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 thr
the module.
Justin
nt_jnewman at yahoo.com
From: Ron Joffe
To: asterisk-users@lists.digium.com
Cc: Asterisk Asterisk
Sent: Monday, February 16, 2009 11:05:24 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at
575-613-4392
That
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 modul
>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
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Gondar Monn ; nt_aster...@yahoo.com;
nt_jnew...@yahoo.com
Sent: Tuesday, February 17, 2009 9:10:38 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at
Accuracy should be 10%-15% better on Wed or Thu.
From: Jason Aarons (US)
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, February 17, 2009 10:48:07 AM
Subject: Re: [asterisk-users] Please help test the gender detection moduleat
575
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 MachineDe
Martin Hoffmeister
To: nt_jnew...@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 575-61
, Steve Totaro
wrote:
On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk
wrote:
This module detects gender and approximate age range. I'm working on getting
it's accuracy to 80%+ on a consistent basis, after implementing filters to
remove background noise and other artifacts.
It&#
ification? 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".
>
> Were you able
Don't most?
From: Nhadie
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
To: Asterisk Asterisk
Sent: Thursday, February 19, 2009 12:55:05 PM
Subject: Re: [asterisk-users] check if not human
How can I detect how many ring
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
To: Asterisk Users Mailin
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
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, February 1
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 Asterisk wrote:
> Steve,
>
>
That's interesting - I haven't noticed this with any of my installs. What
version of firmware and SIP?
From: Barry D. Hassler
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, February 20, 2009 8:41:33 AM
Subject: [asterisk-users] Pol
rs,
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 variab
ion 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
wrote:
You sure you don't have a pony tail? :) Hehe.
It happens to the best of us. Hopefully
, 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
You can probably use combo of NVLineDetect, NVGenderDetect, and AMD
(NVMachineDetect).
From: Edwin
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
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, February 19, 2009 5:18:07 PM
Subject: Re: [asteris
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
> To: Asterisk Users Mailing L
you
have any thoughts, feedback, or suggestions.
From: Tzafrir Cohen
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,
__
From: Steve Underwood
To: Asterisk Users Mailing List - Non-Commercial Discussion
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
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 ya
5-613-4392
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
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 --- Begin Message ---
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 => 1234,1,Macro(voicemail,${
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
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 driver in o
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 car
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.
How can I do that ?
[x1]exten => 300,1,Dial(SIP/300)
include => pstnlocal
[x2]exten => 301
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, >
Hello !
I try to run asterisk with real time config from database.
I use AMP to configure .
Everythig it ok , I can set new 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
;sip_additional.conf;iax_
Yes I've checked . these pakeche I have instaled.
But it does not work.
echo "libxml2" rpm -qa|grep libxml2 echo "libtiff" rpm -qa|grep libtiff echo "libtiff-devel" rpm -qa|grep libtiff
Yes I've checked . these pakeche I have instaled.
But it does not work.
echo "libxml2" rpm -qa|grep libxml2 echo "libtiff" rpm -qa|grep libtiff echo "libtiff-devel" rpm -qa|grep libti
Hi
I confiured Gasnstream phone 100. Firmware ver:Program--1.0.5.16 Bootloader--1.0.0.21 HTML--1.0.0.41 ï VOC--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 phon
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 Busi
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
; (d
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 x10
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 Dig
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 fro
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
--
_
-- Band
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
--
: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf 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
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
ted directly 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.co
Thanks. Will need some time to look into.
On Thu, Jun 16, 2011 at 3:56 PM, Tzafrir Cohen wrote:
> 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 t
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 wrote:
> On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk
> wrote:
>
>> Can this non gmail.com GV number be terminated at some sip accoun
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 comma
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 wrote:
> 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
> >
> > Whe
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 => 1002,n,
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 Asteri
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 wrote:
> Any has Skype For Asterisk (SFA)
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 wrote:
> From what I gather, it costs extra for eac
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?
--
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 wrote:
> > I
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
--
___
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 is
m:* 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 detect
; *From:* 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
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 :
MD5Secret:
Remote Secret:
Context : from-cmphone
Subscr.Cont. : device-hi
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 k
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 Colocat
Siemens IP A580 works fairly well.
2011/3/9 Sébastien BERGER
> 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
> sebast...@ab2l.eu
>
>
>
${STRFTIME(${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 wrote:
> On Mon, Mar 21, 2011 at 09:23:29PM +0800, aster...@ck-lee.com wrote:
> > The timezon
Thanks,
You give me the right answer.
On Mon, Mar 21, 2011 at 10:19 PM, Barry Miller
wrote:
> 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 ti
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 wro
same here. Something seriously wrong after upgrade
Don't upgrade now.
On Sun, Apr 10, 2011 at 9:32 PM, Frank Tarczynski wrote:
> 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 this
> error messa
may-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 in
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 wrote:
> I used succesfully huawei E1550
>
> On 24 April 2011 16:45, Dovid Bender wrote:
>
>> Hi List,
>>
>> I am looking to "play around" with chan_datacard. Any advice
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,
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 <
ronaldo
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 203.8
terisk-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"
I fix the problem now because of the outbound CID issues.
On Tue, Aug 10, 2010 at 6:14 AM, asterisk asterisk wrote:
> 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 this error
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 S
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 SI
Could you share your AGI script?
CK
On Wed, Aug 18, 2010 at 5:43 AM, Johann Hoehn wrote:
> 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 X-Lite
> > ? . Any
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 wrote:
> Hi,
>
> With my Gigaset C470IP (with latest 02223 firmware), I can't find a way to
> access SMS settings from web configuration app or
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 wrote:
> On Wed, Sep 15, 2010 at 1:43 PM, Olivier wrote:
>
>>
>> On the S675IP SMS is here:
>>>
>>> Messaging -> SMS -> Settings
>>>
>>
>> No SMS entry is showing on Settin
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 wrote:
> > 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
>
> Mo
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
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8 headers 0 lines) ---
Really destroying SIP dialog '
67f6129e02db3377276c62f209913...@sip.etransmed.net' Method: OPTIONS
On Thu, Oct 14, 2010 at 7:55 AM, Paul Belanger wrote:
> On Wed, Oct 13, 2010 at 6:48 PM, asterisk asterisk
> wrote:
> > Appreciate if help or direct
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 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 outbound proxy. Ev
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 a
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 u
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