On Sun, 2003-11-30 at 06:28, Mark Spencer wrote:
I'm coming to Paris Dec 19. I was wondering if there was any interest in
having an Asterisk get together in Paris sometime near there. Any one out
there interested?
Dam, just used my paid excuse to go to Paris last week. Can't tempt you
to
Hi Steven,
I am happy to contribute to the voicetronix channel section.
Paul.
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Count me and one of my collegue's in.
How long are you staying in Paris ? The 19th might be a bit early for us,
but then again maybe not :)
Zoa.
At 23:28 29/11/2003 -0600, you wrote:
I'm coming to Paris Dec 19. I was wondering if there was any interest in
having an Asterisk get together in
Hi all,
We have an asterisk server running on redhat 9, with a TE410P card
connected to 4 PRIs, on a dual 2.6GHz xeon server with 2GB of RAM. We
also have 10-12 agents using IAX based softphones on winxp. Almost all
of the usage consists of outbound calls done automatically through the
PRIs and
On Fri, 2003-11-28 at 15:26, Ernst Lehmann wrote:
Hi All,
I am a newbie to asterisk, and here is my first problem, where I do not
know any further.
I have to grandstream BT100 connected to asterisk. Working fine, for
calling to each other, and to call via a IAX-Link to the outside.
If
Mark Spencer wrote:
I'm coming to Paris Dec 19. I was wondering if there was any
interest in having an Asterisk get together in Paris sometime near
there. Any one out there interested? Anyone in Paris who could help
organize something like that? :)
Mark
Hello,
I`m having rouble with app_queue. What I would like:
1. calls are distributed between operators via leastrecent strategy
2. when operator is answering talking on the phone with customer he is
not getting those pesky call waiting signals ( really pesky on Granstream)
3. When all operators
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, November 30, 2003 1:02 PM
Subject: RE: [Asterisk-Users] * Party in Paris
Mark Spencer wrote:
I'm coming to Paris Dec 19. I was wondering if there was any
interest in having an
I deleted all the asterisk related directories and their subdirectories
from /usr/src/ and did a brand new check out of zaptel, zapata, libpri,
asterisk-addons and asterisk.
AJ
On Sat, 29 Nov 2003, Tilghman Lesher wrote:
On Saturday 29 November 2003 20:36, [EMAIL PROTECTED] wrote:
Quoting
- What's the console output after the crash when starting asterisk with
-gvvvc?
- After the crash, run a backtrace of the core file and send the output
here
...perhaps this should be on the FAQ?
...and perhaps the FAQ should be linked to from asterisk.org?
roy
On Sunday, Nov 30, 2003, at 14:14
Anton Yurchenko wrote:
also if I build my dialplan like :
exten = 101,1,Answer
exten = 101,2,Queue(phila)
The musionhold plays only until the track is finished, and then it
hangsup. How to make it loop?
--
Anton Yurchenko[EMAIL PROTECTED]
Digital Generation
On Sat, 2003-11-29 at 03:41, Paul Bagyenda wrote:
Hi Steven,
I am happy to contribute to the voicetronix channel section.
If you have any documentation, you can send it to me for inclusion into
the book. I am going to try and find some time to input stuff today,
but if you do not have it
I have several phones running Cisco's 6.0 SIP software release at
this time. Two of the phones have not shown any abnormal behaviors,
but one of them has an unsettling propensity to lock up after several
hours, where the softkey labels disappear and the phone stops
registering, requiring the
count me in
Michael devenijn
DKMA
Van: Mark Spencer [mailto:[EMAIL PROTECTED]
Verzonden: zo 30/11/2003 6:28
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] * Party in Paris
I'm coming to Paris Dec 19. I was
Iain Stevenson wrote:
I've been tinkering with ENUM and found that the lack of a debug message
in enum.c that says it has actually succeeded in resolving an address is
a bit of a nuisance. It makes it difficult to see if failures with ENUM
are due to problems with parsing NAPTR records (in
I think you need to better define your Queue Environment in
extensions.conf. Below is what I've got in mine, and it seems to work
quite well:
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten =
From the console, I see where the call comes in and I can see where the
party from the outside hangs up. The next thing that is said is as
follows:
libgcc_s.so.1 must be installed for pthread_cancel to work.
Now I've taken a look on my system and I do in fact have the libgcc_s.so.1
on my
Cees de Groot [EMAIL PROTECTED] said:
[nothing, he just quoted a whole post]
Sorry, editor slip. I meant to say:
If you are running GS phones then the only codec availible to you is the
G.711, using this codec you will not need all that powerful a server.. A
1Ghz and above processor with
Joe Dennick wrote:
I think you need to better define your Queue Environment in
extensions.conf. Below is what I've got in mine, and it seems to work
quite well:
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
I'll be there until jan 5. The 19th would definitely be too early, maybe
the 20-22? Possibly even after the new year, jan 2 or 3.
Mark
On Sun, 30 Nov 2003, zoa wrote:
Count me and one of my collegue's in.
How long are you staying in Paris ? The 19th might be a bit early for us,
but then
On Sun, Nov 30, 2003 at 01:36:09PM -0600, Mark Spencer wrote:
I'll be there until jan 5. The 19th would definitely be too early, maybe
the 20-22? Possibly even after the new year, jan 2 or 3.
Mark
That's great news, I was sorry I would be abroad on the 19th. I would
definetly be ok for a
Hi,
Two months ago I call Grandstream and I ask the same question... The answer
was NO! In the future will be possible... bla, bla, bla...
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd Wallace
Sent: Saturday, November
Make sure and let us know anytime you are stopping by London.. :)
(Just not between the 2nd-22nd December cos I will be away)
Later..
Mark Spencer wrote:
I'll be there until jan 5. The 19th would definitely be too early, maybe
the 20-22? Possibly even after the new year, jan 2 or 3.
Mark
On
So what's a call for asterisk?
* Something that's set up between two endpoints through the dialplan.
Simple can send messages within a call, like
* A calls B with SIP (INVITE-ACK-ACK)
* B sends a URL to A with SIMPLE within the SIP session
The problem that we have, if I understand Mark, is
Just thought of another, I think an iax_func or iax_custom would be great.
It would act like an rpc and on the other end you'd have a dialplan or
similar to parse and return the result. eg iax_func(fetchvoicemailcount) or
iax_func(getactivecallcount) - this would allow for phones to be very
Mark - I am more than happy to put together whatever you need done.
Obviously could do with knowing how many people, what time of day, do you
want to eat - drink or just somewhere quiet.
Depending how we take this forward can supply full contact details mobile,
personal email address.
I
Ok, so you've read the Wiki and gotten call routing using ENUM to work
(http://www.voip-info.org/tiki-index.php?page=Asterisk%20E164%20Call%20Routing)
with your own ENUM-alike domain, e164.example.com.
But how do you populate it with data? You can do it manually, but that gets
very tedious very
[EMAIL PROTECTED] wrote:
From the console, I see where the call comes in and I can see
where the
party from the outside hangs up. The next thing that is said is as
follows: libgcc_s.so.1 must be installed for pthread_cancel to work.
Now I've taken a look on my system and I do in fact have
Hi.
Isn't possible to have a statically linked version for linux?
[EMAIL PROTECTED] iaxcomm]$ ./iaxcomm
./iaxcomm: error while loading shared libraries: libwx_gtk_xrc-2.4.so: cannot open
shared object file: No such file or directory
[EMAIL PROTECTED] iaxcomm]$
:(
isn't very useful under
Also I must point out that your NAPTR record is a bit wrong:
wrong:(bind9)
!+(.*)!iax2:foofone/1!
Correct:
!\\+(.*)!iax2:foofone/\\1!
Thats how I have it setup.
bkw
On Sun, 30 Nov 2003, William Waites wrote:
Ok, so you've read the Wiki and gotten call routing using ENUM to work
Does anyone know where to get cisco 7960 power suplies? What should they
cost?
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Cisco
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lists
Sent: Monday, 1 December 2003 10:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cisco 7960 power suplies?
Does anyone know where to get cisco 7960 power suplies? What
Anyone have tried * with kerio SIPPS softphone?
It registers ok with *, but
I get missing sdp body message when dialing any extension.
Thanks.
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Steve Kahn and I were having this very discussion the other day on the
iaxclient-devel list. I know that Steve is now aware of it and I believe
he's going to pass the the same suggestion to Mike VanDoselaar. I can't
speak for either Steve or Mike but I think you will probably be seeing it
in
On Sun, 30 Nov 2003 19:09:00 -0500 (EST), [EMAIL PROTECTED]
wrote:
Steve Kahn and I were having this very discussion the other day on the
iaxclient-devel list. I know that Steve is now aware of it and I believe
he's going to pass the the same suggestion to Mike VanDoselaar. I can't
speak for
Hi John-
Over the past couple of years, I've built several calling card platforms for
companies in the UK.
My experience, unfortunately, has been similar to yours with regard to rate
info. Many (most?) phone card companies, like the ones I worked with, end
up dealing with second-tier
You can get them from any cisco reseller.
If you are in the US, the part number is CP-PWR-CUBE=
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lists
Sent: Sunday, November 30, 2003 6:49 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cisco
http://www.expercom.com/product_detail.html?PRODUCT_ID=206301
https://www.buymicro.com/secure/default.cfm?itm_code=411522src=4
http://www.mtmnet.com/CP-PWR-CUBE.htm
http://www.thenerds.net/productpage.asp?un=183483s=1
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax:
[crossposted to isp-clec and asterisk-users]
As part of several larger projects, the question of rate importation
from termination carriers has come up. If a firm has four different
LD partners, as an example, then it is obvious that the firm needs to
determine at the origination of a call
In the following scenario, the user calling from a SIPphone registered
phone is able to transfer the called user to another extension.
sip.conf:
[general]
port = 5060
context = from-sip
register = number:[EMAIL PROTECTED]
extensions.conf:
[from-sip]
exten = s,1,Dial(SIP/111SIP/117)
exten =
Hi I'm currently running asterisk with an fxo X100P and aTDM one port card
in a small not world connected subnet, I've sucessfully setup two sip phone
and one analog extension everything works fine with the analog phone but
when you talk to someone on the sip phone the person at the sip phone can
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