inc.
Arkansas Online Internet Services
(870) 857-3287
IAXTEL (700) 857-3287
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: Tuesday, December 09, 2003 6:25 PM
To: [
Dial(iax2/[EMAIL PROTECTED]/18708573287)
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> James Schenck
> Sent: Tuesday, December 09, 2003 7:04 PM
> To: Asterisk Users List
> Subject: [Asterisk-Users] Outbound iax dialing to one #
>
>
> What I a
true, I didn't realise that but really, as soon as someone gets the program
they can legally release it for free - (the FAQ after the one you posted).
So one person buys the software and just posts the source on their website.
Also, does the FAQ answer you posted entail that the buyer can give the
On Tue, 2003-12-09 at 19:38, Chris Albertson wrote:
> I just bought a machine with a "micro atx" form factor
> for Asterisk. It has no CDROM, keyboard, or video. just
> RAM and a CPU and a very small disk (4GB) I paid
> $50 each for the CPU, M/B, RAM and case for $200 total.
What were th
What I am trying to do is in the 3rd option dial my cell# thru voicepulse I
just can't figure how to construct the line
[inevans]
exten => s,1,setcallerid(${CALLERID})
exten => s,2,Dial(MGCP/aaln/[EMAIL PROTECTED],10,tr)
exten => s,3,Dial(iax2/[EMAIL PROTECTED]/
Where do I put the # to dial 1870
Thanks for your response.
Chris Albertson wrote:
Non expert reply: Yes it will do what you want. You do
not need to pay for the DID. "Iconecthere" will let you
make as many outbound calls as you want for a flat 2.5 cents
per minute plus a small monthly fee. Higher monthly fees
buy you prepaid
I'm using Asterisk to do audio as well as H.263 video over SIP.
Actually the video works pretty well but I have trouble with the audio.
I'm wondering if someone can suggest codec/jitter settings or other
tweaks. The system looks like this:
Softphone <---ulaw> Asterisk #1 <--IAX (usual
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 5:37 PM
Subject: Re: [Asterisk-Users] IpDialog phone issues.
> On Tue, 2003-12-09 at 15:48, Ariel Batista wrote:
> > - Original Message -
> > From: "Steve
Adam Hart wrote:
Hence why I ask for a company name. Small correction to your post, if it's
distributed to anyone, the source must be available to EVERYONE.
IANAL, but I don't think that's quite accurate. If this person wanted
to, they could only ofter an offer for the source to people who bough
Are the Aastra PTXXX phones ADSI? How compatable are they with asterisk? I
see some PT450s on eBay for reasonable prices, and I just may be tempted to
pick one up for fun. Is it worthwhile?
Thanks!
Pat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Must be your DNS..
[EMAIL PROTECTED] asterisk]# host -a voip-info.org
Trying "voip-info.org"
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 39902
;; flags: qr rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 1, ADDITIONAL: 1
;; QUESTION SECTION:
;voip-info.org. IN ANY
;; ANSWER
Non expert reply: Yes it will do what you want. You do
not need to pay for the DID. "Iconecthere" will let you
make as many outbound calls as you want for a flat 2.5 cents
per minute plus a small monthly fee. Higher monthly fees
buy you prepaid "free" minutes. That $8.00 allows _inbound_
cal
- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 4:27 PM
Subject: Re: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
> - Original Message -
> From: "Dan" <[EMAIL PROTECTED]>
> To: "Asterisk Us
- Original Message -
From: "Carl Youngblood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 4:20 PM
Subject: Re: [Asterisk-Users] VoicePulse for outbound dialing
> > Next get a VOIP service provider to provide you with a PSTN DID
> > (A phone number) VoicePul
On Tue, 2003-12-09 at 15:48, Ariel Batista wrote:
> - Original Message -
> From: "Steven Critchfield" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, December 09, 2003 3:10 PM
> Subject: Re: [Asterisk-Users] IpDialog phone issues.
>
>
> > Do you remember to answer the line
This maybe a stupid question. Pardon me.
I see everyone talking about purchasing the channel bank from ebay.
1. As a user who has never used ebay, are these used equipments ?
2. Are these reliable in terms of all ports working and all hardware intact?
3. Is there a huge price difference betw
I just bought a machine with a "micro atx" form factor
for Asterisk. It has no CDROM, keyboard, or video. just
RAM and a CPU and a very small disk (4GB) I paid
$50 each for the CPU, M/B, RAM and case for $200 total.
OK, so back to your question: Buy a stack of the above
boxes and put three FX
On Tue, 2003-12-09 at 15:18, Michael Rowley wrote:
> Hey guys,
>
> has anyone put 6 of the wildcat X100P cards in one machine?
> I am thinking about putting 6 in one machine, what is everyone elses
> experience
Read the docs. 2 card maximum sane install.
--
Steven Critchfield <[EMAIL PROTECTED
For the last few days I can not resolve voip-info.org from many DNS
servers. It does resolve with some DNS servers but I suspect it may be
related more to caching.
Using the host command:
host -a voip-info.org 130.179.16.23
Trying "voip-info.org"
Using domain server:
Name: 130.179.16.23
Addres
Do not put spaces after the commas.
Mark
On Tue, 9 Dec 2003, Terry Wilson wrote:
> I am trying to get the vmail.cgi script to work with my setup and am
> obviously doing something wrong... I am using voicemail2, and have
> created the mailboxes addmailbox.
>
> Example voicemail.conf
> ...
> [vm
On Tue, 2003-12-09 at 15:06, Andrew Kohlsmith wrote:
> > I'd like to know if there is a way for multiple asterisk servers to
> > share a common SIP and/or IAX registry.
>
> > The setup I imagine would be something like :
> > - several asterisk servers called sip1.isp.com, sip2.isp.com, ...
> > - a
On Tue, 2003-12-09 at 15:00, Andrew Kohlsmith wrote:
> > $500 - T100P card
> > $500 - decent PC
> > $800 - bad ebay day for a channel bank
> > $480 - 16 analog phones at $30 (att 957 with speakerphone)
>
> that $800 will be very good actually if you can find an Adit600 with FXO
> ports... they a
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 3:10 PM
Subject: Re: [Asterisk-Users] IpDialog phone issues.
> Do you remember to answer the line?
I seem not to understand something here!
>
> On Tue, 2003-12-
One more idea. How about this. Only works with SIP,
not ZAP or the others.
You get a SIP router like "SER" all your SIP phones are set to
use SER as the proxy. SER when it gets a registration will look
for "good" Asterisk servers and will resend the registration to
_each_ good Asterisk server.
I am trying to get the vmail.cgi script to work with my setup and am
obviously doing something wrong... I am using voicemail2, and have
created the mailboxes addmailbox.
Example voicemail.conf
...
[vm-group1]
1234 => 1234, Bob Jones, [EMAIL PROTECTED]
...
When logging into vmail.cgi, I log in
- Original Message -
From: "Dan" <[EMAIL PROTECTED]>
To: "Asterisk Users" <[EMAIL PROTECTED]>
Sent: Friday, December 05, 2003 4:59 AM
Subject: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
> Hi all,
>
> A new version (0.9.6) of DIAX is available for download at:
> As usu
Next get a VOIP service provider to provide you with a PSTN DID
(A phone number) VoicePulse will do this for about $8.00/month
pluss outgoing per minute cost. So you get as many incomming lines
as you need and you have zero hardware interface at your site.
(other then your DSL line.)
Can I use this
Hey guys,
has anyone put 6 of the wildcat X100P cards in one machine?
I am thinking about putting 6 in one machine, what is everyone elses
experience
Michael Rowley MD
FP
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman
On Tue, 2003-12-09 at 13:28, Dorian Gray wrote:
> one thing I have been wanting to try is simply undefine CONFIG_DEVFS_FS
> and either create the device nodes manually (a la the Makefile) or in
> devfsd.conf; just haven't had time to do it yet...
>
> cheers
> ++dg
>
You may want to look into wh
I think this is a great solution...I'm doing the same thing.
The only snag I'm running into is finding a VIOP provider that will
provide DID numbers in my local calling area! (Spokane,WA area code:509)
You might want to double check that you can find a VOIP provider that
will give you a DID for
I got 2 'InterStar' boxes - by dsg. Is there any way to use it with Asterisk
or any other OS software? What protocol is it using?
--
Witold Kręcicki (adasi) adasi [at] culm.net
GPG key: 7AE20871
http://www.culm.net
___
Asterisk-Users mailing list
[EMAIL
> I'd like to know if there is a way for multiple asterisk servers to
> share a common SIP and/or IAX registry.
> The setup I imagine would be something like :
> - several asterisk servers called sip1.isp.com, sip2.isp.com, ...
> - a DNS alias sip.isp.com pointing to all the addresses (thus
> pr
> $500 - T100P card
> $500 - decent PC
> $800 - bad ebay day for a channel bank
> $480 - 16 analog phones at $30 (att 957 with speakerphone)
that $800 will be very good actually if you can find an Adit600 with FXO
ports... they are scarce on ebay and always command higher prices.
If he wants di
Hi,
I seem to remember someone once wrote a patch to configure a minmessage
parameter, much like the maxmessage parameter is now, as to avoid empty
messages of 2 seconds being stored in the mailbox by slow people on the other
end ;-)
I cant find this in the bugtracker nor in my archives, but p
On Tuesday 09 December 2003 05:18, Kris Edwards wrote:
> Hello all,
>
> Like many of you I'm sure have done, I'd like to build an * box at
> home for testing/educational purposes (and to eventually have one
> killer fancy schmancy phone system for my SOHO).
>
> If I have two standard phone lines in
Hi!
> great. The only problem is with voicemail2. When I access voicemail the
> voicemail2 will not respond to the digits. I can transfer to any extension
> and Asterisk picks up the digits then. But once in the Voicemail2 program
> it fails? Any Ideas? Here is my configuration in the sip.con
Michael Rowley wrote:
> Hey,
>
> Here is a quesion for you. I am still battling with the phone system
> for my new buisiness.
>
> 6 incoming lines, 1 fax, DSL. 8 phones max, will provably start with
> 5
> to save money.
>
> I was thinking of using Asterisk, but having difficulty finding
> app
Hi!
I have one ipdialog working well with cvs 10/09 but with latest cvs i have
the same problem.
regards
miklos
- Original Message -
From: "Ariel Batista" <[EMAIL PROTECTED]>
To: "Asterisk User List" <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 5:34 PM
Subject: [Asterisk-Users
Do you remember to answer the line?
On Tue, 2003-12-09 at 13:34, Ariel Batista wrote:
> I have gotten this phone to work with SIP configuration. It really sounds
> great. The only problem is with voicemail2. When I access voicemail the
> voicemail2 will not respond to the digits. I can transfe
On Tue, Dec 09, 2003 at 08:40:36AM -0800, Chris Albertson wrote:
It would be a major change to the code but I think what you'd
want to do is have the Asterisk server store _all_ of it's information
in something like a database, The dail plan, SIP registrations,
everything would have to go there
Hi Folks,
I'm setting up my Iaxtel connection now and I'm getting
some annoying warnings
What means:
WARNING[7176]: File chan_iax2.c, Line 436 (iax_error_output): Ignoring
unknown information element 'Unknown IE' (31) of length 4
?
And how can I fix it?
Thanks,
Isamar
_
Hi list,
This morning, my asterisk box (PRI trunk) just stops to make calls.
NOTICE[16401]: File app_dial.c, Line 506 (dial_exec): Unable to
create channel of type 'Zap
without any reason. Now, asterisk doesn't brings up d-channel
DEBUG[11276]: File chan_zap.c, Line 64
Comments on this are welcome. Here is my opinion...
I just went through this. Your office size is not economical.
Actually smaller or larger would be better. Getting a channel bank
and then using only 8 ports is a waste. OK if you have 24 extensions
but 3x to expensive if you only use 8 port
On Tue, 2003-12-09 at 13:02, Michael Rowley wrote:
> Hey,
>
> Here is a quesion for you. I am still battling with the phone system
> for my new buisiness.
>
> 6 incoming lines, 1 fax, DSL. 8 phones max, will provably start with 5
> to save money.
>
> I was thinking of using Asterisk, but hav
Hi,
Citeren Nicolas Bougues <[EMAIL PROTECTED]>:
> > It would be a major change to the code but I think what you'd
> > want to do is have the Asterisk server store _all_ of it's information
> > in something like a database, The dail plan, SIP registrations,
> > everything would have to go there.
what city are you in?
Mike
[EMAIL PROTECTED]
Michael Rowley wrote:
Hey,
Here is a quesion for you. I am still battling with the phone system
for my new buisiness.
6 incoming lines, 1 fax, DSL. 8 phones max, will provably start with 5
to save money.
I was thinking of using Asterisk, but havi
Paulo Mannheimer wrote:
At the same site, DTMF recognition is functioning badly, sometimes
duplicating digits and sometimes totally missing others.
We have checked already /proc/interrups, there is no interrupt being
shared.
FWIW right now this problem is the most vexing one we're facing,
actually
Hello,
I have made some changes/improvements/bug fixes to the Asterisk GUI client I
have written in Perl/TK and have released a second beta version on
sourceforge:
http://sourceforge.net/projects/astguiclient/
Here are the screen shots of the client application running on Linux and
Windows:
htt
Hi Andrew,
Citeren Andrew Thompson <[EMAIL PROTECTED]>:
> > Instead of DIAL(IAX/sip.isp.com) could you not
> > DIAL(IAX/sip1.isp.com&IAX/sip2.isp.com&IAX/sip3.isp.com) to reach a
> similar
> > effect ? (or chain them in different lines so it tries to reach the first
> > one, then the second one if
> 6 incoming lines, 1 fax, DSL. 8 phones max, will provably start with 5
> to save money.
> I had thought of using a channel bank, but what a pain in the ass that
> is becoming. For one, they are expensive, and I then have to buy the
> T1 card for the phone server. I though, why not go with
We are putting together a 4-5 Location Asterisk system - each location
with approximately 40-50 phone users in Dallas/Fort worth Area in Texas.
This project will involve:
1. Setting up these systems installing and configuring Asterisk on Linux
with security
2. Setting up dial plans and iax confi
Happens quite a lot if you have synch problems with the other side. try
checking the clocking of your E1 part.
On Tue, 2003-12-09 at 21:06, Paulo Mannheimer wrote:
> At the same site, DTMF recognition is functioning badly, sometimes
> duplicating digits and sometimes totally missing others.
>
> W
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