Hello,
I got a SPA3102 and everything works fine except calling from voip to phone
on fxo port. The phone ring but doesn't get any sound. I connected SPA at my
asterisk server and i want to call from asterisk through SPA to fxo port
where i have a regular phone. Thank you for support.
Hello,
i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with
cdr. Well all I want is to receive incoming calls from pstn on specified sip
account (suppose 8000), and to initiate outgoing calls from all my asterisk
sip accounts through SPA3102 device. Someone can explain me
Hello,
i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with
cdr. Well all I want is to receive incoming calls from pstn on specified sip
account (suppose 8000), and to initiate outgoing calls from all my asterisk
sip accounts through SPA3102 device. Someone can explain me
How about required MTU and jitter? I think openvpn will add some latency and
frames will be charged with supplementary encapsulation bits.
On 08 May 2007 19:03:09 +0200, Benny Amorsen [EMAIL PROTECTED]
wrote:
NM == Noah Miller [EMAIL PROTECTED] writes:
NM If it helps at all, I read a study
Hello,
i just want to activate SMS service between my asterisk local sip accounts
and between asterisk and local sip accounts. How can i do this thin? Also i
tried smsq to an account but all i obtained is a error message:
---Cut Here---
May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open
Thank you for reply. Can you send me some working configs? I'm still
confusing about this sms option.
On 5/22/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
Hello,
i just want to activate SMS service between my asterisk
Can you tell me how may i do that?
On 5/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Anselm Martin Hoffmeister [EMAIL PROTECTED]
Date: Tue, 22 May 2007 13:41:43 +0200
Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
Hello,
i just want to activate SMS service between my
Was for [EMAIL PROTECTED]
On 5/23/07, Jonson Player [EMAIL PROTECTED] wrote:
Can you tell me how may i do that?
On 5/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Anselm Martin Hoffmeister [EMAIL PROTECTED]
Date: Tue, 22 May 2007 13:41:43 +0200
Am Dienstag, den 22.05.2007, 13:21 +0300
to scan
service '/var/spool/asterisk/outgoing/smsq.mttx.0.1179907051-32570.1'
---And Here---
On 5/22/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Dienstag, den 22.05.2007, 17:35 +0300 schrieb Jonson Player:
Thank you for reply. Can you send me some working configs? I'm still
confusing
Hello,
I'm wandering how can I make voicemail notification when i got a messages in
asterisk mailboxes. For the moment i have e-mail notifications, but I readed
that I can do also a sms notification to local sip accounts. Also I'm
wandering if i can make something like callback from asterisk to
Hello, I want to limit calls per sip account user. How may I realize this
setting? For example I want to limit to 10 min all possible calls from an
account or to limit external calls to 10 min and local call remain
unlimited. Thank you for support guys.
Hello,
I just want to put all my sip accounts in mysql and asterisk use it from
mysql. How can I do that, could you be more specific because I readed alot
on wiki and i'm lost... I don't know what to modify in Makefile from channel
directory. I use asterisk 1.4.4, that is already compiled and i
extensions.conf then configure extconfig to map the newly created tables.
Joss.
On 5/27/07, Jonson Player [EMAIL PROTECTED] wrote:
Hello,
I just want to put all my sip accounts in mysql and asterisk use it from
mysql. How can I do that, could you be more specific because I readed alot
on wiki and i'm
Hello I just settup a realtime mysql table for sip_peers. All peers
(friends) is autenticateing but when i want to initiate a call between them
i got the following error. Someone have some ideea? Thank you.
---Cut Here---
pbx*CLIconsole dial 1014
== Console is full duplex
-- Executing
Hello,
I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to
configure voice part on it. I cannot get it if I can use like peer for my
asterisk. Please help me with some tips.
Thank you guys.
Regards,
Jonson.
___
--Bandwidth and
=rfc2833
canreinvite=no
qualify=1000
context=incoming
port=5061
[FXS_username ]
disallow=all
allow=alaw
type=friend
username= FXS_username
secret= FXS_password
host=dynamic
dtmfmode=rfc2833
canreinvite=no
qualify=1000
context=outgoing
Best regards
Mihaela MJ
On 1/26/07, Jonson
Hello,
I'm interested too in analyzer/statistics/billing system. Can we develop
together something simple? What scripts do you recomand me?
Thank you,
Jonson.
On 2/22/07, nik600 [EMAIL PROTECTED] wrote:
I am planning to develop an open source (GPL) queue statistic/analyzer.
Can i use that
Hello,
i just installed asterisk 1.2.15. I got this error message. Somebody can
help me? Thank You.
Feb 27 11:47:43 NOTICE[17086] cdr.c: CDR simple logging enabled.
Feb 27 11:47:44 WARNING[17086] pbx.c: Already have an application 'Pickup'
Feb 27 11:47:44 WARNING[17086] loader.c:
Hello Francis,
I also hev asterisk and sipura. Can we chat online on gmail/yahoo. Let's
make some experiments... I hev the same problem like you.
On 4/12/07, Francis Augusto Medeiros [EMAIL PROTECTED] wrote:
On 10 de abr de 2007, at 23:05, James Harper wrote:
2 - How can I gain full
Hello,
i hev a subscription to a international voip provider and I want
all calls for numbers _001xx to go through my voip provider. I
tried many settings in sip.conf, extensions.conf and iax.conf. Please
give me some simple example for how can i transfer the specified calls
to my external
Hello if someone found some method to authentificate to asterisk with nokia push to talk clients please send me all your documentations and the tests results, I really need this for a project of main and i wanna dig deeper to solve this mister. Thank you guys for your cooperation. Alex i put you
Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking
about SPA3102.
What you guys think about it. Is ok, is working with asterisk, can i use it
like voip peer. Thank you for your advice.
Jonson.
___
--Bandwidth and Colocation provided by
Okay, i'll move my discuss to asterisk-users.
Thank you.
On 1/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Jan 17, 2007 at 04:39:03PM +0800, 黄宗宁 wrote:
Jonson Player wrote:
Hello, I intend to buy a FXO/FXS device from Linksys.
I'm thinking about SPA3102. What you guys thik
Hello,
I don't know if this list is appropriated to this subject but I want to ask
you if there's some list where I can make an advertising announce for a new
sip web site that was just launched.
hank you.
--
_
-- Bandwidth and
Hello,
I have strange situation with asterisk 1.8.18.0 , randomly i got this
message in cli:
WARNING[15925] asterisk.c: No more connections allowed
All connections freeze and all extensions doesn't work anymore. Is any bug
or is any setting that can solve this problem?
Thank you.
Jonson.
this analyzing script to monitor when i have
strange rise of use channels to prevent attacks or brute force.
On Tue, Nov 20, 2012 at 3:44 PM, Joshua Colp jc...@digium.com wrote:
Jonson Player wrote:
Hello,
Hola,
I have strange situation with asterisk 1.8.18.0 , randomly i got this
message
Hello Danny,
Could you tell me how can i put time out at execution of remote commands
with asterisk -rx show sip channels.
I think that is my problem... after i execute asterisk -rx commands
something remain stalled and somehow i think that could block my asterisk...
I mean all new connections
/2012 03:32 AM, Jonson Player wrote:
Hello,
I have strange situation with asterisk 1.8.18.0 , randomly i got this
message in cli:
WARNING[15925] asterisk.c: No more connections allowed
All connections freeze and all extensions doesn't work anymore. Is any
bug or is any setting
Hello,
I want to use an Huawei stick model K3765 which support voice with
asterisk. I'm begginer with this kind of interaction from asterisk
with external devices.
Can someone guide me what should i configure to use this device?
Thank you for support,
Regards,
Jonson.
---
www.Mobile-Wi.Fi
--
Hello guys,
I looking for some dial plan which can mach on +xxx numbers instead of
00xxx numbers.
Some users of main use + instead of 00 for international dial. Is there any
solution for this problem?
As far as i readed in asterisk is some kind of replacement of characters in
dial plan command.
Hello Adam,
Thank you very much for your info.
Regards,
Jonson.
On Tue, Jun 11, 2013 at 12:34 AM, ad...@3a.hu wrote:
Hi,
On 06/10/2013 22:26, Jonson Player wrote:
Some users of main use + instead of 00 for international dial. Is there
any solution for this problem?
swap the + sign
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