Hi All
Has anyone used Cubix / Firefly successfully with Asterisk? When
someone calls a Cubix softphone, Cubix never seems to answer the call
correctly. The other person just hears ringing even though it has been
answered. I am using IAX as the SIP support doesn't seem to 100%
either.
On 2006-10-10 20:25:44 -0700, Nic Bellamy [EMAIL PROTECTED] said:
On 2006-10-08 21:28:08 -0700, Nic Bellamy [EMAIL PROTECTED] said:
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
snipI wonder if there
On 2006-10-10 18:12:23 -0700, hugolivude [EMAIL PROTECTED] said:
An Internet browser uses port 80. I might have two or more behind a
NAT both using port 80. Isn't that the same thing?
Remember that the browser INITIATES all activity on the port 80
transfers. There is no data coming in out
Dear
I am using
a2billing accounting software, how can I charge on the destination target not
at the caller side
Ex: if user
A have 10$ and user B have 10$ ,and the onnet call charge cost 1$
When user A
call user B for 1 minute ,user A amount remains 10$ and user B amount be 9$
Friends in the Asterisk community,
I've been talking for years about the new version of the SIP channel.
I've been trying to get funding
and get going. Well, the funding part remains to be handled, but I
have other news - if you kan keep
it to yourself.
...I've began coding. Finally.
With
Hi list!
I recently upgraded to FreePBX 2.1.3 although I am not sure if this has
something to do with it.
I do a nightly restart of Asterisk, just in case. This has been working
fine months but since a few days asterisk seems to die and I am not able
to restart it again, I keep getting a
On Tue, Oct 10, 2006 at 11:43:09PM +0100, David Bath wrote:
Hi All,
I've tried to find the solution to this, but sadly met with failure.
I've got an asterisk box with two X100P's in, and both cards have the
same strange problem. After 2min 40seconds (always: within +/- 1sec)
they drop
Hi all!!,
I haven't the 'r' options in the dial command. I also try to turn off
busydetect and callprocess obtaining the same result..
If I turn off polarityswitch, I get hangup instead busy...
The peer isn't busy because I'm trying with my movil phone, and whit
known
Folks,
I am absolutely new to asterisk for the Voice Over IP. I have set up my
own server using asterisk, successfully connected and be in position to
test the voice over IP by connecting to the digium server and testing
the echo system working absolutely fine. My therefore comes, how to work
Hi,
Installed 1.4.0 libpri and 1.4.0
zaptel and everything went smoothly. I configured asterisk 1.4.0 with no
problems (./configure), but when I compile it (make), it fails with this
error:
[LD] res_snmp.o snmp/agent.o -
res_snmp.so/usr/bin/ld: /usr/local/lib/libz.a(gzio.o): relocation
On Wed, Oct 11, 2006 at 09:14:55AM +0200, Remco Barendse wrote:
Hi list!
I recently upgraded to FreePBX 2.1.3 although I am not sure if this has
something to do with it.
I do a nightly restart of Asterisk, just in case.
Why?
This has been working
fine months but since a few days
On Wed, 11 Oct 2006, Tzafrir Cohen wrote:
On Wed, Oct 11, 2006 at 09:14:55AM +0200, Remco Barendse wrote:
Hi list!
I recently upgraded to FreePBX 2.1.3 although I am not sure if this has
something to do with it.
I do a nightly restart of Asterisk, just in case.
Why?
Sometimes
Hi Aaron!Could you please provid me your patch for 1.2? I didn't get you, it was a problem for you to get the messages into mp3 format?Did you have any problem until now with this patch on *1.2 ? My box is
1.2.5 and still very stable until now:)Hope you can help me, i can't figure out why no one
Hi,Is is possible to implement this:Hicom150 --- BRI (QSIG) AsteriskI've been reading Siemens documentation and they say:Digital nailed connectionsCorporate communication networks can be implemented over digital S0 or
S2M nailed connections between several Hicom systems using the CorNet
Hi * guys,
I had a perfectly working * (1.2.0 version). I updated it to 1.2.12 and now
VoiceMail app doesn't find entries in voicemail.conf any more. I recompiled
only * 1.2.0 and installed it again and now Voicemail is up again, with no
configuration's change!
Anybody knows anything about this?
I experienced a similar problem, but with AT-RG 623TX (ISDN BRA
gateway). I can only tell you that there is no Asterisk problem. You
should try to debug hardware / driver problems.
Question: is(are) the user-agent(s) still authenticated with Asterisk
after the call is dropped? You should also set
Have a look at the book: Asterisk: The future of Telephony. It will teach
you almost everything that you need to know. Also you have the wiki
(http://voip-info.org) and remember google is your friend.
- Original Message -
From: Ahmed Ndaula [EMAIL PROTECTED]
To: Asterisk Users
Hi,
situation is the following:
There's an inbound call, that rings on SIP/tel21 (ATA is PAP2). At the
time, bobody there, but a lazy people sits by SIP/tel22 (about 5m
distance) and he want to takeover the call. How can I do this whit
asterisk?
Ok. I can do with call parking, but with call
I have been seeing this problem for a long time and it occurs in 1.4.0b2
(as well as 1.2.0-1.2.12.1).
If the internet connection is lost and I have SIP services that require
me to register, any SIP devices attached to the system stop working.
I have an IAX phone connected to one of my
If you are an ITSP provider, you could do with SPC tools (provided by
Linksys to ITSPs)
Regards
Curt Shaffer escribió:
Can anyone point me to a good source for provisioning WRT54GP2 from a
central server?
Thanks
Curt
On Tue, Oct 10, 2006 at 05:03:30PM -0400, hugolivude wrote:
I understand how sitting behind a NAT could cause problems for a SIP
UA. The SIP UA would create SIP mesages using IP addresses from
inside the network (i.e. 192.#.#.# or 10.#.#.#) and these IP addresses
are of course
On Tue, Oct 10, 2006 at 05:03:30PM -0400, hugolivude wrote:
Similarly, why do we need a timeout on a SIP registration? Does this
work the same way as a heartbeat enabling disconnected UA to be
unregistered?
Yes, that's the purpose: so that if you unplug a SIP phone without giving it
On Wed, Oct 11, 2006 at 04:06:06AM -0400, Ahmed Ndaula wrote:
I am absolutely new to asterisk for the Voice Over IP. I have set up my
own server using asterisk, successfully connected and be in position to
test the voice over IP by connecting to the digium server and testing
the echo system
Hello everybody!
[Introduction]
This is a quite long message, but I think the problem is interesting.
[The problem]
Does anyone know how can I tell Asterisk that a certain user
has a certain telephone number (or address)? For example, I have some
registered users, but nor the
Hello everybody!
I have an Asterisk 1.2.12.1 server with SIP as the VoIP
protocol.
What I want to do: I want to talk to the outside
world via MGCP.
I suppose I must set an MGCP peer to route outgoing calls.
So, I must set the endpoint syntax of the Asterisk server (Asterisk will
Folks,
I hope this is not a FAQ or some other kind of dumb question. I am
currently running 1.2.10-BRIstuffed-0.3.0-PRE-1s using a straight-forward
configuration mostly only for ISDN. However, I am also accepting
anonymous SIP connections for external people calling me. This always
worked until
Hi!
Has someone ever used the sendfax option of new chan-capi to send fax? I
need some help regarding the sff format:
How can I generate sff format? I found sfftobmp, not nothing the other
way round.
Is there a nice way to get the sff out of an Windows application (like
virtual printers
Lets say that I could modify some
stuff in register_verify function (which returns -2 for my request), but I
would also need to modify the sip_request struct and this implies things I dont
know very well.
As I can see, struct sip_peer doesnt
contain any information about user number
Hello asterisk-users,
I have problem with E1 line between Asterisk computer and our PBX
Matra:
asta*CLI pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200
Hi,
I redifined the transfer key in Asterisk 1.2.11 svn from the default # key
to ** and when I do a show features in CLI I get:
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# **
How can I generate sff format? I found sfftobmp, not nothing the
other way round.
You can use ghostscript:
gs -dNOPAUSE -dBATCH -sDEVICE=cfax -sOutputFile=output.sff input.ps
Is there a nice way to get the sff out of an Windows application
(like virtual printers for hylafax) or at least
I quits on my as well, when I try to make a second call.
There is a bug report on it:
http://bugs.digium.com/view.php?id=7972
--
#Joseph
On Wed, 2006-10-11 at 09:14 +0200, Remco Barendse wrote:
Hi list!
I recently upgraded to FreePBX 2.1.3 although I am not sure if this has
something to
On 10/11/06 21:15 Joseph said the following:
I quits on my as well, when I try to make a second call.
There is a bug report on it:
http://bugs.digium.com/view.php?id=7972
this seems like a configuration error within FreePBX and isnt really a bug
in asterisk.
--
Regards,
I have NFAS setup on several quad port T1 cards (Sangoma).
It mostly works well with the exception that calls coming in on channels
48,72, and 96 have no audio. I tried removing these channels from
zapata.conf with hopes that the channels would not come up or be used.
Now I get Ring
Hi Jens!
Thanks for the script.
Do you generate and notifications (succeeded, failed) or retransmit in
case of failed sending? Or does that CAPI internally?
regards
klaus
Jens Vagelpohl wrote:
How can I generate sff format? I found sfftobmp, not nothing the other
way round.
You can use
Hello,
When I restart asterisk the cisco 7960/7940 phones (sip fw 7.5) fail to
re-register themselves with asterisk, even though I put
timer_register_expires: 60 in SIPDefault.cnf
Is there a way to have these phones register themselves every 60
seconds?
Alternatively, can asterisk be made
I've uploaded a patch to my host, it only does the volgain in int format
(we use +7 which seems to work well). We've had no problems with it
since we set it up back in February, and everyone seems to love it since
nobody's blowing out their speakers anymore lol.
The patch we use actually does a
Steve Totaro wrote:
I have NFAS setup on several quad port T1 cards (Sangoma).
It mostly works well with the exception that calls coming in on channels
48,72, and 96 have no audio. I tried removing these channels from
zapata.conf with hopes that the channels would not come up or be used.
Hello,
Has somebody installed this configuration: Asterisk + E1 with MFC/R2
(Telefónica Argentina) in Argentina before? I need to know if it´s
possible with MFC/R2 argentine variation.
Thanks in advance.
R.R. Libera
___
--Bandwidth and Colocation
Most want the 2.0.1 firmware for a few reasons:A) They have the latest; although, this is a poor reason, it's still a reason people download and use the latest firmware - remember here always, "If it's not broken, DON'T fix it!"B) They are hoping to fix a previous problem they've had in the past
That's a bug with the 7.5 firmware. I would suggest upgrading to the
8.4 version, we've been running it for a few weeks in a test environment
and everyone's been pretty satisfied with the new firmware (read:
nobody's complained). If the server goes out, they re-register after
the timeout without
I'm not saying Microsoft is the standard (they usually aren't by FAR), but how Microsoft handles presence and interoperates with presence on various IP phones is what Polycom calls a "standard" (guess I should have quoted that word originally).I believe there is some RFC for presence out there
Hi, everybody:
I have just set up a system with a regional VOIP provider.
I have two IAX channels to this provider.
Incoming calls ring a configured SIP extension immediately, but outgoing
calls are delayed for about 8 to 10 seconds before the remote PSTN end
starts ringing:
-- Called
Dean,Try obtaining the latest bootrom again, should be 3.2.2, we've seen this happen before for various odd reasons and Polycom's recommended fix is get the "non-engineering version" of the bootrom (don't ask please, just do it).So download the bootrom again and attempt it once more, while you're
Limit was increased in firmware 2.0.1.NOTE: a new Polycom Administrator's guide is now also available covering the 2.0.1 features. Re-obtain this manual if you haven't from your reseller or from Polycom direct if you're certified. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV:
Kristian Kielhofner wrote:
Steve Totaro wrote:
I have NFAS setup on several quad port T1 cards (Sangoma).
It mostly works well with the exception that calls coming in on
channels 48,72, and 96 have no audio. I tried removing these
channels from zapata.conf with hopes that the channels would
externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio
vm-audio uses 'sox -e' to determine how much to scale by without clipping
and then
Then 'sox -v' to scale the sound file.
This happens after the email message is sent, but by changing the order of a
few lines in the app_voicemail.c program you can
Kristian Kielhofner wrote:
Steve Totaro wrote:
I have NFAS setup on several quad port T1 cards (Sangoma).
It mostly works well with the exception that calls coming in on
channels 48,72, and 96 have no audio. I tried removing these
channels from zapata.conf with hopes that the channels would
We
must have had the magic version of 1.6.x then, because we increased our buddy
watch limit from 8 to 48 in that version.
-Original Message-From: Jessee J Holmes
[mailto:[EMAIL PROTECTED]Sent: Wednesday, October 11, 2006 8:18
AMTo: Asterisk Users Mailing List -
I lost my internet connection today for a short time.
During that time 1.2.12.1 stopped talking to my phones.
Asterisk was still working as I got 2 voicemails. I have TDM analog
cards for incoming calls.
Anyway my cisco phones had X's (lost registration) and my uniden phones
said Registration
Sangoma has excellent technical support, and usually pretty quick to respond
IF you are sure it isn't a configuration issue, your best resource is
Sangoma
Please report back when it is resolved.
John Novack
Steve Totaro wrote:
I have NFAS setup on several quad port T1 cards (Sangoma).
It
Hi Jesse,
4 x ip500s
Ive held off upgrading the bootrom
past 2.62 as I understand this is a one way trip to 3.01 and above.
As Im a second hand hardware user I
dont have access to Polycoms direct firmware and have been
upgrading from freedomphone.net
Cheers,
Dean
Hello to all, I have a question. I am installing te110p, when I give
ztcfg him - v leaves the following error to meZT_CHANCONFIG failed on channel 25: No such device or address (6)- That means east error?- It is a physical damage
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Klaus,
The incoming fax script will generate an email with the fax attached,
and there is another script, sendfax_status.py, which is run as a
DeadAGI after the outgoing fax has been sent, it retrieves status
information and sends it to a
http://bugs.digium.com/view.php?id=6682
Thanks I patch my installation with the patch on the above URL. It works fine
now. Thanks Moises.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote:
That's a bug with the 7.5 firmware. I would suggest upgrading to the
8.4 version, we've been running it for a few weeks in a test environment
and everyone's been pretty satisfied with the new firmware (read:
nobody's complained).
Jessee,
The reason for me upgrading to 2.01 is we wanted to add some 430s
to our system which from what I understand have a problem with 1.67, at
this point we will just go with more 501s instead.
What is the procedure to go back to 1.67?
Will you be adding 1.67 to your FTP site?
Has somebody installed this configuration: Asterisk + E1 with MFC/R2
(Telefónica Argentina) in Argentina before? I need to know if it´s
possible with MFC/R2 argentine variation.
I have not tested in Argentina, but support is included in the code,
so I suppose it should work.
Regards
--
Su
Jerry Geis wrote:
I lost my internet connection today for a short time.
During that time 1.2.12.1 stopped talking to my phones.
Asterisk was still working as I got 2 voicemails. I have TDM analog
cards for incoming calls.
Anyway my cisco phones had X's (lost registration) and my uniden
Would you be able to tell me which lines must be reordered in app_voicemail.cOn 10/11/06, Cullin J. Wible [EMAIL PROTECTED]
wrote:externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio
vm-audio uses 'sox -e' to determine how much to scale by without clippingand thenThen 'sox -v' to scale the sound
Hi Tzafrir,
Many thanks for reply.
Busydetect is also disabled. There's no chance of an actual busy
signal, as it happens exactly 2m 40 seconds (give or take 1s) into an
active call with both parties connected and talking away.
Zapata.conf copied below:
[channels]
signalling=fxs_ks
On Wed, 2006-10-11 at 11:19 -0400, Jerry Geis wrote:
I lost my internet connection today for a short time.
During that time 1.2.12.1 stopped talking to my phones.
Asterisk was still working as I got 2 voicemails. I have TDM analog
cards for incoming calls.
Anyway my cisco phones had X's
Hi DiegoF,
I had a similar problem, it was a zaptel.conf misconfiguration. Maybe
for you is the same. Post your zaptel.conf to give more details.
Giorgio Incantalupo
DiegoF wrote:
Hello to all, I have a question. I am
Hi,
Im searching for GPLed softphones. I found
WengoPhone but actually not available for Asterisk PBX, only for Wengo network.
I found Kiax but only for IAX protocol.
Did you know a good GPLed softphones which works on
Windows ?
Thanks
Greg
This reply helps me how?
Of course I am pursuing the issue through their support channel.
Thanks,
Steve
John Novack wrote:
Sangoma has excellent technical support, and usually pretty quick to
respond
IF you are sure it isn't a configuration issue, your best resource is
Sangoma
Please report
-- Forwarded message --From:Doug Lytle
[EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate:Tue, 10 Oct 2006 16:25:11 -0400
Subject:Re: [asterisk-users] How big is *your* dialplan??Steve Murphy wrote: Hello! In my
-- Forwarded message --From:Doug Lytle
[EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Date:Tue, 10 Oct 2006 16:25:11 -0400
Subject:Re: [asterisk-users] How big is *your* dialplan??Steve Murphy wrote: Hello! In my
Hi to all,
I've a segmentation fault while using asterisk relatime conf with mysql db.
I've cretate sip_buddies and extensions tables into db and edit
res_mysql.conf, extconf.conf without any issues.
So when I start asterisk and my phone try to register using sip user
configured in my db,
Hi Paul,
Thanks for reply.
It is only recently that I have added an X100P - the asterisk server has
been doing purely SIP and IAX2 (to an ISDN gateway) before and
everything is perfect. There are no agents dropped etc.
It is purely that the zap channel (to X100P) gets released with no
errors
Hi C.,
Check out the pickupgroup and callgroup options in sip.conf -- these
should accomplish what you're looking for:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
More about this feature is defined here:
Hello all,
we want to use asterisk queues for a call center application. Depending on
the average waiting time in a queue, we want to make a decision to either
enqueue a call or transfer it to another site.
Are the applications available to query the average waiting time of a queue,
if possible
Steve Totaro wrote:
Kristian Kielhofner wrote:
Steve Totaro wrote:
I have NFAS setup on several quad port T1 cards (Sangoma).
It mostly works well with the exception that calls coming in on
channels 48,72, and 96 have no audio. I tried removing these
channels from zapata.conf with hopes
Hi,
I had a question. I am installing Asterisk on a windows
machine Astwind. I was wondering if it works with Dialogic card or if
it needed only digium card. Is there anyway Asterisk can work with a Dialogic
card or a Pika board?
Thanks in advance.
Vijay Naidu
Never Interrupt
Dear Issac,Makes sense.We got asked about moving back to firmware 1.6.7 as well and the official answer from Polycom is "not a problem"! Put the firmware on your server and remove the 2.0 firmware from this server and when the phone reboots it will grab the 1.6.7 firmware and load it on the phone.
Issac Simchayof wrote:
Polycom 601 with Sip 2.01
Anyone using Sip 2.01? I have upgraded my phones and now presence no longer
functions.
Buddy list shows all phones online but status does not change when someone
is on a call. Also blf does not function.
I am using trixbox, 1.67 was working
Dean,Tough call ... I haven't played with an IP 500 in a long time now and all that I know is Polycom officially doesn't support them.I'm sure the 2.0.1 firmware wasn't designed to ever work with bootroms 2.xx. I'm sure the problem lies with either the phone not supporting it or the bootrom not
hola a todos de nuevo, tengo el siguiente error cuando compilo el libunicall despues de compilar spandsp y libsupertone. esto es en fedora 5hello to all, I have the following error again when I compile
libunicall after compiling spandsp and libsupertone. this is in
fedora 5testcall.o: In function
Douglas, it seems to me that you don't understand how the extensions
of an asterisk dialplan relate to real life. As an example:
-= 135 extensions (657 priorities) in 31 contexts. =-
This from a box (yes one box) that has just 10 phones, and 6 lines.
Every s extension is considered an extension.
Has anyone built and run asterisk 1.4 beta2 on an intel mac?
Did it work?
I've got it building ok (once I installed Xcode, wget and bison)
However Asterisk hangs on startup (halfway through loading the modules).
I have not (yet) had time to debug it, but I wondered if anyone else had
done this
On Wed, Oct 11, 2006 at 11:25:08AM +0200, Remco Barendse wrote:
On Wed, 11 Oct 2006, Tzafrir Cohen wrote:
On Wed, Oct 11, 2006 at 09:14:55AM +0200, Remco Barendse wrote:
Hi list!
I recently upgraded to FreePBX 2.1.3 although I am not sure if this has
something to do with it.
On Wed, Oct 11, 2006 at 06:23:48PM +0200, Gregory Duchatelet wrote:
Hi,
I'm searching for GPLed softphones. I found WengoPhone but actually not
available for Asterisk PBX, only for Wengo network.
Have you actually tried it? Were you actually able to build it?
I found Kiax but only
The X-Ten is probably the most know free soft-phone availible. You can
find it at
http://www.xten.com/index.php?menu=Productssmenu=xlite
Gregory Duchatelet wrote:
Hi,
I’m searching for GPLed softphones. I found WengoPhone but actually
not available for Asterisk PBX, only for Wengo network.
On Wed, Oct 11, 2006 at 10:29:44AM -0500, DiegoF wrote:
Hello to all, I have a question. I am installing te110p, when I give ztcfg
him - v leaves the following error to me
ZT_CHANCONFIG failed on channel 25: No such device
Whats the max headcount you can have in a
conference bridge using ztdummy...since it is all sip based
incomming?
Don
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Wed, Oct 11, 2006 at 11:32:57AM -0400, Jens Vagelpohl wrote:
The call file created by the outgoing script file2fax.py specifies
3 retries in case of failure.
Fax may fail even if the phone call was successfull.
This just retries it within Asterisk, I
don't know if I could have
H,
hugolivude wrote:
For various reasons, I'm not too partial to UPnP, but maybe there needs
to be a SIP UA that uses UPnP to configure a NAT router for it, when an
RTP stream is begun?
Not following this part...
While I could probably never bring myself to enjoy (Microsoft's?)
Universal
Title: Re: [asterisk-users] How big is *your* dialplan??
No
one's system is redundant? :O
-Original Message-From: Douglas Garstang
[mailto:[EMAIL PROTECTED]On Behalf Of Douglas
GarstangSent: Tuesday, October 10, 2006 10:58 PMTo:
Asterisk Users Mailing List -
Hello,
I'm trying to upgrade an Asterisk 1.2 linux box to Asterisk 1.4. I
installed the following
-rw-r--r-- 1 root root 10908541 Sep 21 13:25 asterisk-1.4.0-
beta2.tar.gz
-rw-r--r-- 1 root root 993921 Sep 21 13:25 asterisk-addons-1.4.0-
beta1.tar.gz
-rw-r--r-- 1 root root
hola, este lo copie de internethello, this it copies it of Internetspan=1,1,0,cas,hdb3cas=1-15:1101dchan=16cas=17-31:1101
loadzone = usdefaultzone=usthanksOn 10/11/06, Giorgio Incantalupo [EMAIL PROTECTED]
wrote:Hi DiegoF,I had a similar problem, it was a zaptel.conf misconfiguration. Maybe
for
On Wed, 2006-10-11 at 11:19 -0400, Jerry Geis wrote:
I lost my internet connection today for a short time.
During that time 1.2.12.1 stopped talking to my phones.
Asterisk was still working as I got 2 voicemails. I have TDM analog
cards for incoming calls.
Anyway my cisco phones had X's
Hi,
I'm searching for GPLed softphones. I found WengoPhone but actually not
available for Asterisk PBX, only for Wengo network. I found Kiax but only
for IAX protocol.
Did you know a good GPLed softphones which works on Windows ?
Thanks
Greg
Apparently (from what I gathered
Asterisk can only be the proxy/server for MGCP, you connect other
devices to it. Asterisk can not be a user agent connecting to other
MGCP server.
On 10/11/06, Paul Ianas [EMAIL PROTECTED] wrote:
Hello everybody!
I have an Asterisk 1.2.12.1 server with SIP as the VoIP protocol.
What I
OK I have been battling with echo problems with asterisk on ISDN for a
few weeks now, and still can't solve it (although I think I have tried
everything I can find.)
I will try a post everything I think is possibly relevant that I can
remember with the hope someone can point me in the right
-= 1967 extensions (2838 priorities) in 285 contexts. =-
Shared services PBX with a dozen or so customers.
-ejay
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Murphy
Sent: Tuesday, October 10, 2006 3:17 PM
To: asterisk-users@lists.digium.com
Lol - use a real PC maybe :P
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: Wednesday, 11 October 2006 1:02 PM
To: asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
Xlite is not GPL!
Joe Dennick wrote:
The X-Ten is probably the most know free soft-phone availible. You
can find it at
http://www.xten.com/index.php?menu=Productssmenu=xlite
Gregory Duchatelet wrote:
Hi,
I’m searching for GPLed softphones. I found WengoPhone but actually
not available
On 10/11/06, Dinesh Nair [EMAIL PROTECTED] wrote:
On 10/11/06 21:15 Joseph said the following:
I quits on my as well, when I try to make a second call.
There is a bug report on it:
http://bugs.digium.com/view.php?id=7972
this seems like a configuration error within FreePBX and isnt really
Kristian Kielhofner wrote:
Steve Totaro wrote:
Kristian Kielhofner wrote:
Steve Totaro wrote:
I have NFAS setup on several quad port T1 cards (Sangoma).
It mostly works well with the exception that calls coming in on
channels 48,72, and 96 have no audio. I tried removing these
channels
As further info, here's the tail of the verbose logging (as enabled in
logger.conf). I have the complete log (but there are lots of irrelevant
SIP transactions for other phones/providers) which I can send if it
becomes helpful.
NB. The mysql server was down for maintenance at the time, so the
[EMAIL PROTECTED] wrote:
Hello all,
we want to use asterisk queues for a call center application. Depending on
the average waiting time in a queue, we want to make a decision to either
enqueue a call or transfer it to another site.
Are the applications available to query the average waiting
1 - 100 of 166 matches
Mail list logo
