True. I think Digium's USA bias is clearly demonstrated by their lack
of a BRI ISDN product. Most of the rest of the world use it in
abudnace yet Digium do not see fit to service this market because it
is not big in the US. very poor...
On Thu, 31 Mar 2005 18:32:40 +0900 (JST), Isamar Maia
Hi all,
Just got a brand new server and a Digium TE410P. I get the sequential
(knight rider) lights before loading the zaptel driver. As soon as I
load the driver all loghts go off. It appears the card is not
generating interrupts.
[EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0
0:
Just out of interest, what BRI card are you with Asterisk?
On Sun, 26 Dec 2004 16:26:37 +1100, Shaun Ewing [EMAIL PROTECTED] wrote:
On Fri, 24 Dec 2004 12:17:50 +1000, Gary [EMAIL PROTECTED] wrote:
Hi folks,
this is specifically directed to Australia Asterisk users..
We are having
Hi all,
Just got a brand new server and a Digium TE410P. I get the sequential
(knight rider) lights before loading the zaptel driver. As soon as I
load the driver all loghts go off. It appears the card is not
generating interrupts.
[EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0
0:
And I thought it was just me going crazy. I have the exact same issue
on a HP-Compaq DL360 G4 server (1U rackmount version). I have tried
everything that has been mentioned here and more. Even replaced the
TE410P card (so know it's not the card). I have tried with FC2, FC3
and RHEL 3. Have tried
,
Adam
On Tue, 2005-01-04 at 17:05 +1100, Eric Bishop wrote:
And I thought it was just me going crazy. I have the exact same issue
on a HP-Compaq DL360 G4 server (1U rackmount version). I have tried
everything that has been mentioned here and more. Even replaced the
TE410P card (so know
I will certainly try that. Please also let me know your progress..
On Tue, 4 Jan 2005 22:12:23 +0200 (SAST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Mon, 3 Jan 2005 [EMAIL PROTECTED] wrote:
Has anyone had success using a TE410P card in an HP-Compaq DL380 G4
server?
Thanks
Well it's clear now that this is not an isolated issue. Has anyone
been in touch with Digium about this issue? I have logged a support
issue with them, but thus far have not received a response. Anyone
had better luck with Digium support and the Compaq/HP G4 server
series?
On Wed, 5 Jan 2005
The problem definately seems to be G4 server related, most folks with
G3's seem to be unaffected. Anyone know where there might be a
changelog of G3-G4? That might reveal some pertinent information.
On Thu, 6 Jan 2005 16:53:09 +0200 (SAST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Thu,
Is it a HP-Compaq DLXXX G4 machine? Because there is a thread here
about the TE410P not generating interrupts with these servers...
On Sat, 8 Jan 2005 05:14:39 -0800 (PST), Sid [EMAIL PROTECTED] wrote:
Hi Scott, and Jack,
--- Scott Stingel [EMAIL PROTECTED] wrote:
Sid-
Try connecting
Hi Peter,
Basically they told me that they have several people complaining of
the problem with G4 series servers and they their hardware engineers
are going to order some of these servers and look into it. Currenly
the only solution they have is to use a different motherboard.
On Fri, 14
I logged a support issue with HP and their response was that it's not
their server that is the problem and if other cards show interrupts
(which they do) there's nothing more they can do
On Fri, 14 Jan 2005 16:30:25 +1000, Joshua McAdam [EMAIL PROTECTED] wrote:
Has anyone logged a support
It's most definately something to do with the G4 series both DL360 and
DL380. Most G3 series owners are reporting it working OK.
On Fri, 14 Jan 2005 20:50:40 +1100, Adam Goryachev
[EMAIL PROTECTED] wrote:
On Fri, 2005-01-14 at 09:23 +, Steve Hanselman wrote:
Has anyone also logged a
Yup, I found their support very unhelpful and unwilling to go the
extra (or even the first) mile..
On Sat, 15 Jan 2005 18:27:49 +1300, Matt Riddell
[EMAIL PROTECTED] wrote:
Eric Bishop wrote:
I logged a support issue with HP and their response was that it's not
their server
I had the same issue. did you ever find a solution. The Fritz card
worked fine with FC2, but no go with FC3, I think it has to do with
udev.
On Thu, 06 Jan 2005 19:36:17 +0100, Bruno Hertz [EMAIL PROTECTED] wrote:
Though you probably won't use them, I'd still like to mention fyi that
I too had the exact same issue today with FC2 and both stock and
vanilla 2.6.9 kernels... still remains unresolved. I think it could be
a broken CVS -stable..
On Mon, 17 Jan 2005 13:29:58 -0500, David Petruzzella
[EMAIL PROTECTED] wrote:
I am unable to compile the zaptel drivers on
Can I ask what BIOS version you are running? Also what MPS Table Mode
are you using (in the BIOS Advanced Options). I have thus far been
getting no interrupts on either a G3 or G4 DL360.
On Mon, 3 Jan 2005 14:14:56 -0500, Karl H. Putz [EMAIL PROTECTED] wrote:
-Original
it going and got it going
would you also be kind enough to tell us how?
On Mon, 17 Jan 2005 08:14:07 +1030, Peter Childs
[EMAIL PROTECTED] wrote:
Thanks for that. I'll keep an eye on the list, and cross my fingers :)
Cheers,
Peter
-Original Message-
From: Eric Bishop [mailto
Are these the G3 or G4 series?
On Mon, 03 Jan 2005 11:52:09 -0800, Scott Stingel [EMAIL PROTECTED] wrote:
Hi Steve-
My customer has five DL320's and ten DL360's - all running asterisk with
TE410P's with no problems - no 380's though. I'm using Fedora core 1
and 2.4.xxx kernel..
I'm
?
On Tue, 18 Jan 2005 23:38:01 +1100, Eric Bishop [EMAIL PROTECTED] wrote:
Can anybody who has got a working DL360 G3 or G4 with the TE410P
showing interrupts please be kind enough to post to the list their
BIOS version and settings as well as their zaptel.conf and
zapata.conf. Also useful
Has anyone had any luck with this issue and new Asterisk/Zaptel
releases (1.05/1.04)? I am still searching for a solution and waiting
for that Eureka! moment..
On Thu, 20 Jan 2005 09:20:09 +0100, Tais M. Hansen [EMAIL PROTECTED] wrote:
On Wednesday 19 January 2005 23:15, Eric Bishop wrote
Did anyone get anywhere with this thread? Any HP G4 series servers working?
On Wed, 26 Jan 2005 09:46:31 +1100, Eric Bishop [EMAIL PROTECTED] wrote:
Has anyone had any luck with this issue and new Asterisk/Zaptel
releases (1.05/1.04)? I am still searching for a solution and waiting
Do you have a config sample on how to handle digital PPP sessions in Asterisk?
On Sat, 29 Jan 2005 15:16:51 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] wrote:
On Sat, 29 Jan 2005, David Norton wrote:
Currently I only have 1 PRI which I am using for dial-in customers. The line
is
Can any give me or point me to a short and simple explanation of what HDLC is?
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Would you show us your Sipura 3000 setup? I've got mine passwing call
to Asterisk and all I just can't get it pass through caller ID (I
always get the sipura's extension as the caller ID)..
On Mon, 31 Jan 2005 22:18:31 +1000, Peter Illmayer [EMAIL PROTECTED] wrote:
Nathan
If you want
Has anyone implemented hot seating in any neat way? This where
people can log in to any phone in the company and have their
calls/voicemail come to that particular handset.
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Would you be kind enough to share your Sipura 3000 setup? I've got
mine passing calls
to and from Asterisk, I just can't get it pass AU caller ID (I always
get the sipura's extension as the caller ID)..
On Mon, 31 Jan 2005 22:18:31 +1000, Peter Illmayer [EMAIL PROTECTED] wrote:
Nathan
If
Hi All,
I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and
out of Asterisk. I am using PSTN Ring Thru Line 1 (on the PSTN
Line tab) so Asterisk answers the call rather than the SPA-3000. It
is all working perfectly except I can't get the SPA-3000 to pass
caller ID to Asterisk.
Hi All,
I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and
out of Asterisk. I am using PSTN Ring Thru Line 1 (on the PSTN
Line tab) so Asterisk answers the call rather than the SPA-3000. It
is all working perfectly except I can't get the SPA-3000 to pass
caller ID to Asterisk.
)
-- Original Message ---
From: Eric Bishop [EMAIL PROTECTED]
To: Peter Illmayer [EMAIL PROTECTED]
Sent: Sat, 5 Feb 2005 15:38:47 +1100
Subject: Re: Fw: [Asterisk-Users] AU caller ID with Sipura SPA-3000
Hi Peter,
The caller number I get into Asterisk is the username
Hi all,
Can someone give me a simple rational explanation why a $5 analog
handset gives me no echo whatsoever on an analog PSTN line, but
PSTN-VoIP devices such as the TDM400 and Sipuras do and thus require
software-based echo cancellation. Surely a $5 analog handset does not
have an echo
OK I understand that the $5 handset may indeed have an echo but that
it occurs so fast that it is not preceived as an echo. I pose the
following questions:
1. Is the echo (regardless of it's speed) a side effect of long
distance communications or is it there by design for some technical
purpose?
Just out of interest,
When echo occurs (the type where I hear myself echoing as I talk) what
is bouncing against. Is it the other caller's equipment, the central
office or something in between?
On Fri, 11 Feb 2005 02:53:03 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Fri, 2005-02-11
Hi all,
I have call recording enabled via the Monitor command and it seems,
the call stops being recorded after the call is transferred. Is this
normal behavior? If so how can I continue recording of calls after
they have been trasnferred
thanks...
Hi all,
I have call recording enabled via the Monitor command and it seems,
the call stops being recorded after the call is transferred. Is this
normal behavior? If so how can I continue recording of calls after
they have been trasnferred
___
Hi all,
I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the same thing or is there some technical
difference. Even Newton's telco dictonary seemed a bit fuzzy on this
topic. I have seen it
Hi all,
I have call recording enabled via the Monitor command and it seems,
the call stops being recorded after the call is transferred. Is this
normal behavior? If so how can I continue recording of calls after
they have been transferred
___
heard from Eric Bishop (on the 1st march) was that he had
received an updated card from digium, but it didn't function in his DL380...
I can let you know the outcome of the test if you'd like.
Cheers,
Peter
-Original Message-
From: Mark F. Vickers [mailto:[EMAIL PROTECTED
Hi All,
I have a problem (at least I think it's a problem) where the wcfxs
module causes constant CPU usage spikes. The card being used is a
Digium Wildcard TDM400P with 3 FXO modules and 1 FXS (TDM31B).
Monitoring my otherwise idle asterisk box (with top) I see once every
3-5 seconds hi
. Let me know if you're
interested in using your system as a guinea pig.
Cheers,
Jim.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric Bishop
Sent: December 15, 2004 5:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] wcfxs
I checked out issue 2901 at http://bugs.digium.com . The response
from Digium simply says not a bug and the issue was closed. So are
we to assume this is normal behaviour?
On Wed, 15 Dec 2004 19:27:04 -0700, Michael Welter [EMAIL PROTECTED] wrote:
Richard Scobie wrote:
I'm really discouraged
This is not fantastic tech support from Digium!
This is a serious issue with their product and so far has not been
addressed for about a month (that we know of).
On Wed, 15 Dec 2004 22:02:41 -0500, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
On December 15, 2004 09:50 pm, Eric Bishop wrote:
I
Hi All,
Can someone explain to me the difference between g711's ulaw and alaw
codecs? Is it just different header info or is the actual payload in
each encoded differently? I have thus far noe been able to find any
difinative information onthe matter. All I've managed to find out
that they are
Can anyone confirm whether other Digium cards/drivers especially the
Wildcard TE410P have sililar problems?
On Thu, 16 Dec 2004 07:06:07 -0700, Michael Welter [EMAIL PROTECTED] wrote:
Andrew Kohlsmith wrote:
On December 15, 2004 09:27 pm, Michael Welter wrote:
Yes? Is there a workaround,
I'm not sure if it is the CPU spikes or not but there definately is a
high level of flakiness with the card. ie very low gain levels and
often for no reason will refuse to accept calls until the driver is
reloaded. Honestly at this time I am recommedning all my customers
with Analog lines to stick
Does any body know if it is possible to group SIP channels just like
it is possible with Zap channels? I have a group of FXO gateways
(Sipura 3000's if you must know) and I would like to treat them as a
group the same as I would Zap channels. IS this possible?
Does any body know if it is possible to group SIP channels just like
it is possible with Zap channels? I have a group of FXO gateways
(Sipura 3000's if you must know) and I would like to treat them as a
group the same as I would Zap channels. IS this possible?
Hi all,
Information on this topic seems a little scarce, so I thought I'd try
the list
Apart from the the coolness factor can anyone explain to me in what
situation one would use TDMoE rather than IAX for communication
betwwen 2 Asterisk servers?
Does any body know if it is possible to group SIP channels just like
it is possible with Zap channels? I have a group of FXO gateways
(Sipura 3000's) and I would like to treat them as a group the same as
I would Zap channels. Does anyone know if this is this possible?
however utilize the local channel
to accomplish something like rollover. Check out forking in the wiki
On Tue, 21 Dec 2004 17:26:18 +1100, Eric Bishop [EMAIL PROTECTED] wrote:
Does any body know if it is possible to group SIP channels just like
it is possible with Zap channels? I have
Hi All,
Is it possible to match caller ID on incoming calls against say text
file of know numbers and diaplay the name rather than the numerical
caller ID?
I know some handsets such as the SNOM 190 can do this from within the
handset, but I would like it done and updated centrally at the
the second one and set the CLI
to area name number
Doing this has no apparent delay to the call.
Peter
-Original Message-
From: Eric Bishop [mailto:[EMAIL PROTECTED]
Sent: 22 December 2004 06:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users
Do you have step by step instructions on how you created these RPMs. I
would like to create a few of my own but compiled for my own custom
kernel and patchea and am not very familiar with RPM packagingOn 1/27/06, Andrew McRory [EMAIL PROTECTED]
wrote:Available in the usual place.
Kevin,
I have experienced the same issue. I get worse echo with the VPM
installed than with software EC. Have had it at 2 different sites with
2 different TE411P's.
- EricOn 2/6/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Stagg Shelton wrote: I just implemented a system using a TE411P hardware
Or perhaps slow them down or pipe to a file?
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Nope only bad feedback here. The software EC in Asterisk worked much better for me than did the VPM on the TE411P.On 2/13/06, Isaac Xiao (KVB Kunlun Pty Limited)
[EMAIL PROTECTED] wrote:
What version of Asterisk and Zaptel you were using? Did
you try latest Asterisk 1.2.4 and Zaptel
Hi All,
I have a Digium card in my Asterisk server configured as pri_net and I
want to introduce latency on it in order to simulate PSTN conditions
and test some echo canceller hardware. Is it possible to purposefully
introduce latency and echo in a controlled fashion in order to do so?
Is this with the TE411P? Also what do you mean by pulled the zaptel trunk source?On 2/17/06, Stagg Shelton
[EMAIL PROTECTED] wrote:This is my last update to my issue.Finally my echo problem is
resolved.On Monday morning 2/13/06 I pulled the the zaptel trunksource.That night after my customers
On inbound calls from my SIP provider I get multiple warnings as follows:
WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid host
Everything else works but these warnings are a pain and I don't know what
they are about Nothing on previos lists or Google explains...
I second that request
On 1/25/07, Kenneth Padgett [EMAIL PROTECTED] wrote:
I ran into this problem with an early batch of IP650s. Polycom's
firmware
version 2.0.3b made this issue go away.
Speaking of Polycom firmware, anyone have an up to date source for the
stuff? The site I ordered from
Hi,
Out ITSP has told us to user SIP privacy headers to hide outbound caller
ID. Does anyone know how or if this can be done in Asterisk. I tried
exten = s,3,SIPAddHeader(privacy=on)
prior to executing Dial but no luck.
___
--Bandwidth and
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Eric Bishop
*Sent:* Sunday, February 04, 2007 15:43
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] SIP privacy headers
Hi,
Out ITSP has told us to user SIP privacy headers to hide outbound
Would be greatly appreciated
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Hi all,
I am trying to implement native format (ulaw) voice prompts and music on
hold. Different documentation has different file extensions. Does Asterisk
recognise them all? So far I have .ulaw .ul .pcm . Which should I use so
Asterisk recognises them as native uLaw files
The conference bridge runs Ulaw codec by default. If you let people connect
with GSM or other codecs, Asterisk will use CPU power to convert audio
between codecs ... What about alaw channels is there any transcoding work
being done there?
___
Any kind Polycom dealers out there?
-- Forwarded message --
From: Eric Bishop [EMAIL PROTECTED]
Date: Feb 14, 2007 8:10 PM
Subject: Can anyone help me out with Polycom 2.1 firmware please?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the
I have one Asterisk box registering to another via SIP and on the registar
console I keep getting:
-- Got SIP response 603 Declined (no dialog) back from xxx.xxx.xxx.xx
Anyone know how to turn off this feature?
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Surely there must be a simpler way than patching the Asterisk code? After
all this is Asterisk-to-Asterisk registration not some third party
softswitch idiosyncrasy. Would setting up fake voicemail boxes help?
On 2/22/07, Davy Chan [EMAIL PROTECTED] wrote:
**I have one Asterisk box
I do need MWI notifcation, just not on this particulary trunk. Is there
anyway to to turn off MWI on a particular trunk or can it only be done
globally?
On 2/22/07, Olle E Johansson [EMAIL PROTECTED] wrote:
22 feb 2007 kl. 08.24 skrev Davy Chan:
**I have one Asterisk box registering to
show dialplan keeps showing contexts created by AEL. I tried deleting
/etc/asterisk/extensions.ael but kept getting these messages in the Asterisk
log:
Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open
'/etc/asterisk/extensions.ael': No such file or directory
Feb 14 21:39:53 WARNING[6074]
Hi all,
I want to implement certain actions based on SIP response codes. Is there a
similar variable such as ${DIALSTATUS} that comes back with the relevant SIP
response code for a call?
--- Thanks
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Once the call is hung up it is too late. I need to interpret the SIP
response codes prior to hangup so I can play an appropriate recorded voice
announcement.
On 4/9/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Eric Bishop wrote:
Hi all,
I want to implement certain actions based
Does anyone know if is possible to purposely set red alarm status on PRI
circuit for testing purposes (other than unplugging it). I have looked for a
console command which might allow this
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When ever we do a roll out of Asterisk in a small business environment
replacing an old key system or legacy PBX the receptionist always asks
us, How do I know if someone is on a call before transferring them?.
My typical answer is why do you need to know, just do an attended
transfer and if they
Hi all,We are looking to interconnect 2 Asterisk boxes at seperate sites via a TDM leased line, rather than IP mainly for commercial reasons. Our network provider is offering us either a 31x64kbps leased line or an E1. Am I just ignorant or are these the same thing? An E1 has 30 B channels and 1 D
are pros and cons of each service for use in conjunction with each. Could I run a PRI protocol over either one since I will cintrol both ends?
On 9/23/06, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, Sep 23, 2006 at 08:22:18AM +1000, Eric Bishop wrote:We are looking to interconnect 2 Asterisk
Hi All,When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions?
___
Does anyone have an end-to-end summary of how they have successfully set up the buddy feature including all the relevant Asterisk and Polycom config snippets. All I have been able to do so far is scrounge up bits and peices from the list and Wiki - nothing that covers the entire process... I think
Do you have anything special in your sip.conf for the Polycom phones?On 10/4/06, Scott Higginbotham [EMAIL PROTECTED]
wrote:Here is an example of what I have:in extensions.conf:exten = 2111,hint,SIP/2111
exten = 2111,1,Dial(SIP/2111,60)my Polycom's all pull config's via TFTP.Due to the nature of
Anyone have a sane rc script for FOP on CentOS/RHEL systems?
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Yes, I have some advice. Use Fedora Core 2. I have battaled for almost
a year to get fcpci and udev-based distributions working with very
limited success.
On 7/21/05, Adrià Vidal [EMAIL PROTECTED] wrote:
Someone have info about install an AVM fritz into FC3 ?
I'm getting problems with
+1000, Eric Bishop wrote:
Yes, I have some advice. Use Fedora Core 2. I have battaled for almost
a year to get fcpci and udev-based distributions working with very
limited success.
On 7/21/05, Adrià Vidal [EMAIL PROTECTED] wrote:
Someone have info about install an AVM fritz
Hi all,
chan someone who has tried BOTH chan_capi and chan_mISDN with a
passive Frtiz!Card PCI comment on one versus the other. Which had
better sound quality. I am consistently have issues with chan_capi
and echo.
Thanks
___
Asterisk-Users
Hi all,
chan someone who has tried BOTH chan_capi and chan_mISDN with a
passive Frtiz!Card PCI comment on one versus the other. Which had
better sound quality.
Thanks
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Craig,
You obviously have has experience with chan_mISDN in AU and the Fritz.
Have you tried chan_capi? I am currently using a Fritz with chan_capi
in AU and am not entirely happy with it. Is chan_mISDN any better?
On 7/27/05, Craig Guy [EMAIL PROTECTED] wrote:
The mISDN Fritz! driver supports
Hi all,
Anyone able to remotely reboot their password protected Sipura
SPA-3000 from command line. I am trying:
Sipura SPA-3000 from command line:
# wget http://admin:[EMAIL PROTECTED]/admin/reboot
The strange thing is it works fine when I go to
http://admin:[EMAIL PROTECTED]/admin/reboot with
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
Hi all,
We have several sipura 3000's working well for outbound calls, however
the issue we have is that when calls are sent to the Sipura with
Dial(SIP/${EXTEN:[EMAIL PROTECTED]) the Sipura does a SIP answer immediately
and then proceeds with the call in band therefore sending dialing
sounds
an on-hook forward to asterisk
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop
Sent: Tuesday, 31 May 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sipura 3000 dialing noise
Hi
Hi all,
We are successfuly running an Asterisk server with standard SIP hard
phones and it is working well. We are looking to deploy some soft
phones on our Linux desktops. There seems to be several floating
about. Anyone out there with some good/bad experiences with particular
Linux softphones.
Did you ever get your SP4220 going?
On 5/12/05, scott [EMAIL PROTECTED] wrote:
Hi Peoples
I would be interested to hear from anyone who has managed to get the Octtel
SP4220 and asterisk talking together.
I am using the Octtel as a gateway for a PSTN line. It passes
Did you ever get your SP4220 going?
On 5/12/05, scott [EMAIL PROTECTED] wrote:
Hi Peoples
I would be interested to hear from anyone who has managed to get the Octtel
SP4220 and asterisk talking together.
I am using the Octtel as a gateway for a PSTN line. It passes
Would you mind posting a config. Also can you comment on the quality
especialliy in relation to
1. Sound quality
2. Echo
3. Hang up detection
Very much appreciated.
Thanks...
On 6/16/05, scott kerschner [EMAIL PROTECTED] wrote:
Yes I did
-Original Message-
From: Eric
Hi all,
We have our Asterisk server running smoothly with a SIP BRI gateway
for inbound calls. However if the Internet connection goes down and a
DNS server becomes unreachable Asterisk basically does not function.
By this I mean it does not answer call coming in from the gateway
(which is on the
I'd really rather not run a DNS server if I don't have to. Surely ther
must be a way to tell Asterisk not to rely on DNS?
On 6/21/05, Guido Hecken [EMAIL PROTECTED] wrote:
We have our Asterisk server running smoothly with a SIP BRI gateway
for inbound calls. However if the Internet
I actually do not need DNS at all as I refer to all hosts via IP
addresses but Asterisk still seems to need DNS perhaps to do reverse
lookup or something like that..
On 6/22/05, Eric Bishop [EMAIL PROTECTED] wrote:
I actually do not need DNS at all as I refer to all hosts via IP
addresses
Hi All,
We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM
3.1.3 as per http://www.polycom.com/resource_center/1,,pw-492,00.html
Can someone help? We have legitimately obtained these phones but even
our official distributor can't get their hands on updated firmware. The
only
Is it possible for Asterisk to set the ptime attribute on outbound calls in SDP invite?
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