This is my SJphone story, this is why I removed it:
I installed SJphone without too much trouble, I found a voip-info
article on configuring it and tried configuring it. Apparently I failed
to configure it properly since it did not attempt to register to my
asterisk server (in fact, selecting the
Do you remember anything else about the Microsoft thingy from the
developer resources or whatever so I can google for it a bit? Anyway,
not working reliably is not going to stop me, since I really don't
expect it to work reliably! But being able to use my PDA to make an
_TEST_ call would be really
On 5/23/07, Philipp von Klitzing [EMAIL PROTECTED]
wrote:
Hi!
Googling arround I found a number of pocket pc softphones. Of those I
was only able to install SJ-something (removed it).
Is there one (pocket pc softphone) that works?
Windows Mobile 6 comes with a SIP client, however on my HTC
Can you tell me how may i do that?
On 5/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Anselm Martin Hoffmeister [EMAIL PROTECTED]
Date: Tue, 22 May 2007 13:41:43 +0200
Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
Hello,
i just want to activate SMS service between my
Was for [EMAIL PROTECTED]
On 5/23/07, Jonson Player [EMAIL PROTECTED] wrote:
Can you tell me how may i do that?
On 5/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Anselm Martin Hoffmeister [EMAIL PROTECTED]
Date: Tue, 22 May 2007 13:41:43 +0200
Am Dienstag, den 22.05.2007, 13:21 +0300
Dear All,
I have a tiny dial plan like:
[testing]
exten = 454,s,Ringing()
exten = 454,n,Wait(4)
exten = 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10)
exten = 454,n,Hangup
This connects fine when I dial 454 from any extension in my system,
but there is never any audio?
Where can I start to look
Gavin,
Does the Asterisk server's route to 192.168.45.18 traverse a firewall or
router that may be blocking non-SIP ports that are dynamically allocated?
SDP -- part of the SIP INVITE transaction payload -- negotiates arbitrary
ports between the two endpoints for actually passing media.
I tried ... still same errors:
---Cut Here---
May 23 10:56:35 WARNING[31660]: pbx_spool.c:347 scan_service: Unable to open
/var/spool/asterisk/outgoing/smsq.mttx.0.1179906994-32569.1: Permission
denied, deleting
May 23 10:56:35 WARNING[31660]: pbx_spool.c:389 scan_thread: Failed to scan
service
Gommidh Riadh wrote:
Hello,
Did someone have a solution for a line fax detection for outgoing call
Err, if you start your extension definition with answer, then if it
detects a fax signal it will try to redirect to the extension fax.
For exemple
I call number 0123456789
- if it is a
Greetings list,
What are people's experiences with WiFi SIP phones?
When I last looked into them about 18 months ago, they were incredibly
expensive, had very limited range and poor battery life. In the end, it worked
out much more cost effective to simply use ATAs + DECT cordless phones where
Gommidh Riadh wrote:
For exemple
I call number 0123456789
- if it is a fax then redirect to extension A
- if it is a line then redirect to exention B
whats ia want its somthing like AMD application that i use for the
answering machine .
http://www.voip-info.org/wiki/view/NVFaxDetect
On Wednesday 23 May 2007 01:01:53 pm Chris Bagnall wrote:
Greetings list,
What are people's experiences with WiFi SIP phones?
When I last looked into them about 18 months ago, they were incredibly
expensive, had very limited range and poor battery life. In the end, it
worked out much more
Am Dienstag, den 22.05.2007, 20:37 -0500 schrieb Eric ManxPower
Wieling:
David Florella wrote:
Thank you knox. Finally, I have chosen this solution : find
/var/spool/asterisk/voicemail/default/*/Old/ -atime -7|xargs rm –f, executed
every night by the CRON. However, I would have preferred
Hi all, this is my first email to this list. hoping to mingle with you all
from now on.
I have been working on asterisk for about a year. Now my company is planning
to use openser with asterisk having asterisk at backend for features and
voicemail. So is there any book for starters for openser
I'd just like to say that I purchased a siemen S450IP recently and so far so
good it's a nice handset and works better than previous wifi phones I've
used. This is most likely due to it being dect gap where the base station
handles the voip side and not the phone thus avoiding issues with 802.11
In all honesty, things have NOT moved very far since you last saw them.
Battery life has, overall, gotten somewhat better. Range is still
abominable in most of them, and they're not, as a general rule, all that
easy to deal with. We've mucked about with the Linksys WIP3XX series,
the
If I use a call file (/var/spool/asterisk/outgoing)
is it possible to have a video phone connect to a camera (linksys wvc200)
and show the camera stream and hear the audio?
How would I do that?
Jerry
___
--Bandwidth and Colocation provided by
Hi all,
Does someone know if it's possible to use hint function with skinny ?
Can anyone send me an example ?
Thanks in advance, Alex.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
On 23/05/07, Alex Balashov [EMAIL PROTECTED] wrote:
Gavin,
Hi.
Does the Asterisk server's route to 192.168.45.183 traverse a firewall or
router that may be blocking non-SIP ports that are dynamically allocated?
Nope, all internal.
SDP -- part of the SIP INVITE transaction
Hello,
In Asterisk 1.4.4, the SIP channel names (SIP//peer/-/id/) do not seem
to be unique. That means that the id associated to a peer is not random.
Is that normal ? Because other asterisk versions give random id for each
generated SIP channels of a peer.
Regards,
- marc
If you use a computer witha webcam and use Eyebeam for the softphone set
to autoanswer, you should be able to do this.
However, I would tend to use a browser based security camera such as the
DLink camera which would be a simpler setup.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int:
If you use a computer witha webcam and use Eyebeam for the softphone set
to autoanswer, you should be able to do this.
However, I would tend to use a browser based security camera such as the
DLink camera which would be a simpler setup.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int:
Hmm, what am I doing wrong then... Because it says it Unable to
create channel of type 'SIP' (has no route to destination)
How do I setup a SIP trunk? But I think that's not important because I
am on the same subnet and I don't have to make outbound calls...
It's all very vague I think...
Hello
SetCallerPres function seems to be removed from Asterisk 1.4.
What function or application replaced it? Bit of a problem if you want to use
CLIR on your PRI connections.
Jon
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.467 / Virus Database:
Tim,
I have not used DUNDi with SIP, only IAX, but here is what I can tell you
about my config. I have an IAX2 peer named priv, which you should have a SIP
peer like that. When DUNDi does a lookup against server1 from server2 for
the priv mappings and finds a match it creates a a dial string for
All,
Just a quick reminder that *this Sunday* will be the first official meeting
of the
Kansas City Asterisk User Group (http://www.kcaug.net). The location of
the
meeting will be at the Daily Dose Bar Coffee House, in Overland Park.
Future
meetings will continue to be held on the last Sunday
I'm personally interested in one of those VoIP/DECT phones (where the
VoIP is handled by the base and the base-connection is wired) but I
wander if they are better than a standard DECT phone + an ATA (I've
already got two DECT phones pluged into ATA's around the office + 1
@home and I know for
I have commented out the zapateller line now.
The problem persists.
I wonder if there is a problem with the tones generated by some cell phones
when choosing an extension. At this point the problem seems to come from
cell phones only. My wife, for instance was pressing send after choosing an
Jon Schøpzinsky wrote:
Hello
SetCallerPres function seems to be removed from Asterisk 1.4.
I have it in 1.4.4
drdos*CLI core show application setcallerpres
drdos*CLI
-= Info about application 'SetCallerPres' =-
[Synopsis]
Set CallerID Presentation
[Description]
We wanted a cheap last resort fail-over. A few really cheap pots lines
are easy to run buy, as we can get them for a really low cost. My
understanding with DIDs (and its limited), is they have to belong to a
PRI. The only way that is cheaper than a few pots lines is if you needed
8 or more pots
http://www.dlink.com/products/?pid=295
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
Hello,
I'm wandering how can I make voicemail notification when i got a messages in
asterisk mailboxes. For the moment i have e-mail notifications, but I readed
that I can do also a sms notification to local sip accounts. Also I'm
wandering if i can make something like callback from asterisk to
I am trying call transfer with asterisk. blind transfer (#) is working
perfectly, but attended transfer doesn't fonction (*2).
I don't know what is the problem.
Anyone could help?
_
Lancez des recherches en toute sécurité depuis
For of all thanks for the answer, I'll write my comments between your mail.
2007/5/23, Bruce Reeves [EMAIL PROTECTED]:
Tim,
I have not used DUNDi with SIP, only IAX, but here is what I can tell you
about my config. I have an IAX2 peer named priv, which you should have a SIP
peer like that.
http://www.dlink.com/products/?pid=295
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: netconcepts_anguilla at yahoo.com http://lists.digium.com/mailman/listinfo/asterisk-users
Chris,
Sorry for the
Hello, I am trying to install a Billion ISDN on Asterisk
I have Debian Etch and I installed theese packages:
apt-get install linux-headers-`uname -r`
apt-get install make
apt-get install ncurses-base ncurses-bin ncurses-term
apt-get install libncurses5 libncurses5-dev
apt-get install bison
Rob Schall wrote:
My understanding with DIDs (and its limited), is they have
to belong to a PRI.
DID can be delivered over a PRI, a channelized T1 or over analog trunks.
If you use the analog route method, you can get any number of trunks.
The only way that is cheaper than a few pots
?
Membre eBay: zzolivier a crit:
Hello,
I'm wandering how can I make voicemail notification when i got a
messages in
asterisk mailboxes. For the moment i have e-mail notifications, but I
readed
that I can do also a sms notification to
Found the problem.
I thought SetCallerXXX family of applications was retired in 1.4, so I didn't
compile app_setcallerid. But seems that SetCallerId survived, and that
SetCallerPres is located in the app_setcallerid.c
Maybe somebody should move it to its own module.
Jon
-Original
Alex Balashov wrote:
Sure, it's called a DID trunk. It's basically just a regular analog
phone line but the CO switch sends down the digits dialed in one of a
Sean, I am curious--what do these look like these days? Are they
ordinary T1s? CAS/robbed-bit? Do these just use the signaling
Hi all,
I have some peers configured in SIP.CONF file with parameters
incominglimit and outgoinglimit set up to 10. By doing that, I expect
that this peer will not be allowed to handle more than 10 incoming calls
and 10 outgoing calls at the same time.
However, since I upgraded to Asterisk
Here's a silly question, if these are standard POTS you obviously know which
number corresponds to which line, being the case couldn't you tell that ZAP/1
is POTS 555-1234, ZAP/2 is POTS 555-1235, etc etc?
I'm assuming you're trying to identify the inbound number from the telco that
was
Rob Schall wrote:
understanding with DIDs (and its limited), is they have to belong to a
PRI. The only way that is cheaper than a few pots lines is if you needed
This is not true, since analog DID trunks do exist.
I was hoping for a solution more along the lines of Use this x
variable
Are the DECT phones two channel or do they share a channel like most other portable phones?
The thing I like about the wired SIP phones is that they handle the echo issue fairly well. ATAs just reintroduce the echo issue of single pair type phones.
I'd just like to say that I purchased a siemen
Hi,
I was able to work out SIP trunk between Asterisk and CCM 4.x without
any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk.
Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not
replying. For the same reason Asterisk is marking it as UNREACHABLE.
Anybody got
No python code needed . Check .call files at
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
On 23/05/07, Brad Sumrall [EMAIL PROTECTED] wrote:
Can anyone guide me to a how to on automating a call?
I know a little piece of code (normally python) has to be place some where
Hello All,
Has anyone implemented Asterisk behind D-Link Router?
Got one pain in butt customer who wants to setup * system behind D-Link
router model DI-624?
Can anyone share their conf?
Thanks,
Nitesh
___
--Bandwidth and Colocation provided by
On Wed, 23 May 2007, Sean M. Pappalardo said something to this effect:
Give a call to your phone service provider and ask to speak with a
technical group.
I do not share your optimism about the revelation this would entail. :-)
But thank you!
-- Alex
--
Alex Balashov [EMAIL
Hello All,
By any chance anyone has setup Asterisk PBX for big Hotels using Hotel
Management System from Micros System Inc. URL:
http://www.micros.com/Industries/HotelsAndResorts/
Or any other solutions are welcome?
Thanks,
Nitesh
___
--Bandwidth
On Wed, May 23, 2007 at 03:51:36PM +0200, Josu Lazkano wrote:
Hello, I am trying to install a Billion ISDN on Asterisk
I have Debian Etch and I installed theese packages:
Here's one alternative:
apt-get install asterisk asterisk-bristuff zaptel-source
m-a a-i zaptel
genzaptelconf -sdv
Nitesh Divecha wrote:
Hello All,
Has anyone implemented Asterisk behind D-Link Router?
Got one pain in butt customer who wants to setup * system behind
D-Link router model DI-624?
Can anyone share their conf?
Thanks,
Nitesh
___
--Bandwidth and
Hi,
I have installed asterisk-1.4.4 and asterisk-addon-1.4.1.
I followed every step to configure RealTime but something is not working
properly; the warning that I am geting is:
WARNING[32709]: config.c:1229 find_engine: Realtime mapping for
'sippeers' found to engine 'mysql', but the engine
I'm thinking to connet Asterisk software (and if necessary, specific hardware)
to a nortel telephone 3904 model how to read the CDR from incoming call and
use it to query my database how to associate caller id to her notes.
How I can do it ?
--
kaleida
Hello All,
I need to implement a clustered PBX System where parent * is connected
to one of the outbound carrier and other child * will register to parent
*. Reason for this implementation is because some of the child * are
behind NAT. Parent * is on Public IP Address and its connected to
Andre Courchesne - Consultant wrote:
Hi,
Anyone has details or information on how to use the SMS command to
send SMS to Fido, Bell Mobility and Rogers Wireless in Canada?
Bad news on that.
SMS() requires a PSTN port or device in order to work. None of Rogers,
Fido, Bell or Telus provide
On 12:19, Wed 23 May 07, Alexandre VERNIOL wrote:
Hi all,
Does someone know if it's possible to use hint function with skinny ?
Can anyone send me an example ?
Thanks in advance, Alex.
What version of asterisk are you using?
hints on chan_skinny work in -trunk
--
Michiel van Baak
Does the non-Asterisk server _answer_ the line? :)
Gavin Henry wrote:
Dear All,
I have a tiny dial plan like:
[testing]
exten = 454,s,Ringing()
exten = 454,n,Wait(4)
exten = 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10)
exten = 454,n,Hangup
This connects fine when I dial 454 from any extension
On Wed, 2007-05-23 at 19:53 +0530, Vamsi Pottangi wrote:
Hi,
I was able to work out SIP trunk between Asterisk and CCM 4.x without
any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk.
Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not
replying. For the
SIP//peer/-/id/ is so confusion until I realized it was SIP/peer-id with
peer and id made italics...
Mojo
Marc Hurstel wrote:
Hello,
In Asterisk 1.4.4, the SIP channel names (SIP//peer/-/id/) do not seem
to be unique. That means that the id associated to a peer is not
random.
Is that
The 2 most common problems I've seen for no audio in one or both
directions is usually either a firewall (which you already said you don't
have) or a CODEC problem.
Make sure both sides are negotiating the same CODEC. I've often seen
situations where something like the Asterisk server will
Jeremy,
This is the best thing i was able to come up with.
All incoming pots lines go to the zapchans context
[zapchans]
exten = 3,1,Dial(ZAP/1-1);ZAP3
exten = 3,2,Hangup()
exten = 4,1,Dial(ZAP/2-1);ZAP4
exten = 4,2,Hangup()
exten = s,1,Answer()
exten = s,2,Goto(${CHANNEL:4:1},1)
exten
Hey folks,
I have a Digium TE205P working as a man in the middle:
PRI line Asterisk/TE205P PBX
The PBX is a Panasonic KX - TVP 100.
Everything is working great except for one little issue. Asterisk isn't
hanging up the PRI B channel when the PBX channel is hung up.
I
Hi
I have a strange problem.
I use the agi command stream file for my vertical services like *98
If the call comes from a sip phone with dtmfmode=inband in sip.conf then it
works.
But if I call the same script from an external line the stream file doesn't
work properly
The audio is
Hi
While testing I found a solution to my problem. I don't understand it maybe
someone here can explain it.
In my script,
if I call a Playback just before my stream file then everything works ok.
Without the playback then the digits are not captured
I will playback a silence to patch my
Hello All:
I'm having some difficutly getting res_config_mysql from the 1.4.1 addons
package to compile ( I need it for Realtime)
First of all, when I make everything appears to compile ok with no errors
however the res_config_mysql doesn't get compiled. So I tried make
res_config_mysql
All,
I am trying to find a SIP ITSP that honors dial around compensation. We
are adding a Flex ANI code to our outgoing SIP invites by appending an
isup-oli tag to our From: address, like this:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
Tony Plack wrote:
Are the DECT phones two channel or do they share a channel like most
other portable phones?
DECT is a digital standard, quite distantly comparable to GSM. There are
multiple channels (I believe the standard allows for 12 channels, but
the last time I actually worked on DECT is
And yet, it's shorter than your HTML/image-ridden sig. :-)
CP
Wiley Siler wrote:
Disclaimer at the bottom still looks ridiculous even in Spanish… LOL
*Wiley E. Siler
**Director of Information Technology*
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
Hi guys,
During a time, i'm looking for a softphone that work fine in linux,
present good features, good audio quality and good interface.
Recently i found gizmo and i'm testing it with my asterisk box. For my
purpouse he fit perfectly.
I'm worried about security, because i need to inform my
Hello, I want to limit calls per sip account user. How may I realize this
setting? For example I want to limit to 10 min all possible calls from an
account or to limit external calls to 10 min and local call remain
unlimited. Thank you for support guys.
Hi Jonson,
On Wed, 23 May 2007, Jonson Player said something to this effect:
Hello, I want to limit calls per sip account user. How may I realize this
setting? For example I want to limit to 10 min all possible calls from an
account or to limit external calls to 10 min and local call remain
I use the UTStarcom F3000. It's not the best phone but does the job.
- Original Message -
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, May 23, 2007 1:01 PM
Subject: [asterisk-users]
Just because you are paranoid doesn't mean they aren't out to get you!
Frederico Madeira wrote:
Hi guys,
During a time, i'm looking for a softphone that work fine in linux,
present good features, good audio quality and good interface.
Recently i found gizmo and i'm testing it with my
We use this macro, which works quite well:
[macro-checkuservoicemail]
; ${ARG1} - Device extension(s) to check for mail
; Usage
; in main context do exten = 1000,1,Macro(checkuservoicemail,101)
exten = s,1,NoOp(Entering CheckUserVoiceMail for ${MACRO_EXTEN})
exten = s,n,GotoIf($[${MACRO_EXTEN} =
Hi everyone,
We have a Asterisk-based call center deployment with around 40 SIP users,
attending incoming calls from two PRI lines (2xE1) using agents and queues.
The problem is that Asterisk stops accepting new incoming calls to the PRI
lines without reason, although there should be free
I must say that I've VERY happy with my Aastra 4801 CT phones. I think that
they're DECT. Each can have up to six cordless handsets. Technically its a 9
line phone, but if you use G.729
you can only sustain two calls at once. I can have a call on the portable and
easily take another on the
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Gommidh Riadh
Sent: Wednesday, May 23, 2007 3:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fax detection
Gommidh Riadh wrote:
For exemple
I call
On Wed, May 23, 2007 at 05:32:11PM -0300, Frederico Madeira wrote:
Hi guys,
During a time, i'm looking for a softphone that work fine in linux,
present good features, good audio quality and good interface.
Recently i found gizmo and i'm testing it with my asterisk box. For my
purpouse he
We are running 1.2.14 with an IVR in the dialplan.
If I connect to the IVR with a SIP phone (Polycom or Xlite) and press a
couple of digits very rapidly (I found this with 33 on a sticky keypad)
which are an invalid response, Allison will go into a loop saying 'I'm
sorry, that is an invalid
On Wed, 23 May 2007 10:43:04 -0400, in
gmane.comp.telephony.pbx.asterisk.user you wrote:
Has anyone implemented Asterisk behind D-Link Router?
Got one pain in butt customer who wants to setup * system behind D-Link
router model DI-624?
sip.conf:
[general]
externip =
Frederico,
Gizmo Project is a US company.
I hate to tell you, but under US law, they HAVE to be able to record not
only detailed CDRs about your call, but also your DTMF codes and full
conversations. Now, this would only be in the case of a warrant issued
by the US Government to retrieve
On 5/23/07, Michael Graves [EMAIL PROTECTED] wrote:
I must say that I've VERY happy with my Aastra 4801 CT phones. I think that
they're DECT. Each can have up to six cordless handsets. Technically its a 9
line phone, but if you use G.729 you can only sustain two calls at once. I
can have a call
Hi All,
I have asterisk 1.4 up and running. My next target is to install NVFAXDETECT on
my box.
Does anyone has detail steps on how to install NVFAXDETECT on Asterisk 1.4 ?
In case i mess up with the current asterisk installation, how do I roll back or
unstall
NVFAXDETECT ?
Regards
I travel a lot for work. I frequently find hotels that have wifi, free or
otherwise available. But I've yet to find it anywhere near sufficient to
support voip applications. At least not good enough to
compel me to not use my cell phone. If you have control of the host LAN then
you can ensure
Hi all,
I'm having a problem with an asterisk server being unable to call certain
cellphones and answering machines. Anytime the person answers the phone
call, everything works well. But when the call goes to voicemail or an
answering machine, I get the error message below:
I am trying to configure a new server for use in a small Call Center.
I want to use realtime queues and agents and after following the
instructions I can get the queue to show up on the system but no agents.
I am using Asterisk 1.4.4 on a CentOS 5 machine.
I have this in extconfig.conf:
Try a Nokia E61/E62... Version 3 supports SIP and WiFi and they have a big
battery that allows long talking and standby times.
CS
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Justin Moore
Gesendet: Donnerstag, 24. Mai 2007 10:16
An: Asterisk
I have a recent dual gsm /wifi from e28 via Skyvoice. (http://myskyvoice.com/)
Its built to use voip or gsm and is about the same
price as existing wifi phones. My main hassle is it doesn' yet do WPA - WEP's
okay and they say WPA is only a firmware load away ;-)
, and it has a browser to login
I work for ABP Technology and lurk on this list so I hope I'm not breaking
any taboos...
ABP is now carrying a dual GSM/Wifi phone. We tested 2 models, 1 had
Windows-CE on it. Some reason we only have the Non-CE version public right
now.
http://www.abptech.com/products/Pirelli/DPL10.html
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Remco Post
Sent: Wednesday, May 23, 2007 10:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] WiFi SIP phones
Tony Plack wrote:
Are the
90 matches
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