please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN .
Yahoo! Mail
Use Photomail to share photos without annoying attachments.___
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Greetings,
I''m looking for an IAX ATA w/ an FXO port. Does such a device exist in the market?
SH
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2006/3/15, Gidean Chan [EMAIL PROTECTED]:
Can Asterisk @ home receive incoming call using a external modem?
In general, modems can't be used for voip because most of them can't
do full duplex. On the other side, an SPA3000 may be cheaper than some
good external modems. Some softmodems uses chips
RAHEEL HASSAN wrote:
please tell me that what sip based softphone will beused with Asterisk
so that i can trasmit and receive video on my LAN .
Yahoo! Mail
Hi,
i have used eyebeam exten for video. However it is not
Hey group,
I just got back from the VON expo. It was insanethere were so many
companies there. The #1 thing ***EVERY*** company focused on was
convergance - getting all your communication devices to intergrate with
eachother. There were some nifty products out there that did some
Hey guys,
Got some feedback on this from Polycom. See my post Feedback from VON
expo!
- Gabe
- Original Message -
From: Noah Miller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 15, 2006
Hi All,
I have two Asterisks, one on the LAN that handles the internal calls
with a PSTN interface and one on the DMZ with a public interface. I
would like to route SIP calls from the internal users to the Internet
over IAX2 to the DMZ and onwards.
All users have soft phones so they would enter
Could be the same problem I had with my Aastra - progressinband=no
worked for me.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 15 March 2006 18:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi all,
I have a couple of asterisk servers running. When one asterisk server dials another asterisk server
over IAX, i want to match that call in both of the cdr's. How do i make both asterisk servers use
the same uniqueid for that call, if this is possible. Or is this a dumb question since
I'm using asteriskathome 2.5. I'm using 2 spa3000 for dialing-out. I
configured a trunk for each one with maximun channels=1 and an
outbound route that includes both trunks. When a second outgoing call
is placed, Asterisk tries to place it in the same that is already in
use resulting in a busy
hi
i think that the only way to refresh data on page without reloading is
to use ajax
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Hi All,
Thanks for your replies.
I need many contexts because I have around 1000 DID's each with 5-10
Extensions.
These DID numbers are changed or added very frequently and whenever there is a
change I have to change Extensions.conf manually. So please tell me how can I
do this dynamically
2006/3/16, Bart J. Smit [EMAIL PROTECTED]:
Can Asterisk do this? I am relatively new to Asterisk. I guess I'm after
something like an email smarthost feature for SIP.
Yes, Asterisk can do protocol conversion as well as codec conversion.
Just configure phones and asterisk to connect correctly
RAHEEL HASSAN wrote:
please tell me that what sip based softphone will beused with Asterisk
so that i can trasmit and receive video on my LAN .
I'm using Vizufon CIP-5500.
--
Best regards,
Bartosz Piec
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Hi!
Does anyone know how to configure flash operator panel to be able to
transfer/hang up calls? I'm trying to set it up, but for me, it only works
as a status monitor, because if (for example) I try to drag a phone icon
to transfer a call, it ask me to insert the security_code, then I digit the
Hi all
I have badly NATed Clients proble with one way Voice
After reading some documents people ask me to use STUN Server
But still i have some problem with one way Voice
I have setup like below
iam trying with 2 extensions
1 extention in the same LAN where the * installed
2 extension in
Look also at AudioFrames setting on your phone.
I read that it needs to match 20ms packet size of Asterisk packets and it
depends from codec you use.
Mimmus
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From http://nerdvittles.com/:
Max Channels Bug Remains. A bug has been reported because of a deprecated
command that makes [EMAIL PROTECTED]'s calculation of maximum channels invalid.
To fix it, goto AMP-Maintenance-Config
Edit-extensions.conf-macro-dialout-trunk and comment out line s,7 so that
On 03/16/06 04:45 bails said the following:
Hi whatever I set the span line to in zaptel.conf
ie span=1,0,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4
span=1,2,0,ccs,hdb3,crc4
why are all your spans numbered 1 ? surely they should be numbered 1,2,3,... ?
[i'm assuming that
Thanks Alejandro,
I'm sure the codecs are fine, as I can make calls inbound to the LAN
Asterisk.
Can you tell me which configuration changes I need to make on each
Asterisk to route these calls?
Bart...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
That's in the [general] section of sip.conf, yes ?
How would that affect the 7.4 phones ? Wouldn't want to annoy them ;)
Julian.
Lee Archer wrote:
Could be the same problem I had with my Aastra - progressinband=no
worked for me.
Lee
-Original Message-
From: [EMAIL PROTECTED]
Why not just set it for the affected extensions in sip.conf? I did it
globally and my GXP's didn't mind.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 16 March 2006 09:41
To: Asterisk Users Mailing List - Non-Commercial
Hi all
The first problem I noticed, is that I get very choppy sound, when there are
users wich connect to meetme via GSM. Is there a way to force meetme to user
aLaw even if the user is connected via gsm?
The second problem is that I already had two server freezes just after a gsm
user
ram wrote:
Hi all
I have badly NATed Clients proble with one way Voice
After reading some documents people ask me to use STUN Server
But still i have some problem with one way Voice
use stun on dinamic ip :)
I have setup like below
iam trying with 2 extensions
1 extention in the same
Kevin Bockman wrote:
Barry Flanagan wrote:
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when
we do the transfer and dial the new number, it times out after 3 rings
and then the callee is put back to the original agent.
Where can I adjust the timeout which applies to
Dinesh Nair wrote:
On 03/16/06 04:45 bails said the following:
Hi whatever I set the span line to in zaptel.conf
ie span=1,0,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4
span=1,2,0,ccs,hdb3,crc4
why are all your spans numbered 1 ? surely they should be numbered
1,2,3,... ?
Hi,
Does anyone know what would be acceptable RTT. Is 200ms OK?
Regards,
Stojan Sljivic
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Martin Joseph
Sent: Wednesday, March 15, 2006 18:48
To: Asterisk Users Mailing List - Non-Commercial
Hi everyone,
I have an issue which is kind
of a catch 22 situation. I had outgoing calls to my new PSTN provider working
perfectly. Then I started focussing on incoming calls. It seems that I can
solve an error which gets my incoming calls working but that in turns means my
outgoing
Hi all, I've attached a copy of the debug from the Trend, if anyone
cares to look.
I'll probablt get more of a response fromt the list than the automated
response from digium :(
Thanks
Bails
---BeginMessage---
Dinesh Nair wrote:
On 03/16/06 04:45 bails said the following:
Hi whatever
On Thu, March 16, 2006 12:08, Stojan Sljivic - GDS said:
Hi,
Does anyone know what would be acceptable RTT. Is 200ms OK?
Regards,
Stojan Sljivic
When any of my VPN tunnels get over 100ms I start to get worried! Avg
speeds on the tunnels are below 45 ms...
I guess it depends on the level
Hi
thanks for the reply
ya the default is NAT=YES only
if i keep reinvite=no, the my server b/w consuming lot
since i have bottleneck of server bandwidth
ram
On 3/16/06, Andrei Sotirov [EMAIL PROTECTED] wrote:
ram wrote: Hi all I have badly NATed Clients proble with one way Voice
After
browsing the web i don't find any opensource (and free of charge )
software for the web statistic about queues...
i've tries queue_stats made from asteriskguru, it is a good tool, and
it is free of charge, but it's not open-source :-(
i'm considering to develop myself a web application, before
On 13:11, Thu 16 Mar 06, nik600 wrote:
browsing the web i don't find any opensource (and free of charge )
software for the web statistic about queues...
i've tries queue_stats made from asteriskguru, it is a good tool, and
it is free of charge, but it's not open-source :-(
i'm considering
how can I send text directly to a specific device, something like:
exten = 103,1,SendTextToDev(SIP/7, hello) ??
I don't think you can send to a particular device, but you can send it
to the device calling if it support it.
See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SendText
snip
In the US I think this illegal? Aren't you supposed
to have some sort
of notification or beeping to indicate a recorded
call to the other
party?
/snip
yes. and that is why a lot of times you will hear
calls may be monitored for quality control purposes
/snip
Nothing complicated really Just a carrier class
solution, with advanced custom routing, incoming and
outgoing number blocking (at user/company and global
level) and whitelisting, findme/followme, user
specific pic codes and rate centres based on number
dialled, blocking of specific
Michiel van Baak wrote:
On 13:11, Thu 16 Mar 06, nik600 wrote:
browsing the web i don't find any opensource (and free of charge )
software for the web statistic about queues...
i've tries queue_stats made from asteriskguru, it is a good tool, and
it is free of charge, but it's not
snip
It's making you miserable, and YOU are making a lot
of us miserable with
your incessant and childish whining.
/snip
1)I will go out here and defend doug. It has been a
lng time since we have heard him whine. Back in
the day his emails werent th best and we asked him to
change his
OK now your question is starting to make
sense.
What happens if bosses line is buzy do calls 'rollover' to
line 2 and 3?
What criteria will define access to the other
lines?
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faisal
InamSent: Thursday,
Try it with qoutes "mono
script.exe"
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nello
GaudinoSent: Thursday, March 16, 2006 2:08 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Problem
with System() command.
Hi, I have an
Hi,
Need help from you guys. I had my Asterisk Set-up using PRI card TE110p, and
everything working ok. However, I had bad experience with Asterisk answering
call
The problem was, when outsider calling into Asterisk...
Asterisk answered call...
CLI Accepting Overlap call from
Without your configs it ill be hard to see what is going on.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
chan (Alpha Trilogies Networks)
Sent: Thursday, March 16, 2006 8:15 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Also be aware Asterisk is probably runing
in its own, non-root account. It needs execute access to the program,
and you need to specify full path. At least thats what worked for me J - dialing 500 on my box
does System(/sbin/reboot) !
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Look
Eyebeamof Xterm.
-Mensaje original-De: RAHEEL HASSAN
[mailto:[EMAIL PROTECTED]Enviado el: Thursday, March 16, 2006
4:05 AMPara: asterisk-users@lists.digium.comAsunto:
[Asterisk-Users] How to transmit Videoplease tell me that
what sip based softphone will beused with
Brian McEntire wrote:
I looked on the voip-info wiki and found sparse and conflicting
information on how to do this with Asterisk...
My incoming lines are all on Zaptel. Is there a simple why to
implement a '*363 (do not disturb) toggle via the dialplan?
I have an extension for
Hi all,
I can't figure out why my TDM400P (with one FXO plugin) won't answer any
calls. There are no messages in the Asterisk console when a call is
placed to the FXO line from the PSTN. Any suggestions would be most
appreciated.
The wctdm and zaptel modules are loaded:
[EMAIL PROTECTED]
Jerry Geis wrote:
however all the phone shows is the initial config from my office.
Its either not picking it up, rejecting it or something???
Doing a diff between the txt files from my office and the second
location shows only the
proxy and UID and password fields as being different.
Yes, after putting in the code drag it again. Also what does your
op_panel.cfg look like?
FOP has it's own mailing list, you should try it there.
On 3/16/06, Giuseppe [EMAIL PROTECTED] wrote:
Hi!
Does anyone know how to configure flash operator panel to be able to
transfer/hang up calls? I'm
Here is the patch file which I use (I manually removed some other
parts of the patch, so I hope it is okay!) - It should be sufficient
to get you going.
cd into the zaptel-1.0.9.2 source directory, and
patch -p1 zap-patch.txt
Cheers,
Steve
On 3/15/06, Colin Anderson [EMAIL PROTECTED] wrote:
Is
Matt Florell ha scritto:
I've noticed this as well from pre 1.0 versions through to 1.2.5
across 12 separate Asterisk servers. The severity seems to be random
mostly. I still haven't figured out what is causing it.
MATT---
Your file system is journaled ? this is another common thing that
Grretings to all,
I am having a problem with a customer's queue setup that I don't really
understand.
Background: Customer has 5+ queues and is using dynamic login to the queues
based on SIP/XXX for example. There is a litle script that runs that allows
agents to log into particular queues via
I am not sure what I did but blind transfers do not work. The Polycom does not allow me to dial the extension of the person I want to transfer to after I hit:transfer - blindthanks "Mojo with Horan Company, LLC" [EMAIL PROTECTED] wrote: When you hit the polycom's transfer button, a
oops. attachments are blocked :) I'll email it directly to anyone who
provides an email address.
Regards,
Steve
On 3/16/06, Steve Davies [EMAIL PROTECTED] wrote:
Here is the patch file which I use (I manually removed some other
parts of the patch, so I hope it is okay!) - It should be
Yep I use ext3, have you run test with any other file system?
MATT---
Your file system is journaled ? this is another common thing that came
to my mind (ext3)
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Gabe.
Who was the call-center program from?
Thanks,
Jim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gabriel
Afana
Sent: Thursday, March 16, 2006 2:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Feedback
Can anyone help point me in the right direction please?
I'm based in the UK and I want to start using a Premium Rate number with
Asterisk - I think the equivalent in the US would be a 900 number.
Effectively the caller pays much more to call such a number than a
normal national or local call.
Hi
You could use windows messenger, kapanga or sipps (deppends on what you want)
Jose Manuel Cortes David
X Semestre Ingenieria Electronica
PONTIFICIA UNIVERSIDAD JAVERIANA
De: [EMAIL PROTECTED] en nombre de RAHEEL HASSAN
Enviado el: Jue 16/03/2006 3:05
2006/2/22, Alejandro Vargas [EMAIL PROTECTED]:
I want to do inbound routing of calls comming from sip trunks. Is
there a way to force the DID that comes from a trunk that does not
have DID support? (something like using the outgoing caller-id for the
trunk?)
I answers myself. To identify from
Hi John,
I had the same error when configuring our TE110P. The only way I was able
to fix this error was to physically move the card to a different PCI slot.
Please note the server I used was the IBM x206 server.
Hope this is of some use.
Cheers,
Phil.
Matt Florell ha scritto:
Yep I use ext3, have you run test with any other file system?
MATT---
No, I will do when I have time (and a server to test on)
Your file system is journaled ? this is another common thing that came
to my mind (ext3)
Hi all,
I'm trying to get a junghanns QuadBRI to coexist in the same machine as a
Digium TDM400P card (so I can run the ISDN lines in and bridge with analog
phones plugged into the TDM).
I'm having a problem loading the modules. If I follow the BRIstuff
(0.3.0-pre-1l) install method it's
Anyone using ODBC voicemail storage in mySQL?
For what volume of voicemail?
Any performance issues?
Seems like a key piece of the failover clustering puzzle
(vs. syncing file systems).
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Great Email. I'm going to respond to some of the points.
Q: What are the plans for HA?
A: With a configuration using DNS-SRV and DUNDi, you can create a
pretty resiliant setup now.
That's BS. Last time I checked, Asterisk's support of SRV was to only grab the
first SRV entry. Period. If
Can someone explain me attended transfer with Grandstream GXP-2000?
Hitting TRNF button, I get:
Dial number (BLIND) or
Select line (ATTENDED)
What's the exact meaning of 'Select line'?
Thanks
Mimmus
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Q: What are the plans for HA?
That's BS. Last time I checked, Asterisk's support of SRV was
to only grab the first SRV entry. Period. If it doesn't try
any more SRV hosts after the first fails, just exactly how is
that redundant?
This is for the phones to fail over NOT Asterisk,
If you have Line 1 on hold, and you on a call on Line 2, then hitting TRNF
and hitting Line 1 will transfer Line 2 to Line 1. Same concept as
Conference.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mimmus
Sent: Thursday, March 16, 2006 7:30 AM
- Original Message -
From: rnacharya
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 15, 2006 5:56 AM
Subject: help required configuring card
Hii,
I've a Digium TE205P card and I'm
running [EMAIL PROTECTED] in a box.I want to
configure this card in that box.But as
-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 16, 2006 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on
*HAandPolycomphone!!
Q: What are the plans
Faris Raouf wrote:
Can anyone help point me in the right direction please?
I'm based in the UK and I want to start using a Premium Rate number with
Asterisk - I think the equivalent in the US would be a 900 number.
Effectively the caller pays much more to call such a number than a
normal
I need to test QoS on an IAX link between a server in Colorado and a
server in Europe. I know I could install a Milliwatt extension on the
European server and just listen, but is there a more scientific method
to collect QoS metrics?
Thanks
P.S. I'm getting a lot of Page Not Found on
If we get an incoming call I can edit the callerID provided to add the
leading '90' and set the name so that sales calls can be identified
according to the number called.
If however the callerID is unavailable then setting the callerID name or
number fails (it shows as unavailable on the phone).
If you are in Southern California and would like to attend the Asterisk Users
Group Meeting, it is tonight from 6-9pm at the Heritage Park Library.
Irvine Heritage Park Library(949) 936-404014361 Yale AveIrvine,
CA 92604
Tonight we will be having a demo of SIPX, a review of the SNOM 320
I see this every once in a while. I will have channels that just don't
seem to hang up.
When I do show channel...
Elapsed Time: 24h38m15s
Any suggestions?
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To
Maybe this will shed some light about what I'm trying to do:
This is some output from dmesg after this load order:
modprobe zaptel
insmod wcfxs
insmod qozap
Zapata Telephony Interface Registered on major 196
Freshmaker version: 73
Freshmaker passed register test
Module 0: Installed -- AUTO
I wish to create a voip phone system used by different people accross
the internet.
I want certain people dotted around the country to be able to connect
via voip to our main office.
At first this will be using software phones but could extend to hardware
based phones if it works well.
I would
Anyone ever seen MeetMe cause * to crash? Specifically, it happens
consistantly if someone begins to enter a conference and then decides to
hangup while Allison is introducing them - like playing back
conf-onlyperson. This has been seen with the MeetMe participant connecting
via IAX and SIP (not
Sorry forgot to mention I am running 1.2.0RC1 (dont ask :) )
Here is the macro used to set the callerid.
[macro-uksales]
; UK SALES
; ARG1 = Caller ID Name to display on phone
exten = s,1,Set(CALLERID(name)=${ARG1})
exten = s,2,Dial(SIP/6030IAX2/6030SIP/6514,15,t)
exten = s,3,Voicemail(u6030)
Hello.
Is there any way to forward incoming call to voicemail in
one ring if the person on the extension is busy.
Regards
---
Navneet Shah
Systems Administrator
YL Consulting, Inc.
210-340-0098
We are looking for an Asterisk programmer to perform maintenance and
upgrading programming to a Asterisk telephony project in Indiana. You
must have experience in Asterisk dialplans, digium T1 and analog
hardware cards, complete knowledge of VoIP, MySQL/Asterisk
integration, VoIP protocols
Philip Edelbrock wrote:
At some point (in a few months, probably) we'll turn off the Toshiba
and put viop phones on everyone's desk (including some people's at a
remote office and homes).
It should also cut our phone bill down to a 1/10th of what it is now!
Interesting... so, you consider
Hi
Does anyone know the clear advantages over using asterisk rather than [EMAIL
PROTECTED]
Is the home version limited in anyway etc?
Many thanks in Advance
Scott
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Same Asterisk but AAH is easier to setup and get running. There are
no limitations. Test it out.
On 3/16/06, scott [EMAIL PROTECTED] wrote:
Hi
Does anyone know the clear advantages over using asterisk rather than [EMAIL
PROTECTED]
Is the home version limited in anyway etc?
Many thanks
On Mar 16, 2006, at 3:24 AM, Aisling wrote:
x-tad-smallerHi everyone,/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerI have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems
Hi All,
I will again tell what I am trying to do.
I have around 1000 DID's and I have to setup context for each of it's
extension
and I want to do that dynamically and I do not want to change extensions.conf
all the time manually whenever I want to add new context instead I will do it
in
scott wrote:
Hi
Does anyone know the clear advantages over using asterisk rather than [EMAIL
PROTECTED]
Is the home version limited in anyway etc?
Using Asterisk instead of AAH gives you a better understanding of how
things work and what to do when problems arise.
Doug
--
Ben
Sure, just make your voicemail wait 5 seconds before
answering the call.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Navneet
ShahSent: Thursday, March 16, 2006 10:45 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Dialplan :
Forwarding call to voicemail
Okay
think I finally figured this out
it's the modules.conf post-install lines that run ztcfg
You're not supposed to run ztcfg more than once with the multiple zaptel
cards in there I kept running it manually (ztcfg -) not realizing
that after modprobe wcfxs the ztcfg was being run.
Hi,
I am trying to upgrade an Asterisk 1.0 with chan_capi 0.3.4 to a more
recent version, but I cannot find any working combination of Asterisk an
chan_capi any more:
On junghanns.net there is a chan_capi 0.3.6, but this won't compile
against any recent Asterisk (missing channel_pvt.h).
The
Thanks for the reference to http://winscp.sf.net/ .
I always thought that it was command line, so I have always either used wget or
ftp as well.
I have another Linux box that I use for monitoring (mrtg and nagios) and a
helpdesk system (orts) that I loaded samba on to do quick
file edits, but
If you are using Comedian mail, you can to notification at the mailbox.
ref: voicemail.conf
5600 = ,Steven ,[EMAIL PROTECTED],[EMAIL
PROTECTED],attach=no|saycid=no|envelope=yes|delete=no|nextaftercmd=yes
The [EMAIL PROTECTED] will send an email to my phone to let me know there is a
www.aheeva.com
- Original Message -
From: Jim Houser [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, March 16, 2006 6:50 AM
Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on * HA
At 03:04 AM 03/16/2006, you wrote:
Does anyone know the clear advantages over using asterisk rather
than [EMAIL PROTECTED]
Is the home version limited in anyway etc?
If AAH works, it's pretty cool. Personally I needed to do something
it couldn't do so I gave it up after a couple of weeks. I
I need many contexts because I have around 1000 DID's each with 5-10
Extensions.
These DID numbers are changed or added very frequently and whenever there
is a change I have to change Extensions.conf manually. So please tell me
how can I do this dynamically without changing Extensions.conf
Q: What are the plans for HA?
That's BS. Last time I checked, Asterisk's support of SRV was
to only grab the first SRV entry. Period. If it doesn't try
any more SRV hosts after the first fails, just exactly how is
that redundant?
This is for the phones to fail over NOT Asterisk, remember
On Mon, 13 Mar 2006 14:19:05 +0200
Benchev [EMAIL PROTECTED] wrote:
Hmm, both of you recommend a solution with the dial cmd in an
agi-script, i would prefer a direct solution but i guess there is none.
There is - H - Allow the calling party to hang up by hitting the '*' DTMF
digit.
I
[EMAIL PROTECTED] wrote:
Hi,
I am trying to upgrade an Asterisk 1.0 with chan_capi 0.3.4 to a more
recent version, but I cannot find any working combination of Asterisk an
chan_capi any more:
On junghanns.net there is a chan_capi 0.3.6, but this won't compile
against any recent Asterisk
On Thu, 16 Mar 2006, Peer Oliver Schmidt wrote:
[EMAIL PROTECTED] wrote:
Hi,
I am trying to upgrade an Asterisk 1.0 with chan_capi 0.3.4 to a more
recent version, but I cannot find any working combination of Asterisk an
chan_capi any more:
On junghanns.net there is a chan_capi
I have a strange problem in that I have put a budgetone out on the internet
that connects to my * server that's behind a firewall.
They can call me I can call them and it works fine. However, I have setup a
link to sipdiscount on my * server. If the budgetone user calls via my * box
to
Hi -
I am not sure what I did but blind transfers do not work. The Polycom does
not allow me to dial the extension of the person I want to transfer to after
I hit:
transfer - blind
I would strongly suggest getting the latest firmware, and using the sample
configuration files with
I am using the latest firmware and bootrom and this is a problem with all 12 polycom 501s that we have in the office. If I want to transfer to 1005 for example while on the phone with the original caller, I press transfer - blind - type "1", "0" then the phone clears the display and the
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