Hi ,
I have 3 ISDN BRI and 4 analog line .
I would like a smal ofice with 30 exension.
Can you give me it is possibile to work together isdn and analog in a same pc (PBX).
Which isdn and analog card aou recommand ? Is there any support for these card ?
Thaks.
Do you Yahoo!?
Yahoo! Small
Hello ,
I want to install a littel office . I have some question regarding to it.
My office has 8 analog line in and we would 20 line out (analog ).
So as a sow I need 2 TMD fxo card and 5 TMD txs card , Am I right ?
Can I use these card in one PC ?, or I need more PC s ?Can I install driverin one
I have a question regarding to OS platform.
As I see on Wiki -s homepage there are many type of linux version.And in some of them there are reported errors (regarding to asterisk ) for exemole in rad hat .
Can you tell me what is the best linux paltform ,( version ), which is supported by digiroom
Hello ,
I have a question regarding to PRI card (Wildcard TE110P).We want ot use this card in Hungary .So if we have a PRI line (30 B channel (64Kb) and 2 D channel) is is good (I think)
But what happends then when we have only BRI (2-D channel + 1-D channel for signaling) ?
Does it works this
Hello !
I try to run asterisk with real time config from database.
I use AMP to configure .
Everythig it ok , I cansetnew sip and iax extensions, I can see them on mysql db , as well is amp .
But these extension I cannt use in asterisk .
I have seen some new conf file
Yes I've checked . these pakeche I have instaled.
But it does not work.
echo "libxml2"rpm -qa|grep libxml2echo "libtiff"rpm -qa|grep libtiffecho "libtiff-devel"rpm -qa|grep
Yes I've checked . these pakeche I have instaled.
But it does not work.
echo "libxml2"rpm -qa|grep libxml2echo "libtiff"rpm -qa|grep libtiffecho "libtiff-devel"rpm -qa|grep
Hi
I confiured Gasnstream phone 100. Firmware ver:Program--1.0.5.16 Bootloader--1.0.0.21 HTML--1.0.0.41 ïVOC--1.0.0.7.
It workes well everything. If I got a message it blinks. My voicemail no 555 .If I call 555,I can hear voicemail . But I can not configure Message Button on the phone. I set
Hey,
Implementing Asterisk on local Lan spread over 2 campuses on two different
cities is our graduation project.
Having done all the research and reading stuff. I started with the practical
work.
Not getting a hand on the linux digium. I installed Red Hat linux 9.0. I was
able
Note: forwarded message attached.
Send instant messages to your online friends http://uk.messenger.yahoo.com ---BeginMessage---
Hey,
I am new to asterisk and softphones. I am able to install astersik and 2 XLite
softphones on three PCs with linux feora core 6. I have also written a basic
Hey,
I am a new to asterisk and softphones. Ihave recently
installed and configured linux and 2 xlite clients all
in linux fedora core 6. I have also made a dial plan
for the two users. But when i dial from one xlite
client to another i can hear the ring tone but when i
answer the call i can not
Hey!
I am trying to configure the voice onference with
MeetMe application for my internal users. I have my
server and 4 clients on same LAN and following is my
extensions.conf file:
[globals]
Ahsen=SIP/222
Tahami=SIP/444
Uzair=SIP/333
Wasif=SIP/555
[internal]
exten =
do.
--- Paul Hales [EMAIL PROTECTED] wrote:
Sure, but you will probably have to recompile
Asterisk to get all the
extra bits.
Should only take you 10 minutes.
later,
PaulH
On Mon, 2007-03-12 at 06:54 +, Asterisk Asterisk
wrote:
Hey! Thanks you are absolutely rite could i
I need your help: please help test the gender detection module at 575-613-4392.
I wrote a gender detection module and thought I'd try it out. It only takes a
second. I've been showing 90%+ accuracy and I want
to make sure it's working correctly. Rain and significant background noise
seems to
nt_jnewman at yahoo.com
From: Ron Joffe ron.jo...@gmail.com
To: asterisk-users@lists.digium.com
Cc: Asterisk Asterisk nt_aster...@yahoo.com
Sent: Monday, February 16, 2009 11:05:24 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613
That's funny. The way I have it phrased, when I called I started talking to it
as well! I have some code for short list voice recognition and thought about
detecting yes and no in there, but I ran out of time...and the prompts were
already recorded.
Thank you everyone for helping test the
found out that the best solution is to use OpenSips as SIP
OpenSIPS is a great free software proxy.
1- Is there any Software limitation on asterisk regarding number of
simulltaneous calls?
There isn't any explicit limitation in Asterisk or OpenSIPS that I'm aware of,
but you are limited to
?
From: Asterisk Asterisk nt_aster...@yahoo.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: Gondar Monn gonda...@gmail.com; nt_aster...@yahoo.com;
nt_jnew...@yahoo.com
Sent: Tuesday, February 17, 2009 9:10:38 AM
Subject: Re
Accuracy should be 10%-15% better on Wed or Thu.
From: Jason Aarons (US) jason.aar...@us.didata.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 17, 2009 10:48:07 AM
Subject: Re:
I will be releasing updated versions to many of the detection modules next
week. They include better support of Asterisk 1.2, 1.4, and 1.6, better
detection, better parameters, an easier build system, and usability is enhanced.
The updated modules include:
* FaxDetect, LineDetect, and
...@yahoo.com
Sent: Feb 18, 2009 4:09 AM
Subject: Re: [asterisk-users] Please help test the gender detection moduleat
575-613-4392
Am Montag, den 16.02.2009, 11:45 -0800 schrieb Asterisk Asterisk:
Let me know how it works when you try the test number at 575-613-4392.
Hi Justin,
I tried your
On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro stot...@totarotechnologies.com
wrote:
On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk nt_aster...@yahoo.com
wrote:
This module detects gender and approximate age range. I'm working on getting
it's accuracy to 80%+ on a consistent basis, after
from being able to do speaker verification? Not
*identification* mind you, but being able to tell that a captured voice is
the same as another that is stored...
Cheers,
j
On Wed, 18 Feb 2009, Asterisk Asterisk wrote:
Steve,
Tried to test and got call could not be completed as dialed
Don't most?
From: Nhadie nha...@gmail.com
To: Asterisk-users@lists.digium.com
Sent: Wednesday, February 18, 2009 6:19:24 AM
Subject: [asterisk-users] US DID
Hi,
Anyone knows a DID provider that can do both outbound and inbound?
Regards
Nhadie
You can probably use combo of NVLineDetect, NVGenderDetect, and AMD
(NVMachineDetect).
From: Edwin Quijada listas_quij...@hotmail.com
To: Asterisk Asterisk asterisk-users@lists.digium.com
Sent: Thursday, February 19, 2009 12:55:05 PM
Subject: Re: [asterisk
You might also check with www.star2star.com (Star2Star Communications). We did
a call park, pickup, and transfer module with similar functionality. Integrates
very nicely.
Justin Newman
nt_jnewman at yahoo.com
From: Jeff LaCoursiere j...@jeff.net
To:
We have a BLF module that maintains device state across Asterisk servers.
Contact me off the list if interested.
Justin Newman
nt_jnewman at yahoo.com
From: Lenz Emilitri lenz.lo...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
@lists.digium.com
Sent: Wednesday, February 18, 2009 4:13:46 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
Pretty cool. I'm almost offended though as I'm not usually guessed as a
female of the species. :)
Darren Wiebe
dar...@aleph-com.net
Asterisk
That's interesting - I haven't noticed this with any of my installs. What
version of firmware and SIP?
From: Barry D. Hassler barry.hass...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday,
as another that is stored...
Cheers,
j
On Wed, 18 Feb 2009, Asterisk Asterisk wrote:
Steve,
Tried to test and got call could not be completed as dialed.
Were you able to connect? If not, please try again. Call volume has been
growing.
How about a moving stress variable that could
-users] Please help test the gender detection module at
575-613-4392
It got my gender correct the two times I tested, even with the TV loud in the
background.
BTW, I love the beep.
On Thu, Feb 19, 2009 at 5:54 PM, Asterisk Asterisk nt_aster...@yahoo.com
wrote:
You sure you don't have a pony
-users] check if not human
NVLineDetect, NVGenderDetect what is that?
amd info voip-info.org or asterisk.org support asterisk book.
i bougth one to support the cause!!!
David
2009/2/19 Asterisk Asterisk nt_aster...@yahoo.com
You can probably use combo of NVLineDetect, NVGenderDetect, and AMD
We've had a 65% success rate across the board (actually 35% incorrect). I'm
working on bringing that up to 85% or better.
From: Ira i...@extrasensory.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
if it correctly identified you?
Cheers,
j
On Fri, 20 Feb 2009, Asterisk Asterisk wrote:
We've had a 65% success rate across the board (actually 35% incorrect). I'm
working on bringing that up to 85% or better.
From: Ira i...@extrasensory.com
To: Asterisk
...@xorcom.com
To: asterisk-users@lists.digium.com
Sent: Friday, February 20, 2009 10:48:10 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
Slightly off-topic,
On Mon, Feb 16, 2009 at 10:29:57AM -0800, Asterisk Asterisk wrote:
I need your help
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 20, 2009 7:27:12 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
Hi,
Asterisk Asterisk wrote:
We've had a 65% success rate across
Tzafrir- Again, very good points. See my responses below.
What does this have to do with using 'Asterisk Asterisk' instead of
'Justin Newman'?
I was sick of all the junk mail in my old accounts and working with digest mode
was a pain, so I quickly created a new yahoo.com account a few days ago
help test the gender detection module at
575-613-4392
Asterisk Asterisk wrote:
You have some good points.
Justin Newman isn't exactly someone we don't know. However I only
I agree that my name wasn't clear, but I was trying to avoid getting a
bunch of spam myself. I'm not sure if I've
Hello,
I'm having a problem with the autoattendant. It won't recognize the
DTMF signals from certain people that call in. I have relaxed DTMF,
upgraded Asterisk from 1.2 to 1.2.12 to 1.2.12.1 as well as the zaptel
drivers. I have stopped X from running then only thing I didn't do
that was on
I did turn it on and off as it does not seem to make a difference. On 9/19/06, Jay R. Ashworth [EMAIL PROTECTED]
wrote:On Tue, Sep 19, 2006 at 05:29:17PM -0400, asterisk asterisk wrote: I'm having a problem with the autoattendant. It won't recognize the
DTMF signals from certainpeople that call
This is my features.conf
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
parkingtime = 45 ; Number of seconds a call can be parked for
;
Check your extensions.conf on the context setted on zapata.conf
probably you have the command answer you should remove it.
On 8/16/05, Hubert Hoefsloot [EMAIL PROTECTED] wrote:
This must be a question asked before but can't find it so here I go:
I have a Asterisk box connected, thou a x100p,
I have a very simple question .
I have 2 internal extension 301 and 300 sip phone . I want to these extesioncan call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal.
How can I do that ?
[x1]exten = 300,1,Dial(SIP/300)
include = pstnlocal
[x2]exten =
of group in wich I am (x1 or x2).
Ronald Wiplinger [EMAIL PROTECTED] wrote:
asterisk asterisk wrote: I have a very simple question . I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal
Hi,
Zoiper is a great software to have both SIP and IAX. As a beginner to
Asterisk, I find very well but to my understanding it does not have linux
version.
X-lite have both Windows and Linux but it is a bit clumsy to set up.
CK
On Fri, Jul 23, 2010 at 5:04 AM, Ronaldo Zacarias Afonso
Hi,
I have problem in initiating an dial out call with SIP response 500 Server
Internal Error
The sip debug as
== Using SIP RTP CoS mark 5
Audio is at 113.253.226.92 port 18284
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
*Subject:* Re: [asterisk-users] SIP response 500 Server Internal Error
Hi,
I have problem in initiating an dial out call with SIP response 500
Server Internal Error
The sip debug as
snip
--- SIP read from UDP
I fix the problem now because of the outbound CID issues.
On Tue, Aug 10, 2010 at 6:14 AM, asterisk asterisk aster...@ck-lee.comwrote:
I try to disable firewall but no working. I use a softphone to connect on
the same lan segment, it works. Dial in is no problem but dial out always
have
Hi,
I have an interesting problem that the dial out via sip always generates 603
error
The following is the sip debug
Your help is appreciated.
CK
== Using SIP RTP CoS mark 5
-- Executing [998560...@dlpn_dp1:1] Dial(SIP/6100-005b,
SIP/13398560...@hkbn2b) in new stack
== Using SIP
Hi,
I have an interesting problem that the dial out via sip always generates 603
error
The following is the sip debug
Your help is appreciated.
CK
== Using SIP RTP CoS mark 5
-- Executing [998560...@dlpn_dp1:1] Dial(SIP/6100-005b,
SIP/13398560...@hkbn2b) in new stack
== Using SIP
Could you share your AGI script?
CK
On Wed, Aug 18, 2010 at 5:43 AM, Johann Hoehn johann.ho...@ecommerce.comwrote:
On 08/17/2010 09:00 AM, Tino wrote:
Hello,
I would like to send sms to some external phone numbers from my
asterisk server. Is it possible to send sms via softphones like
Olivier,
You should find out the SMS tab in the handset but not in the web service.
Did you IP pone work?
CK
On Tue, Sep 14, 2010 at 2:27 PM, Olivier oza_4...@yahoo.fr wrote:
Hi,
With my Gigaset C470IP (with latest 02223 firmware), I can't find a way to
access SMS settings from web
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
Apart from that, any other tricks that I can manipulate within asterisk.
??sip.conf parameter or other??
On Thu, Sep 16, 2010 at 12:07 AM, Luki lugos...@gmail.com wrote:
I am not sure about the problem but note that it may be related to
incorrect
IP being used. Sometimes, WAN 1 and
Hi,
I have some problem with my provider. While the sip registration is
successful, i intermittently encounter problem in dialing out. I receive 603
Declined error in my Sjphone client. The asterisk log shows line is
busy/congestion.
Appreciate if help or direction can be provided.
Thanks.
CK
'
67f6129e02db3377276c62f209913...@sip.etransmed.net' Method: OPTIONS
On Thu, Oct 14, 2010 at 7:55 AM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Wed, Oct 13, 2010 at 6:48 PM, asterisk asterisk aster...@ck-lee.com
wrote:
Appreciate if help or direction can be provided.
21.6.2 603 Decline
The callee's
Put the outboundproxy=192.0.2.1 under individual sip context not under the
[general], it should work.
CK
On Mon, Oct 25, 2010 at 11:43 PM, sipbeast sipbe...@gmail.com wrote:
Hi,
My asterisk's version is 1.6.0.26. I've couple sip providers and I've
for new SIP provider I need define
I have been using rpm version of asterisk 1.6. However, I notice the support
for gtalk is absent from rpm. I tried to compile source code and then moved
to the /usr/lib/asterisk/modules. But the modules cannot be loaded.
Anyone has successful experience.
Mine is using 1.6.2.12.
I also tried in
Hi,
I encounter this problem recently after quite some months of my asterisk.
I have a SIP trunk for dial in and out.
When dial-in, it matches the callerid number and decides. If matched, it
will either go into an a very brief IVR. The IVR allows caller to dial
internal extension.
All along it
:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
*Sent:* Wednesday, February 16, 2011 5:39 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] DTMF not detected, time out
Hi,
I
:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
*Sent:* Wednesday, February 16, 2011 8:58 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] DTMF not detected, time out
I have a sip trunk connecting to a huawei softx3000. At the moment, I can
register and dial in.
However, peer status shows not reachable
sip show peer as follow
* Name : cmphone
Secret : Set
MD5Secret: Not set
Remote Secret: Not set
Context : from-cmphone
HI,
My understanding is that the modem won't work. I believe asterisk does not
support.
I wonder why you do not have the built in ethernet in your motherboard. You
can spare your PCI slot for a proper FXO card and use USB-to-ethernet
For a PCI FXO card, the cheapest will be X100 but be aware of
I totally agreed with Leif Madsen that viable options are available and time
and effort spent on winmodem should be carefully considered.
My system also works with an ATA as PSTN gateway and VOIP SIP provider for
DID and inbound/outbound service. It will save time much more time and
effort while
Hi,
I am using IAXmodem + hylafax to do outgoing and incoming fax with asterisk.
I wonder how to write a dialplan to differentiate incoming call or fax.
I am sharing a line for both voice and fax.
CK
--
_
-- Bandwidth and
Siemens IP A580 works fairly well.
2011/3/9 Sébastien BERGER sebast...@ab2l.eu
My personal experience :
Corded : Snom 3xx and 8xx, Aastra 6731i, 6755i and 6757i and Polycom IP330,
IP650.
DECT : Siemens C470, Polycom Kirk KWS300 and 600v3
Work well
AB2L
+33 (0)367100783
${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
I use the above command to get the system date and time
it returns 20110321-034329
but it is exactly 8 hours early than the system time when I type date in
linux terminal
Mon Mar 21 19:43:35 HKT 2011
I am looking for help.
CK
--
With gmt+8, the result is
-Mon Mar 21 13:47:59 2011
For linux server timezone I set it via webmin and /etc/localtime is my
timezone file i.e. HK at GMT+8
On Mon, Mar 21, 2011 at 9:36 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, Mar 21, 2011 at 09:23:29PM +0800,
Thanks,
You give me the right answer.
On Mon, Mar 21, 2011 at 10:19 PM, Barry Miller
asterisk-us...@notanet.netwrote:
On Mon, Mar 21, 2011 at 07:45:37PM +0800, asterisk asterisk wrote:
${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
I use the above command to get the system date and time
You need a separate Huawei USB stick to do the connection with asterisk.
Your K3765 should work with asterisk via chan_datacard.
http://wiki.e1550.mobi/doku.php?id=requirements
I have just made my K3715 works very well with asterisk.
CK
On Fri, Apr 1, 2011 at 5:45 AM, Alejandro Kauffmann
same here. Something seriously wrong after upgrade
Don't upgrade now.
On Sun, Apr 10, 2011 at 9:32 PM, Frank Tarczynski ft...@mindspring.comwrote:
My AsteriskNow box was updated to Centos 5.6 (2.6.18-238.5.1.el5) and DAHDI
doesn't want to load. I've tried building it from the sources, but get
-dogan.net wrote:
Hello asterisk asterisk,
Am 2011-04-01 06:41:46, hacktest Du folgendes herunter:
You need a separate Huawei USB stick to do the connection with asterisk.
Your K3765 should work with asterisk via chan_datacard.
I have this now installed in my Kernel and tried to configure
Huawei e180, K3715 are good to play around. Both voice and SMS are
supported.
On Fri, Apr 29, 2011 at 2:47 AM, Tiago Geada tiago.ge...@gmail.com wrote:
I used succesfully huawei E1550
On 24 April 2011 16:45, Dovid Bender asteriskus...@dovid.net wrote:
Hi List,
I am looking to play
Hi,
I am looking for tutorial to generate a callfile so that after my program
executes, a callfile is generated and pass to asterisk to send to the
recipient.
Any suggestion?
Besides, do you know if there is a web-based GUI to send sms via asterisk?
Thanks.
CK
--
Hi,
I am a question to handle incoming goggle voice. I have put several GV
accounts into the jabber.conf. How can I direct different accounts to
different extensions?
Help with example is much appreciate
Thanks,
CK
--
_
--
Hi,
I am looking for a simple solution to do this.
I wish to have the user to enter their preferred method of connection i.e.
for the cheapest solution to their desktop phone or mobile phone, then plan
callfile based on the number that user provided and dial to the user.
Any suggestions?
CK
--
Of asterisk asterisk
Sent: Wednesday, June 15, 2011 11:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Goggle voice incoming dialplan
Hi,
I am a question to handle incoming goggle voice. I have put several GV
accounts into the jabber.conf. How can
Do anyone know how to receiving incoming call from GV number associated with
an non gmail.com account? I have custom domains under google and would like
to receiving calls via asterisk.
The google chat function is missing in these GV accounts.
On Thu, Jun 16, 2011 at 11:30 AM, asterisk asterisk
to asterisk.
** **
Otherwise you will need to get a free SIP Account, and route calls to it.*
***
** **
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
*Sent:* Thursday, June 16, 2011 11:39
Thanks. Will need some time to look into.
On Thu, Jun 16, 2011 at 3:56 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Jun 16, 2011 at 11:26:21AM +0800, asterisk asterisk wrote:
Hi,
I am looking for a simple solution to do this.
I wish to have the user to enter
Could you elaborate on how you can associate those non-gmail accounts with
gchat account?
On Fri, Jun 17, 2011 at 2:38 PM, Warren Selby wcse...@selbytech.com wrote:
On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk
aster...@ck-lee.comwrote:
Can this non gmail.com GV number be terminated
I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it
When using GUI to access, I got this error
*** glibc detected *** /usr/sbin/asterisk: double free or corruption
(!prev): 0x0919c070 ***
The server cannot be connected via GUI and the asterisk CLI dropped and exit
into linux
Hi,
I did not find any file with a or i with your suggested commands.
Any other clues?
CK
On Fri, Jul 1, 2011 at 6:23 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On Friday 01 Jul 2011, asterisk asterisk wrote:
I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it
I am encountering problem recently with the chan_mobile that the bluetooth
connection between the asterisk and my Nokia E71 mobile phone frequently
connect and disconnect within seconds. As a result, I can't make any call
using Mobile/E71/{exten:2}.
Any suggested cause?
--
From time to time, I have a DTMF problem. The asterisk cannot recognize my
handset key pressed if I press 9 to start with. However, if I press with 6,
it is ok.
On the other hand, if DMTF key is generated from softphone, it is ok.
My dialplan is as follow
exten = 1002,1,Answer
exten =
I have naive question. I do not have any hardware on my asterisk host. All I
have are either SIP trunk for DID or hardware ATA which bridges the asterisk
to PSTN. Do I need Dahdi install? Do i have ztdummy for timing issue? I
encounter problem in this when I try to install Dahdi latest but I found
Hi,
I used to use Zoiper IAX to connect to my asterisk server from remote site.
On asterisk CLI, I can see my zoiper client registered and stay on line.
HOwever, I don't know why now I can't call this client. It always show up as
Unable to create channel IAX2 (Cause 20 Unknown)
I am using
I can tell you that siptosis is workable but the support has been dropped
recently as well.
It is a great program and especially the paid version with trunk builder
i.e. you can have multiple skype instances
On Wed, Nov 16, 2011 at 8:01 PM, Abdul Basit basit.e...@gmail.com wrote:
Any has Skype
The only inexpensive way is to get siptosis but the developer has stopped
the support and upgrade unfortunately. I have been using it for two years
or more.
Excellent quality and works very well
On Sat, Oct 13, 2012 at 5:17 AM, Philip Bennefall phi...@blastbay.comwrote:
From what I gather, it
Dear all,
I wish to ask a question of the new Motif Channel in asterisk 11.
I successfully compile the binary and run without error. However, when
dialing out, no external connection only ringing.
Any suggestions?
I follow the set up in wiki
CK
--
I experience random crash of machine (full hang, requiring a hard reset)
after trying to test run Asterisk 11.
The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled
from the source and no other software has been installed
Anyone experience similar situation?
--
...@dotr.comwrote:
are you running dahdi ?
We're using 11, System uptime: 3 weeks, 22 hours, 42 minutes, 19
seconds, 231452 calls processed
We did, however, have a problem with dahdi freezing the machine
Julian
On 7 November 2012 22:32, asterisk asterisk aster...@ck-lee.com wrote:
I experience random
I wish to ask if there is way to keep IAX trunk connection up. I have a
small server on Xen VPS but notice that my IAX trunk drops after some time.
I understand there is cron job to function as sip watchdog.
My asterisk is 11.0.1
Thanks for suggestions.
CK
--
Try to use firefox instead of IE. Besides, you may check if there is any
problem in the extensions.conf. My recent experiment of installing gui into
asterisk 11.x is that there is problem in some of the macro script within
extensions.conf.
I delete the sample macro scripts in extensions.conf and
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