Hello,
I tried to send sms for local extensions and i observed that file is
created but sms isn't delivered yet. Can someone help me with this
thing?
rr:/var/spool/asterisk/sms/mttx # cat
../../outgoing/smsq.mttx.0.1322430026-20217.1
Channel: Local/1010
Callerid: SMS <1010>
Application: SMS
Data:
Hello,
I'm using asterisk with 84 extensions (aprox 45 always connected). When i
look to the opened channels i sow many channels opened without reason even i
don't have any active calls.
Is there someone else that en-counted the same problem? Is there any fix to
this bug? I have the following sett
Hello,
I have a strange situation with my asterisk 1.8.7.0 version. I compiled as
usual everything seems to be ok but from time to time when i look on my
console i get the following error message:
[Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread:
Failure to write to tcp/tl
You can use permit/deny to limit which IPs a device can register from.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S.
> Sent: Sunday, September 18, 2011 4:07 AM
> To: Ast
Hello,
Can someone help me with some tips on this?
many thanks
On Wed, Sep 14, 2011 at 5:03 PM, Catalin S. wrote:
> Hello,
>
> For the moment I have the following settings in my sip.conf. I want to
> optimize them to archive the following things:
>
> - for the moment al
Hello,
I use asterisk 1.8.6.0 and I have aprox 100 extensions. I want to lock every
extension to a single registration per device. Many of users tried to log on
my asterisk from 2, 3 devices and I want allow only one.
Is there any solution for fix this?
Thank you.
--
_
Hello,
For the moment I have the following settings in my sip.conf. I want to
optimize them to archive the following things:
- for the moment all my users will re-register too often. I want that only
lagged users to re-register quickly.
- check from time to time all users but no too often to see
udio,local_jitter)})
Thank you.
On Mon, Sep 5, 2011 at 10:58 PM, Leandro Dardini wrote:
> 2011/9/5 Catalin S.
>
>> Hello,
>>
>> I have a problem with some variables in 1.8.6.0. I set on extension the
>> following lines:
>>
>> exten => h, n, Se
Hello,
I have a problem with some variables in 1.8.6.0. I set on extension the
following lines:
exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio,
local_lostpackets)}) ; lost packets by local end **
exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio,
remote_lostpackets)})
Thank you Paul for answers. Please tell me if i upgrade to 1.8 is gtalk
module still working?
On Thu, Aug 25, 2011 at 5:42 PM, Paul Belanger wrote:
> On 11-08-25 10:34 AM, Catalin S. wrote:
>
>> Hello Paul,
>>
>> I choose 1.4.42 version because of iksemel for gt
ug 25, 2011 at 5:25 PM, Paul Belanger wrote:
> On 11-08-25 09:26 AM, Catalin S. wrote:
>
>> Hello,
>>
>> I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in
>> sip.conf at
>> [general] section the following options:
>>
>> transport=tcp
&g
Hello,
I tried but still not works. Can you make some test at your side? Something
is wrong. Thank you.
On Thu, Aug 25, 2011 at 4:35 PM, Andrew Latham wrote:
> On Thu, Aug 25, 2011 at 9:26 AM, Catalin S.
> wrote:
> > Hello,
> > I need to listen on tcp 5060 on my actual
hello,
I tried still not working. :( something is wrong.
On Thu, Aug 25, 2011 at 4:37 PM, Skyler wrote:
> Hi,
>
> On Thu, 2011-08-25 at 16:26 +0300, Catalin S. wrote:
> > Hello,
> >
> >
> > I need to listen on tcp 5060 on my actual asterisk 1.4.42. I trie
Hello,
I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in
sip.conf at
[general] section the following options:
transport=tcp
tcpenable=yes
tcpbindaddr=0.0.0.0
but after all that changes i still not see tcp port raised up. Did somebody
had the same problem and had some solution
Hello,
I wish to make a simple system to limit peers at x minutes depending
of buyer voip packet. Can someone help me with some directions?
I intend to make a separate dial plan and every calls to be in cdr
table in mysql. Is any chance to make some scripts to drop calls after
peer
used all minute
did you also hav qualify and qualifyfreq?
Thank you for reply,
On Mon, Jul 26, 2010 at 1:55 PM, Faisal Hanif wrote:
> We are having good results with
> maxexp 120
> minexp 90
> defexp 100
>
> qualify = yes
> qualify = 500
> qualifyfreq=5
> registerattempts = 0
> registertimeout = 10
> maxexpiry
Hello,
I want to optimize my registrations and calls of peers to my asterisk
with the following options in sip.conf:
---///---
qualify = yes
qualify = 500
qualifyfreq=5
registerattempts = 0
registertimeout = 10
maxexpiry = 60
minexpiry = 20
defaultexpiry = 600
---///---
Can someone more experien
mbers available for your region.
Click to go back."
Do you have some tested sites please? Thank you.
On Wed, Jul 22, 2009 at 2:41 PM, Geoff Lane wrote:
> On Wednesday, July 22, 2009, Catalin S. wrote:
>
>> I lookin' for a call in number from UK or USA. Can somebody offers
On Wed, Jul 22, 2009 at 2:41 PM, Geoff Lane wrote:
> On Wednesday, July 22, 2009, Catalin S. wrote:
>
>> I lookin' for a call in number from UK or USA. Can somebody offers
>> me a peering for this or specify any sip provider that offers this
>> thing?
>
> There
Hello,
I lookin' for a call in number from UK or USA. Can somebody offers me
a peering for this or specify any sip provider that offers this thing?
Thank you very much,
Jonson.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hello,
I have a strange situation with an SPA3102 FXO/FXS device. I'm in
situation that when i receive a call from PBX line I must forward the
calls to 2 VoIP numbers.
Right now i have the following settings: (S0<:1...@gw1>). I want to
forward at 1020 too. I tested (S0<:1010|1...@gw1>) and doesn'
is ok wait for number to be
called.
Anyway did you know how can i send dtmf after is answered?
Thank you.
On Sat, May 9, 2009 at 11:45 PM, Steve Totaro
wrote:
>
>
> On Sat, May 9, 2009 at 4:22 PM, Catalin S. wrote:
>>
>> Hello ppl,
>>
>> I want to make a special
Hello ppl,
I want to make a special dial plan for routing calls to a peer which
has an pin protection.
Normally if you want to call through that peer you must first enter
pin for example 1234#
and after that you hear the tone from line and after that you can dial
desired numbers.
I tried somethin
Hello ppl,
I have a problem with my asterisk when i want to call some destination
through my peers and I must enter DTMF digits to select some
extension/conference number or password to access some features.Every
numbers is accepted but when i must press # key my asterisk interpret
it like transfer
From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S.
> Sent: February 23, 2009 8:01 PM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] strange text message:)
>
> is any chance to use this feature t
n other soft phones like x-lite you will have the same info.
> David
>
> 2009/2/23 Catalin S.
>>
>> Hello guys,
>> I recently observed that my asterisk sends me sms like messages on my
>> phone (Nokia E71), I mean is SMS but is delivered some kind in-band
>
Hello guys,
I recently observed that my asterisk sends me sms like messages on my
phone (Nokia E71), I mean is SMS but is delivered some kind in-band
though VoIP. Is strange because this messages contains informations
about my voicemail and is sent by voicem...@mydomainxxx.com. I noticed
that this
i finally did it... It works excellent. Thank you guys for help.
On Fri, Feb 13, 2009 at 9:44 PM, David Gibbons wrote:
>
> On a similar subject, I have been able to get a 7961 to switch to a SIP
> firmware, has anyone had any luck with this?
>
>
> Yes, I have several 7961s and 7971s running SI
ex Balashov
> : --
>
>
> >
> > Have a look at:
> >
> http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080
> > 094584.shtml#topic2
> >
> > On Fri, 13 Feb 2009 12:06:48 +0200, "Catalin S."
> > wr
The SEP*
> configuration files are for SCCP.
>
> After doing that, the phone will start requesting the correct files. You
> may need to upgrade through various SIP images cumulatively.
>
> On Fri, 13 Feb 2009 11:42:03 +0200, "Catalin S."
> wrote:
>> Hello I recentl
Hello I recently get a Cisco 7940G IP Phone and I try to make several
things with it and I en counted many difficulties:
1.) I tried to unlock the phone and to set manually IP Address,
Netmask, Gateway etc. I don't get any luck.
2.) I tried to upgrade firmware like they said with tftp server... I
Hello guys, thank you for all your answers. I'll will check and i keep you
informed of what's happening next. Note that mysql and asterisk is on the
same machine
so is not a problem of connectivity or mysql machine to be down.
On Tue, Jun 24, 2008 at 3:22 PM, Tilghman Lesher <
[EMAIL PROTECTED]> w
Hello,
I configured asterisk to use mysql for CDR. Well when i check from time to
time I realize
that asterisk loose connection with mysql (i use phpmyadmin and i watch the
processes).
Can anybody tell me how can i solve that problem? I want to have all cdr
statistics logged in mysql,
is very impor
Hello,
i have installed the latest asterisk software and I user soft phones and
hard phones (generally Nokia E-Series with sip and wifi enabled functions).
I want to know how may i send in band messages to my clients. Simple text
messages on their devices/software - clients.
Thank you for any ideas
Hello did you find something? I want to do the same thing. I have asterisk
and nokia e51 phone.. Also i tried several models.
On 5/23/08, Rilawich Ango <[EMAIL PROTECTED]> wrote:
>
> Hi,
> I want to send some text to the phone such that the phone can
> display the text on its display. I have tr
Hello,
i recently installed last version of asterisk (Asterisk 1.4.18.1 built by
root @ h-gw on a i586 running Linux on 2008-03-23 00:26:44 UTC)
and everything is ok but when i call an extension i cannot hear anything. I
don't get any visible error on sip debug... i changed the codecs...
everything
Hello and thank you for reply... I tried with Playback() and is the same
effect. Is curious because sometime there's no pause other time is a long
pause.
Anybody have other idea?
Thank you.
On 12/14/07, Atis Lezdins <[EMAIL PROTECTED]> wrote:
>
> On 12/14/07, Catalin S.
Hello,
i have a simple but annoying problem. I have the following entry in
/etc/asterisk/externsions.conf file:
--
exten => 10100,1,Wait(4)
exten => 10100,2,Playback(transfer,noanswer)
exten => 10100,3,Dial(${PHONE30},30,t)
exten => 10100,4,Background(extension)
exten => 10100,5,Background(is-
Hello PPL, someone have any idea for notifying users that they have
voicemail waiting when they will register after weren't being registered on
asterisk? I need this for nokia terminal e series users. I studied sms
service but seems to be only for PSTN lines. I comes with idea to receive a
call fr
Welcome Andres, we will keep in touch:)
On 8/16/07, Andres Jimenez <[EMAIL PROTECTED]> wrote:
>
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi, all
>
> First post to a new (for me!)list. Netiquette as a must.
>
> My name is Andres Jimenez and I am an spaniard working as System
> Administ
Hello ppl,
is any set of configuration for asterisk that could put asterisk to call
users when they come back online in case they have any voicemail? I think is
a good modality to inform users that they have a voicemail and listen to it.
Thank you for you support.
_
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