2010/9/15 asterisk asterisk aster...@ck-lee.com
Olivier,
You should find out the SMS tab in the handset but not in the web service.
Did you IP pone work?
CK
Hi,
My phone is working OK but there is no SMS menu showing, though you can see
this menu all around the user manual.
How did you
Hi danny,
U r the one responding for me. Thanks a lot.
How do we make it visible to asterisk developers.
Thanks,
Ashik
On Tue, Sep 14, 2010 at 7:30 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
Hello,
I've had the problem again, but there is no core.pid file in my
/etc/asterisk...
I have :
dumpcore = yes ; Dump core on crash (same as -g at startup)
in asterisk.conf
This time my CLI was open, and I was suddenly disconnected... just like
that.
There is nothing in my debug file
On the S675IP SMS is here:
Messaging - SMS - Settings
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New to Asterisk? Join us for a live introductory webinar every Thurs:
The reboot occured a 10:11:11, this my debug log :
...
[Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp'
[Sep 15 10:11:11] DEBUG[12353] pbx.c: Function result is '252227026'
[Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Set'
[Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Macro'
[Sep 15
Hello,
these are my settings in asterisk.conf :
;maxcalls = 10 ; Maximum amount of calls allowed
;maxload = 0.9 ; Asterisk stops accepting new calls if the load average
exceed this limit
;maxfiles = 1000 ; Maximum amount of openfiles
;minmemfree = 1 ; in MBs, Asterisk stops accepting new
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp(SIP/INTERTELin-, 3252480333 = pbx formule) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:2]
Is there a way skip / ignore the member whose status is busy in the Queue.
I have two channel member in queue and i have set the peer limit 2 for these
members.
I want to skip those member who are currently on the call (answered to
calls) and now their status is busy, if Queue see the busy
Hi Group,
I am currently facing a dead end and any help will be much appreciated.
I have an a104d installed in an asterisk box, two of which is configured on
ISDN
pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am
getting one way audio when a local on the hipath
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is busy in the Queue.
I have two channel member in queue and i have set the peer limit 2 for
these members.
I want to skip those member who are currently on the call (answered to
calls) and now their status is busy,
Jonas,
everyone here supports you in your effort to get a good Asterisk
installation going, but could you ... maybe restrain yourself a little
bit and reduce the number of hasty postings you are sending to this
mailing list?
Thank you,
Philipp
--
You mean, I need to check the DEVICE_STATUS of both (sip) users before
sending the caller into queue, otherwise skip the caller from going into
Queue by using ExecIf.
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 3:16 PM, Gareth Blades
list-aster...@skycomuk.comwrote:
Shariq
Yes something like this. Note the Execif syntax I have used is for
asterisk 1.6
exten = s,n,Set(AGENTSBUSY=yes)
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1009} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1010} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten =
Hi Group,
I was able to resolve the problem by disabling the echo cancellation in a104d
and using the same dahdi config.
Thanks...
- Original Message
From: leonimar cape leo_mac...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 15, 2010 6:12:35 PM
Hello Philipp,
I know I post a lot concerning this issue, but this is because this
problem occurs on a production system and I feel very hot breathing down
my neck.
I have tested during several weeks my implementation on a test system
which is similar to the production system. The only
I cant help you with fixing the actual cause but have you considered
moving the mysql and as much of the associated logic to an AGI running
something like a perl or php script. From previous posts that generally
seems to me the more reliable way of making mysql queries.
Jonas Kellens wrote:
Gareth
Usualy the queue has the ability to know if the agent is INUSE and skip
them.. you can simply use ringinuse=no to the queues.conf under the queue
itself or the general section and that's it .. no need for the whole
dialplan.. as you are using SIP members.
Salam
-Original Message-
Hi Ott,
Have you made it work with Asterisk and Aastra IP Phone. I am also trying
the same thing, in Asterisk it shows registered OK but when I dial from
extension to extension, call is failed...
Please let me know have you made it work...:(
On Mon, Jul 13, 2009 at 11:46 PM, Ott Rose
I hope someone has helped poor Rob, I would as I am just over the bridge
in Bristol, UK but some evil internet scammer has stolen all my money! ;)
Cheers!
On 15/09/10 12:14, Rob Fugina wrote:
It is with deep sorrow and broken heart that am sending you this mail.
Am in deep need and my
Rough area. Consider yourself lucky you haven't been ripped apart :P
Pete wrote:
I hope someone has helped poor Rob, I would as I am just over the bridge
in Bristol, UK but some evil internet scammer has stolen all my money! ;)
Cheers!
On 15/09/10 12:14, Rob Fugina wrote:
It is with
2010/9/15 Randy R randulo2...@gmail.com
On the S675IP SMS is here:
Messaging - SMS - Settings
No SMS entry is showing on Settings/Messaging page, here.
How did you set your S675IP ?
Did you use any autoconfiguration or country menu ?
--
I really need you to help me out of here.
On 2010-09-15, Gareth Blades list-aster...@skycomuk.com wrote:
Rough area. Consider yourself lucky you haven't been ripped apart :P
Pete wrote:
I hope someone has helped poor Rob, I would as I am just over the bridge
in Bristol, UK but some evil
- Original Message -
Rough area. Consider yourself lucky you haven't been ripped apart :P
Pete wrote:
I hope someone has helped poor Rob, I would as I am just over the
bridge
in Bristol, UK but some evil internet scammer has stolen all my
money! ;)
Cheers!
On 15/09/10
On 09/15/2010 12:59 PM, Gareth Blades wrote:
I cant help you with fixing the actual cause but have you considered
moving the mysql and as much of the associated logic to an AGI running
something like a perl or php script. From previous posts that generally
seems to me the more reliable way of
http://blog.tmcnet.com/blog/rich-tehrani/google/new-scam-held-up-at-gunpoint-in-wales.html
Can't believe (s)he's tried to convince us (s)he's genuine :)
http://www.railroad.net/forums/viewtopic.php?f=127t=74905
Been stuck in that hotel for at least two weeks apparently! Must have
missed their
Dear Tarek,
IN_USE is other then the BUSY status, i want to skip the BUSY agent but not
IN_USE
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 4:07 PM, Tarek Sawah tareksa...@hotmail.com wrote:
Gareth
Usualy the queue has the ability to know if the agent is INUSE and skip
Hi!
I know I post a lot concerning this issue, but this is because this
problem occurs on a production system and I feel very hot breathing down
my neck.
Why not reduce the pressure and revert to 1.4.30 for the production
system until you have figued out the issue? That will give you more
I also want to hear the experience of yours with Synway Cards.
--
Regards,
Shariq Khan
0333-3501125
On Mon, Sep 13, 2010 at 12:47 AM, Anita Hall anita.h...@simmortel.comwrote:
Hi
Does anyone have experience with Synway cards like SHD-240D-CT/PCI with
asterisk and SynAst driver ?
Are they
On 15/09/10 12:14, Rob Fugina wrote:
It is with deep sorrow and broken heart that am sending you this mail. Am in
deep need and my situation is lamentable. my family and I decide to come
visit Wales,United Kingdom for a short vacation. To our greatest dismay we
were attacked and ripped apart
He's fortunate that the hotel insists he stay there until his situation
improves.
--Don
Rough area. Consider yourself lucky you haven't been ripped apart :P
Pete wrote:
I hope someone has helped poor Rob, I would as I am just over the bridge
in Bristol, UK but some evil internet scammer
2010/9/15, DHAVAL INDRODIYA dhaval.it01...@gmail.com:
Hello i have tried to convert through sphinx as suggested by Nickolay
i am not getting convert my simple audio file.
i am having following error while i fire following command
pocketsphinx_continuous -infile /usr/etc/ask-propertyid.WAV
Just see what the function returns when the agents are busy. You said in
your first post you want to skip the queue if both agents are already on
a call. The dialplan I gave was just an example. You will need to modify
it to do exactly what you want.
I have asterisk emulating a traditional
Hi all,
Recently I have instaled one Digium TDM410 on my Asterisk. After instaled ,
I can do outgoing calls but I cant receive calls. I receive the following
messages:
chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654]
chan_dahdi.c: Got event 18 (Ring Begin)...[Sep
On 09/15/2010 02:03 PM, Philipp von Klitzing wrote:
Hi!
I know I post a lot concerning this issue, but this is because this
problem occurs on a production system and I feel very hot breathing down
my neck.
Why not reduce the pressure and revert to 1.4.30 for the production
Dan Journo wrote:
Anyone else got a theory?
Same message here in the States. The person here had his Gmail account
cracked.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
On 15 Sep 2010, at 13:22, Jonas Kellens wrote:
I have indeed found the core file in /tmp (that is where 'locate' does
not look huh...)
'updatedb'?
S
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On 09/15/2010 07:20 AM, Flavio Miranda wrote:
Recently I have instaled one Digium TDM410 on my Asterisk. After
instaled , I can do outgoing calls but I cant receive calls. I receive
the following messages:
chan_dahdi.c: Got event 2 (Ring/Answered)...
[Sep 14 11:24:44] NOTICE[2654]
On Tue, 14 Sep 2010, Joe Freeman wrote:
Anyone have a good provider for International (US/Canada at least) 800
termination/origination? I have a customer that had us port one of their
800 numbers and apparently didn't realize that they had published that
number in Canada as well. Our current
Hi Jonas!
It indicates to be a binary file, however I have not found instructions on
dealing with this @ the link you gave me.
Can you give me instruction on how to handle the core.pid file ?
Could I ask you again to make an effort to reduce your number of daily
postings to this list? If
On 09/15/2010 02:45 PM, Steve Howes wrote:
On 15 Sep 2010, at 13:22, Jonas Kellens wrote:
I have indeed found the core file in /tmp (that is where 'locate' does
not look huh...)
'updatedb'?
S
Off course I did that, Steve, before I did a locate on 'core'. But
doesn't locate
On Wed, Sep 15, 2010 at 1:43 PM, Olivier oza_4...@yahoo.fr wrote:
On the S675IP SMS is here:
Messaging - SMS - Settings
No SMS entry is showing on Settings/Messaging page, here.
How did you set your S675IP ?
Did you use any autoconfiguration or country menu ?
We don't use SMS on
On 09/15/2010 05:45 AM, Steve Howes wrote:
On 15 Sep 2010, at 13:22, Jonas Kellens wrote:
I have indeed found the core file in /tmp (that is where 'locate' does
not look huh...)
'updatedb'?
S
off topic, but updatedb deliberately doesn't usually look in /tmp
--
Hi,
I went over your dialplan and though it looks fine at first glance, but
because I have no experience with Asterisk 1.6, so I would like to ask if
commas in mysql query are ok without escape character? In my asterisk 1.4 I
would type it like:
SELECT var1\, var2\, var3 FROM ...
Other things
As Kevin said, you need to define an 's' extension where the calls will be
answered. Seems like you are using default configuration. Open file
'extensions.conf' in /etc/asterisk folder and look for context named
[default]. If it is not there, create one and add something under it, e.g.,
[default]
On 09/15/2010 03:47 PM, Zeeshan Zakaria wrote:
Hi,
I went over your dialplan and though it looks fine at first glance,
but because I have no experience with Asterisk 1.6, so I would like to
ask if commas in mysql query are ok without escape character? In my
asterisk 1.4 I would type it
Clearly, if Word cannot explain the anguish in his heart,
Mr. Fugina should be using OpenOffice!
Cheers.
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Yes, only on the handset. My line does not support SMS so sending out is
failed.
On Wed, Sep 15, 2010 at 9:28 PM, Randy R randulo2...@gmail.com wrote:
On Wed, Sep 15, 2010 at 1:43 PM, Olivier oza_4...@yahoo.fr wrote:
On the S675IP SMS is here:
Messaging - SMS - Settings
No SMS entry is
I encounter problem in using Dual WAN with load balancing on asterisk
1.6.2.11.
My problem is registration of one VOIP provider. I can dial out but not
probably answer. It drops. One of the error message is
SIP/2.0 404 not found.
I am not sure about the problem but note that it may be related to
Hello,
I'm having some problems with a total SIP Asterisk scenario, some extensions
when make internal and outgoing calls can't hear very well the other party,
not echo, not packet lostthe problem is that the volume seems to be very
low...what could be happening? i'm not sure what to check
I am changing a system from zap to DAHDI.
I removed everything zap. when doing the command:
sh -x /etc/init.d/dahdi start, I see
initlog -q -c 'modprobe wct4xxp'
sh: /sbin/ztcfg: No such file or directory
FATAL: Error running install command for wct4xxp
doing updatedb then,
locate zap returns
This suddenly started appearing and I'm not sure why. Any ideas?
asterisk*CLI module load chan_skype.so
Unable to load module chan_skype.so
Command 'module load chan_skype.so' failed.
[Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error
loading module 'chan_skype.so':
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Wednesday, September 15, 2010 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] changing from zap to
On 09/15/2010 10:06 AM, Jerry Geis wrote:
I am changing a system from zap to DAHDI.
I removed everything zap. when doing the command:
sh -x /etc/init.d/dahdi start, I see
initlog -q -c 'modprobe wct4xxp'
sh: /sbin/ztcfg: No such file or directory
FATAL: Error running install command for
On 09/15/2010 10:09 AM, Richard Kenner wrote:
This suddenly started appearing and I'm not sure why. Any ideas?
asterisk*CLI module load chan_skype.so
Unable to load module chan_skype.so
Command 'module load chan_skype.so' failed.
[Sep 15 11:08:25] WARNING[12274]: loader.c:429
The Asterisk Development Team has announced the release of Asterisk 1.4.36. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.36 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 1.6.2.12.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.12 resolves several issues reported by the
community and would have not been
You wouldn't have a udev rule set to run ztcfg configured in
/etc/modprobe.d by any chance would you?
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
Shaun
Yes I did in
Have you checked that the codec order on the phone matched the order set
on the server?
On Wed, 2010-09-15 at 17:04 +0200, Danny Dias wrote:
Hello,
I'm having some problems with a total SIP Asterisk scenario, some
extensions when make internal and outgoing calls can't hear very well
the
Jerry Geis wrote:
You wouldn't have a udev rule set to run ztcfg configured in
/etc/modprobe.d by any chance would you?
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
Shaun
On Wed, Sep 15, 2010 at 10:15:03AM -0500, Shaun Ruffell wrote:
On 09/15/2010 10:06 AM, Jerry Geis wrote:
I am changing a system from zap to DAHDI.
I removed everything zap. when doing the command:
sh -x /etc/init.d/dahdi start, I see
initlog -q -c 'modprobe wct4xxp'
sh:
Anybody else notice that the 1.6.2.12 download has a version and
changelog for 1.6.2.12-rc1?
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.12.tar.gz
Ryan
--
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-- Bandwidth and Colocation Provided
On Wed, Sep 15, 2010 at 9:14 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
I have no experience with this, so I post my output :
Read doc/backtrace.txt it will explain how to generate a backtrace
from a core dump.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
I am not sure about the problem but note that it may be related to incorrect
IP being used. Sometimes, WAN 1 and sometimes WAN 2
Most likely. Get a provider that uses IP authentication rather than
registrations, and enable access from both of your WAN IPs. All set.
Luki
--
Yes my friend...CONFIRMED!!! G729 on both sides
2010/9/15 Ishfaq Malik i...@pack-net.co.uk
Have you checked that the codec order on the phone matched the order set
on the server?
On Wed, 2010-09-15 at 17:04 +0200, Danny Dias wrote:
Hello,
I'm having some problems with a total SIP
On Wed, Sep 15, 2010 at 11:54 AM, Ryan Wagoner rswago...@gmail.com wrote:
Anybody else notice that the 1.6.2.12 download has a version and
changelog for 1.6.2.12-rc1?
I can confirm, asterisk-dev notified.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com |
Hi,
I'm experiencing the same problem, with identical symptoms.
I also noticed that after making a call attempt, I see this stuck TCP
connection pair until I stop and restart the asterisk server process.
# netstat -an | grep 1314
tcp0 0 0.0.0.0:13140.0.0.0:*
We use Excel Telecom (recently purchased by Matrix) for International
and toll-free origination and termination.
Alex
On 09/15/2010 06:04 AM, Jeff LaCoursiere wrote:
On Tue, 14 Sep 2010, Joe Freeman wrote:
Anyone have a good provider for International (US/Canada at least) 800
On Wed, Sep 15, 2010 at 6:08 PM, Danny Dias ing.diasda...@gmail.com wrote:
Yes my friend...CONFIRMED!!! G729 on both sides
If the problem happen with SIP to SIP calls and with the same codec, the
problem is inside the phone.
Check if you can pump up the volume inside his configuration.
What
On 09/15/2010 10:35 AM, Tzafrir Cohen wrote:
On Wed, Sep 15, 2010 at 10:15:03AM -0500, Shaun Ruffell wrote:
On 09/15/2010 10:06 AM, Jerry Geis wrote:
I am changing a system from zap to DAHDI.
I removed everything zap. when doing the command:
sh -x /etc/init.d/dahdi start, I see
initlog -q
On 09/15/2010 10:35 AM, Jerry Geis wrote:
Jerry Geis wrote:
You wouldn't have a udev rule set to run ztcfg configured in
/etc/modprobe.d by any chance would you?
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:
On 10-09-15 05:25 AM, Jonas Kellens wrote:
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp(SIP/INTERTELin-, 3252480333 = pbx formule) in new stack
[Sep 15 11:16:32] -- Executing
Hi,
On 09/15/2010 04:04 PM, Danny Dias wrote:
Hello,
I'm having some problems with a total SIP Asterisk scenario, some
extensions when make internal and outgoing calls can't hear very well
the other party, not echo, not packet lostthe problem is that the
volume seems to be very
On 10-09-15 12:13 PM, Paul Belanger wrote:
On Wed, Sep 15, 2010 at 11:54 AM, Ryan Wagonerrswago...@gmail.com wrote:
Anybody else notice that the 1.6.2.12 download has a version and
changelog for 1.6.2.12-rc1?
I can confirm, asterisk-dev notified.
Odd, not sure how this happened, but I'll be
/ After removing everything in modprobe.conf that was ztcfg related:
// alias eth0 tg3
// alias eth1 tg3
// alias scsi_hostadapter ata_piix
// alias usb-controller ehci-hcd
// alias usb-controller1 uhci-hcd
//
// and rebooting it still happens. same error. is there another place I
//
Hello Adriá...
We are using Linksys 942, softphones Xlite...it's a macro pbx, with almost
1000 users, we've checked the gain and volume on the phones :(
2010/9/15 Adrià Vidal adriavi...@gmail.com
On Wed, Sep 15, 2010 at 6:08 PM, Danny Dias ing.diasda...@gmail.comwrote:
Yes my
Thanks Sebastian,
It's the same firmware version for all our linksys phones...and we have
hundreds of pbx's runnning this firmwares versions without any problem
2010/9/15 Sebastian s...@open-t.co.uk
Hi,
On 09/15/2010 04:04 PM, Danny Dias wrote:
Hello,
I'm having some problems with a
On 09/15/2010 12:42 PM, Leif Madsen wrote:
On 10-09-15 05:25 AM, Jonas Kellens wrote:
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp(SIP/INTERTELin-, 3252480333 = pbx
Hi,
On 09/15/2010 04:19 AM, t. k wrote:
Hi
I'm sorry.
I mailed the same question again.
because, it cannot be yet solved.
any ideas with asterisk?
[Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username
mismatch, have, digest has a...@192.168.0.1[aug 20 14:40:12]
On 09/14/2010 06:33 PM, Dan Journo wrote:
Hi,
It seems ive broken my settings and now, asterisk isnt detecting my DTMF
tones.
What kind of diagnostics can I do to work this out?
I've set the extension in sip.conf to everything listed on this page but
no result. I've also played around
Ok. Problem solved .
Thank you very much!!!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Wed, 15 Sep 2010 09:56:36 -0400
From: zisha...@gmail.com
To: kpflem...@digium.com; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] incoming
On 09/15/2010 11:44 AM, Jerry Geis wrote:
/etc/modprobe.conf?
Shaun,
This is what is in my modprobe.conf file presently.
more /etc/modprobe.conf
alias eth0 tg3
alias eth1 tg3
alias scsi_hostadapter ata_piix
alias usb-controller ehci-hcd
alias usb-controller1 uhci-hcd
Sorry
Dear Gareth,
DEVICE_STATE function is not available in asterisk, even DEVSTATE does not
work for me in asterisk 1.4.35. Any other method function to check the
channel status
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 5:11 PM, Gareth Blades
list-aster...@skycomuk.comwrote:
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shariq Khan
Sent: Wednesday, September 15, 2010 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Skip Busy Agents/Channels from
By somehow I made it work by having T38 passthru in both Asterisk and
SPA3102.
Thanks for the comments..
On Tue, Sep 14, 2010 at 7:05 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Hi,
I tried to send fax from Linksys to Grandstream by configuring openSER
account.. that works fineonly
The Asterisk Development Team has announced the release of Asterisk 1.6.2.13.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
This release resolves an issue where the .version and ChangeLog files were not
updated for 1.6.2.12. Asterisk
Sorry about that, that's what you said but I didn't see that. What does
'grep zt /etc/modprobe.d/*' return then?
grep zt /etc/modprobe.d/*
/etc/modprobe.d/modprobe.conf.dist:alias block-major-29-* aztcd
jerry
--
_
--
Hi!
DEVICE_STATE function is not available in asterisk, even DEVSTATE does not
work for me in asterisk 1.4.35. Any other method function to check the
channel status
There is a backport available for 1.4:
http://www.voip-info.org/wiki/view/Asterisk+func+device_State
I assume that with does
nexVortex (http://bit.ly/9bEw9e) can do this. They use Global for TF. They
can support both US and CA origination.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Freeman
Sent: Tuesday, September 14,
I have an Asterisk 1.6.0.28 system, with a queue called 'marketing'.
Everything appears normal, but the status of the members never changes from
'not in use', even if they are being rang or are in a call.
Members are added like so:
queue add member SIP/1406 to marketing penalty 0 as
Hi,
I think ive found a bug but need someone to double check.
Whenever I issue a reload in Asterisk, any realtime extensions stop receiving
calls.
I have to reboot the sip phones in order to get them to re-register.
Can anyone see if they have a similar problem?
Asterisk 1.4.32
Mysql
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin
Sherrill
Sent: Wednesday, September 15, 2010 2:32 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queue member status not changing
I have
On Wed, Sep 15, 2010 at 6:04 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Tue, 14 Sep 2010, Joe Freeman wrote:
Anyone have a good provider for International (US/Canada at least) 800
termination/origination? I have a customer that had us port one of their
800 numbers and apparently
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, September 15, 2010 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Bug with Realtime?
Hi,
I think ive found a
On 09/15/2010 01:48 PM, Jerry Geis wrote:
Sorry about that, that's what you said but I didn't see that. What does
'grep zt /etc/modprobe.d/*' return then?
grep zt /etc/modprobe.d/*
/etc/modprobe.d/modprobe.conf.dist:alias block-major-29-* aztcd
jerry
Somewhere on your system you
By reload you mean sip reload or just any reload in general?
Reload in general.
It might be an issue only with the Polycom sip phones. Not been able to test
any others. I'll try a software phone tomorrow.
--
_
-- Bandwidth
On 09/15/2010 09:41 PM, Dan Journo wrote:
Hi,
I think ive found a bug but need someone to double check.
Whenever I issue a reload in Asterisk, any realtime extensions stop
receiving calls.
I have to reboot the sip phones in order to get them to re-register.
Can anyone see if they have a
Hi,
I'm using the CallTime and a few other variables to name a recording so that I
can then take the wav file name and see when it was recorded, and what the
recording contains.
However, since ${CDR(start)} contains a space in part of the date, the filename
becomes corrupted when I use samba
On 10-09-15 03:41 PM, Dan Journo wrote:
I think ive found a bug but need someone to double check.
Whenever I issue a reload in Asterisk, any realtime extensions stop
receiving calls.
I have to reboot the sip phones in order to get them to re-register.
Can anyone see if they have a similar
Hi,
On 09/15/2010 09:02 PM, Dan Journo wrote:
Hi,
I'm using the CallTime and a few other variables to name a recording so that
I can then take the wav file name and see when it was recorded, and what the
recording contains.
However, since ${CDR(start)} contains a space in part of the
On Wed, 15 Sep 2010, Kyle Kienapfel wrote:
On Wed, Sep 15, 2010 at 6:04 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Tue, 14 Sep 2010, Joe Freeman wrote:
Anyone have a good provider for International (US/Canada at least) 800
termination/origination? I have a
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