Jason Becker wrote:
https://sip-communicator.dev.java.net/
Don't know the current state of functionality with Asterisk. I couldn't
get it to work many months ago - even with help from the developer.
We tried to get it working with two developers without luck...
I think it still has some way
Hi Matt:
That suggestion is possibly on the right track. It made me remember that -
although I'm not using Fedora, but SuSE 9.3, that it went through an
automatic network update just recently. After that, I tried updating the
Zaptel files from CVS and recompiling everything, but to no
On Saturday 20 August 2005 09:58, Scott Brown wrote:
Hi Matt:
That suggestion is possibly on the right track. It made me remember that -
although I'm not using Fedora, but SuSE 9.3, that it went through an
automatic network update just recently. After that, I tried updating the
Zaptel
Hi
Some basic mailing lists ethics:
1. writing in CAPITAL LETTERS usually indicates SHOUTING. Please don't
do that.
2. when you want to start a new message to the list, write a new
message, and don't just reply to an existing list message.
3. Proper English is also preffered, so readers spend
Hi
i just
implemented asterisk and is such a grate solution...i am using
polycom 301 and
501 phoneson lan a iam using g.711 and i have a
16 port linksys
switch...
the problem come
when somebody inside the network is making a call to
other extension
(in the same network) and
Hi
i just implemented
asterisk and is such a grate solution...i am using
polycom 301 and 501
phoneson lan a iam using g.711 and i have a
16 port linksys
switch...
the problem come when
somebody inside the network is making a call to
other extension (in
the same network)
On 03:57, Sat 20 Aug 05, Ing. Marlo R. Beltran G wrote:
Hi
i just implemented asterisk and is such a grate solution...i am using
polycom 301 and 501 phoneson lan a iam using g.711 and i have a
16 port linksys switch...
the problem come when somebody inside the network is making
Hi all
I've been doing some testing on realtime using mysql, an have a little
question that could not find the answer to or maybe its not posible at this
time.
Is there a way use register=.. on a DB using realtime. For the moment I
use it in sip.conf. It will help me a lot if this could
Matt Riddell [EMAIL PROTECTED] wrote:
Jason Becker wrote:
https://sip-communicator.dev.java.net/
Don't know the current state of functionality with Asterisk. I couldn't
get it to work many months ago - even with help from the developer.
Any reason you are looking for SIP and not IAX?
Seems the latest distro's have changed the layout of the linux source tree
needed to compile the zaptel stuff. I'm using FC3, upgraded the kernel, and
have the same issue. Was able to install new sources, but they too are
completely different tree layout compared to earlier stuff. The same is
Hi All
I am having another strnage problem :)
When I dialout on any number from asterisk, it use to
add a leading zero in dialed number
for e.g
I dial a number 5832876
and when I check the tracer's result of PSTN switch
that shows me call request for 05832876
thats why I can dial NWD and ISD
Kevin Walsh wrote:
Matt Riddell [EMAIL PROTECTED] wrote:
Jason Becker wrote:
https://sip-communicator.dev.java.net/
Don't know the current state of functionality with Asterisk. I couldn't
get it to work many months ago - even with help from the developer.
Any reason you are looking for SIP
Hello,
Asterisk is said to handle call routing and codec translation.
I would like to force transcoding function with asterisk but when I try to
force transcoding I get the errors:
codec not compatible or
WARNING[4425]: app_dial.c:1024 dial_exec: Had to drop call because I couldn't
make
hello
i m getting follwing messages in asterisk-1.0.9 what
is the reason can u pls tel me how to solve this
Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new:
Unable to allocate socket: Too many open files
Aug 20 13:06:09 WARNING[7706]: channel.c:311
ast_channel_alloc: Alert pipe creation
Hi
We have asterisk installed on a system and we had static noise
We changed the TDM card from a revision G to a revision F and the static
noise is gone
Any idea what is the difference between the two revisions that could make
this problem ?
Is there a way to downgrade the cards that we
Is anyone else with ViaTalk experiencing an outage right now? My DID
has been down since 5AM (8/20). Asterisk is unable to re-register or
connect for outbound calls. I have also tried calling support and
their number gives a fast busy.
___
I'm a ViaTalk system engineer. I just got up, I'm about to check it out.
Thanks for the heads up, I wouldn't have seen this until later.
I can tell you however, that our monitoring system did not kick any messages
to me about it acting funny in any way. I'll check it out and get back to
you.
I've restarted our switch via restart command from the CLI.
Anyone have a quickie answer as to why asterisk would suddenly just stop
responding? I was able to issue the restart command but I couldn't do sip
show peer num and couldn't show channels, etc This is very
disconcerting
We've
Last note on this, I figured out it was due a freeze in registrations that
we've been having an issue with on asterisk. I'm writing a custom monitoring
script using sipsak for testing registrations, which would SMS the
engineering dept when registrations stop working.
Cheers,
Sherwood McGowan
Thanks for the quick response (and call), its running again!
On 8/20/05, Sherwood McGowan [EMAIL PROTECTED] wrote:
I've restarted our switch via restart command from the CLI.
Anyone have a quickie answer as to why asterisk would suddenly just stop
responding? I was able to issue the restart
Your only recourse is to get your new carrier to realize that the numbers
have been released and to proceed with the porting despite the fact that
they have not received the notification.
Thanks for the info! I've forwarded your message to the new carrier
in hopes that they'll be able to do
concur that the best way around this is to perioically restart.
FWIW this is my restart script which I invoke from cron in the middle
of the night...
#!/bin/bash
ASTERISK=/usr/sbin/asterisk
RMMOD=/sbin/rmmod
MODPROBE=/sbin/modprobe
ZTCFG=/sbin/ztcfg
echo Stopping
$ASTERISK -rx stop when
On Sun, Aug 21, 2005 at 12:32:06AM +1000, Mark Edwards wrote:
concur that the best way around this is to perioically restart.
This is ignoring the problem rather than solving it.
If you both rmmod zaptel and restart asterisk, why not simply reboot?
All of those sleeps there produce a nice
Sorry if this is a resend, but it didn't appear to go the first time.
Sorry if this is not the correct place to post this.
I have downloaded the cid_rewrite scripts that are located at:
http://www.muware.com/asterisk/ to my AAH v1.1 system.
I apologize for my ignorance, but it says
hi
PSTN -- [Teles ISDN / Asterisk] -- SIP client
When call is made through ISDN, no matter if taken from PSTN or
Asterisk side, person in PSTN side can hear perfectly but in Asterisk
side I only hear a very scrambled or very low quality voice, words
repeated several times. Same is with echo test
Why?
--- Innocent Evil [EMAIL PROTECTED] wrote:
Please change the subject to 'Advertisement of a
VoIP Provider'
-Original Message-
From: [EMAIL PROTECTED]
Sent: Thu, 18 Aug 2005 11:55:50 -0700 (PDT)
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Searching
hi,
my ATA186 confige as SIP(600) on my
Asterisk ,it only can be called in , but can not call
out .
between ATA186 and astersik there is
aVPNon two netscreen 5gt.
who can showme some idea ?
ATA 186 configure same as SIP.conf
SIP.conf on Asterisk :
[general]port =
5060
; Port to bind
You can store your entire sip.conf using RealTime. That should allow for
register = to work.
-Matthew
From: Guillermo Krepper [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sat, 20 Aug 2005 13:05:02 +0200
To:
I have a TDM400 running telco lines on ZAP2-4
My after hours config is supposed to receive the incoming call then
divert it to my home phone by calling out one of the other zap channels
available.
console output as such...
Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1,
I am using a HFC-S card in nt mode with zaphfc driver to connect an
internal isdn bus. I would like to signal an incoming call on, let's
say, 4 phones. Right now I use:
Dial(Zap/g1/21Zap/g1/22Zap/g1/24Zap/g1/23Zap/g1/29,,t)
where g1 are my two isdn channels provided by HFC-S card an the
The java client you mention states on it's webpage it has to install a local
.dll/.so and that it only works for x86 Windows or Linux.
Does anyone know of one that's completely in Java?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent:
how many connection do you have from your asterisk to the old pbx?
i think on 1 ISDN connection its only possible to let 2 phones ring, because
1 ISDN 2 channels...
Nico
- Original Message -
From: Arik Funke [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, August
Aug 17, 2005, at 5:44 AM, Tom Hayden wrote:
I have experienced pretty nasty echo on my PRI w/TE110P. The echo was
only coming from other POTS lines, because cell phones already have
echo cancellation, and other PBX's had the same. I resolved the
problem by turning on the AGGRESSIVE option and
I think I spotted what's going on.
When I sift through the sip debug I see that my server is looking for my
number in the context I take the calls into. Problem is its not there.
I was expecting * to dump the call into the exten=s,1,blahblahblah
logic but it's not doing that.
Now I think
On Aug 18, 2005, at 3:07 AM, Stephen wrote:
Hi All,
How can I lock the extension in Asterisk?
For example , my extension is 1000 and I am away for business trip.
I want to lock my extension during my absence.
Can it be done in Asterisk?
regards,
Stephen
You could write a little script to
Does VoicemailMan have to be installed ? Why not available. I have setup a
mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup
up using *97.
My *97 code in extensions.conf:
exten = *97,1,Answer
exten = *97,2,VoicemailMain([EMAIL PROTECTED])
exten = *97,3,Hangup
You spelled Voicemailmain wrong somewhere. Or your extensions are not in
sync with the conf file.
Verify that the extensions.conf is correct then 'extensions reload'.
You can also do show dialplan context to view what is currently loaded
in memory.
-Matthew
From: Angus Comber [EMAIL
Pulu Anau wrote:
The java client you mention states on it's webpage it has to install a local
.dll/.so and that it only works for x86 Windows or Linux.
Does anyone know of one that's completely in Java?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Sherwood McGowan wrote:
Anyone have a quickie answer as to why asterisk would suddenly just stop
responding? I was able to issue the restart command but I couldn't do sip
show peer num and couldn't show channels, etc This is very
disconcerting
Your SIP channel driver was deadlocked. This
Why my X100P detect the ring after 3 o 4 seconds?
The funny thing that when I have an incoming call asterisk receive a
signal but the commands start after 3 or 4 seconds. Moreover, when the
call end the hungup has the same delay.
any ideas?
___
On Sat, 20 Aug 2005, Gulzar Hussain wrote:
I am having another strnage problem :)
When I dialout on any number from asterisk, it use to
add a leading zero in dialed number
for e.g
I dial a number 5832876
and when I check the tracer's result of PSTN switch
that shows me call request for
On Sat, 20 Aug 2005, Nico Giefing wrote:
how many connection do you have from your asterisk to the old pbx?
i think on 1 ISDN connection its only possible to let 2 phones ring, because
1 ISDN 2 channels...
This is a limitation in Asterisk, not ISDN. Asterisk reserves a B-channel
for each
obviously i'm looking for a direct connect w/ asterisk (SIP or IAX), so
no proprietary equip, but if you provide, or know of a provider that has
any of these available, please let me know.
(Tennessee)
423-869-
(Kentucky)
606-337-
606-248-
606-242-
Kevin Walsh wrote on Friday, 19 August 2005 6:58 PM:
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Friday 19 August 2005 21:27, Kevin Walsh wrote:
I'll send the modified Makefiles to anyone who needs them.
May I humbly request they be attached to a feature request on Mantis?
I've been
Eric Wieling aka ManxPower wrote:
Sean Rima wrote:
Does anyone have any experience of these, I have been offered one and am
thinking of adding sticking it onto the back of my Asterisk box and just
ignore the WAN port if possible, It would be to stick my exisiting
phones onto the asterisk box
On Sat, 20 Aug 2005, Paul Hewlett wrote:
This does nothing but tells you what would happen. If your
modprobe.d/zaptel file is correct the the output from this command will be
loading of zaptel,wcfxs and an execution of ztcfg. In other words you do not
have to modprobe more than one
Your only recourse is to get your new carrier to realize that the numbers
have been released and to proceed with the porting despite the fact that
they have not received the notification.
Thanks for the info! I've forwarded your message to the new carrier
in hopes that they'll be able
This is very odd.
I've been able to fix the problem by adding a DID route as follows
exten = 9738281625,1,Dial(SIP/2208)
Without this line it doesn't work. I've even rolled back from the latest
CVS head to the release 1.0.8 and still it don;t work.
I'm flumaxed!!
Mark
Mark Phillips wrote:
Hi All,
I've got Asterisk working and am trying to configure with Sipgate. I can make out going calls. Incoming calls show up on the AMP panel with the trunk showing red. However, the call does not go to the extension.
I initally configured Asterisk by editing the config files. I have followed the
Trevor G. Hammonds [EMAIL PROTECTED] wrote:
Kevin Walsh wrote on Friday, 19 August 2005 6:58 PM:
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
May I humbly request they be attached to a feature request on Mantis?
I've been less than humbly requested not to do that sort of thing any
On Sat, Aug 20, 2005 at 02:57:50AM +0100, Kevin Walsh wrote:
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Friday 19 August 2005 21:27, Kevin Walsh wrote:
I'll send the modified Makefiles to anyone who needs them.
May I humbly request they be attached to a feature request on Mantis?
You can file a complaint with the (insert regulator body here)...
You can call your provider every 5 minutes complaining bout the situation...
Other than that... no..
Regards,
Derek
- Original Message -
From: C. Hatton Humphrey [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
You can file a complaint with the (insert regulator body here)...
You can call your provider every 5 minutes complaining bout the situation...
Other than that... no..
Regards,
Derek
- Original Message -
From: C. Hatton Humphrey [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Hi Anton,
I recently contacted polycoms tech support asking if their phones supported
XML pushed information to which they replied that only model 600 had a
microbrwoser capable of reading dhtml files and such.
My question to the community is: is somebody doing any XML info push to any
On Sat, 2005-08-20 at 13:11 -0600, Tom wrote:
Pulu Anau wrote:
The java client you mention states on it's webpage it has to install a local
.dll/.so and that it only works for x86 Windows or Linux.
Does anyone know of one that's completely in Java?
-Original Message-
From:
We are looking at moving more of our business to distant
locations. Im looking at two different network configurations and would
like some thoughts or comments.
Scenario 1:
-We take our existing T1 route it without any conversion
etc. trans-Atlantic to remote site via carrier Y
-at
On Wed, Aug 03, 2005 at 11:28:19AM -0500, [EMAIL PROTECTED] wrote:
10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary
D-channel of span 1 (Gavin Hamill)
Date: Wed, 3 Aug 2005 15:32:48 +0100
From: Gavin Hamill [EMAIL PROTECTED]
Subject: [Asterisk-Users] Inter-Tel AXXESS
Sean Rima wrote:
Eric Wieling aka ManxPower wrote:
Sean Rima wrote:
Does anyone have any experience of these, I have been offered one and am
thinking of adding sticking it onto the back of my Asterisk box and just
ignore the WAN port if possible, It would be to stick my exisiting
phones onto
Please attach your sip.conf file also
Joshua Abbott wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
Currently we have a server setup for Asterisk(*) and a TFTP server. My
extension has been setup with Asterisk(*) and downloads the
information from the TFTP server correctly (I've
I have grabbed the latest zaptel, libpri, asterisk, and asterisk-addons from
CVSHEAD. Everything complies and installs well, but when I go to run
asterisk it aborts with:
[chan_zap.so]Aug 20 21:18:53 WARNING[11840]: loader.c:314 __load_resource:
/usr/lib/asterisk/modules/chan_zap.so:
Last tuesday I moved the asterisk server from 1.0.7 to 1.0.9, while
leaving the zaptel drivers at 1.0.7 because it was a lunchtime
update. This is a box with two TE405Ps in it, and all eight ports
in use.
Today I unloaded the 1.0.7 drivers and replaced them with 1.0.9 and
oh boy... two of the 8
Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Aug 20, 2005 at 02:57:50AM +0100, Kevin Walsh wrote:
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Friday 19 August 2005 21:27, Kevin Walsh wrote:
I'll send the modified Makefiles to anyone who needs them.
May I humbly request they be
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello
I currently have internet service through MediaCom (Cable Internet)
and need to find a VOIP provider that is compatible with Asterisk and
Cable Internet.
Any ideas?
I'm in Missouri about 1.5 hours west of St Louis, MO in a town called
Hermann
Hello,
I patched asterisk cvs head sources with
http://juraj.bednar.sk/work/software/asterisk/messaging/
and presnce patch without success.
asterisk send 405 method not allowed to sender.
I use polycom ip300.
Harry
Hello,
I patched asterisk cvs head sources with
http://juraj.bednar.sk/work/software/asterisk/messaging/
and presnce patch without success.
asterisk send 405 method not allowed to sender.
I use polycom ip300.
Harry
On 7/7/05, Adam Dobrin [EMAIL PROTECTED] wrote:
Also, around the same time, I isolated the IRQ that my zaptel cards were
on. (so neither zaptel card shared its IRQ).
you can see what IRQ's are in use with
lspci -vb
This is more likely to be the cause of the fix.
Adam Dobrin wrote:
66 matches
Mail list logo
