Hi,
I was wondering if anyone you got asterisk to work with Fax via google voice
? If so, can you please send me extension.conf and sip.conf, jabber.conf
and gtalk.conf settings used. I would prefer faxing with Fax for Asterisk
(FFA) via .call file. I see post where people got it work with
on caller side and reciever side i am using X-Lite phones.
Any suggestions??
Thanks,
Sharath Chandra
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Hi all,
I did not get this error in Asterisk 1.2.5 release. I am testing on Asterisk SVN-trunk-r15187 to avail the PARKEDAT variable.
- I park the call using ParkAndAnnounce
- plays moh.
- accept the call using ParkedCall
The following errors are coming on the console and there is oneway audio
I am getting the following warnings on the Asterisk when i try a call parking scenario. I use Ciso 7920 phones and Cisco2800
Executing ParkedCall(SIP/192.168.50.2-088cde00, 366) in new stack -- Channel SIP/192.168.50.2-088cde00 connected to parked call 366Mar 28 17:07:36 WARNING[10027]:
I am using ParkAndAnnounce to Park the call and explicitly retrieving using ParkedCall app in the dial plan. I am trying to guess the parking lot being used in a particular call by incrementing a counter just before the ParkAndAnnounce and decrement the counter just before the ParkedCall. I am not
with the out call or transfered to voicemail.
Thanks,
Sharath Chandra
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= res_config_mysql.so to
modules.confMojSharath Chandra wrote: Hi all, I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both
the software but when i start Asterisk, it exits with the following dump. Error Text Start
load_modules: Loading module res_config_mysql.so failed!End===
Can someone suggest a solution.
Regards,
Sharath Chandra
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http://www.paesys.com/en/index.htm for the english versionOn 10/01/06, Guillaume de Lafontaine
[EMAIL PROTECTED] wrote:HiI just discovered an interesting product line. Not tested yet...
http://www.paesys.com/fr/lecteurs_VoIP_WiVoip_VideoIP_GSM.htmIn french, sorry...Any feedback
no
problem.
hope it helps
thanks
Chandra Mistry
On 21/11/05, cp [EMAIL PROTECTED] wrote:
Does anyone know is the zyxel p2000w has call waiting? I hear noise when a
second call comes in but cannot find any documentation.
Thanks,
Chip
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I installed Asterisk CVS-NHEAD-05/13/05-01:59:30 and placed few call
in and through successfully. I was trying to set up the Realtime -
picking the sip.conf and extensions.conf from mysql. I was going
through some wiki pages, but what i don't understand is - which
configuration change makes
Hi,
I am in the process of setting up an Asterisk-based PBX at work. I get
the concept of how Asterisk works pretty decently. I am more confused
about the proliferation of TLAs like FXO, FXS, TDP, SIP,
After some intense reading I have come to some understanding of the
hardware I need to
Hi,
Has anyone used Cisco 2800 Integrated services router to intiate SIP
call to Asterisk. I would like to use it as gateway on to which T1
terminates and make Asterisk as my session target for few lines.
Please let me know if there are any issues.
Thanks,
Sharath
Hi,
I am new to Asterisk. Can i use Asterisk as a session target from
softswitch/Call Agent. I mean, is it possible to initiate a SIP call
to Asterisk. My PRI terminates onto Cisco 2800 and i want to send few
numbers to Asterisk to do some application related call control.
Please advice
in the sip.conf user definition
(not in [general]):
disallow=all
allow=ulaw
allow=alaw
and did the same. still didn't work.
what can be done if my nat is actually blocking the udp packets??
chandra
- Original Message -
From: SW [EMAIL PROTECTED
hi i am not talking about * behind NAT. its * outside NAT and GS inside NAT.
chandra
- Original Message -
From: Steve [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 9:50 AM
Subject: Re: [Asterisk-Users] grandstream asterisk configuration
On Wednesday 14
I have been really trying to solve the this problem. Has anyone had a
success on this one? I have asterisk setup outside my NAT with public IP and
I am trying to establish a connection from Budgetone behind NAT with private
IP. Everything seems to be working fine. They are registered, call rings
have u had any luck with this?
cm
- Original Message -
From: Owen Kelso [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 12, 2004 9:51 AM
Subject: Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)
Thanks for all of your responses.
I confirmed that the phone works
; Disallow all
codecsallow=3Dulaw
; Allow codecs in order of preference
dtmfmode=3Dinfo
[grandstream1]type=3Dfriendhost=3Ddynamicsecret=3Dmysecretcontext=3Doutgoingnat=3Dyesreinvite=3Dnocanreinvite=3Dnoqualify=3D2000
has anyone done this before? chandra
me.
chandra
- Original Message -
From: bam [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 6:42 PM
Subject: Re: [Asterisk-Users] grandstream asterisk configuration
Make sure that udp packets can get from the server back to the
grandstream.
At 12:40 14/01/04
same here... with nufone too... i was just getting everyone is busy at the
moment message in CLI... it was working fine before..
was it them or was something wrong with my network? will check tomm.
cm
- Original Message -
From: Burak Balasaygun [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
try this...
http://www.fnords.org/~eric/asterisk/
cm
- Original Message -
From: Ing Isianto Istiadi [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 12, 2004 7:50 AM
Subject: [Asterisk-Users] sip and x-lite
Dear all,
Can you give me the configurations for x-lite
can u give me the configuration for the firewall?? with the same
configuration i can't even talk or hear... its giving me the RTP Read Error
whenever one picks up the phone.
cm
- Original Message -
From: Steve [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 12, 2004 6:09
i also had the same problem temporarily i solved my problem with both
outside NAT. u can also do it if both inside NAT. * outside NAT and
Budgetone behind NAT simply doesn't seem to work. if u ever solve this
problem please let me know too.
thanks
cm
- Original Message -
From: Owen
i guess the user * looks for is the text within []
so i suppose the username and the text within [] should be same. try putting
[nick] in place of [person]. actually, u don't need username as it only looks
for the text between [] as username.
cm
- Original Message -
From:
- Original Message -
From: Chandra [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 08, 2004 10:12 AM
Subject: GrandStream giving an RTP Read Error
hello,
I have the following configuration now
Asterisk(publicIP)-Netgear-Sip(privateIP)
everything is registered
host=dynamic
secret=grandstream2
context=outgoing
nat=yes
reinvite=no
canreinvite=no
qualify=60
and in my grandstream config i have dtmfmode=rtp2833. same for the sip.conf.
can anyone help me out solve this problem?
Thanks
Chandra
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/asterisk
directory... (asterisk source diretory.)
thanks,
chandra
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: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 9:57 AM
Subject: RE: [Asterisk-Users] no results.
have you set up the db schema? and have you entered any sip data into the
db?
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 11:31 PM
To: [EMAIL PROTECTED]
Cc:
Subject: Re: [Asterisk-Users] no results.
there are 4 fields, id, keyword,data, flags..
i really don't know what to put in keyword and data... but i have
something
INTO `sip` (`id`, `keyword`, `data`, `flags`) VALUES ('1234',
'secret', 'blah', '0');
and so on. the 'flags' column allows you to disable an entry
without deleting the entry completely. Hope this helps!
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6
xlite saying login timed out. contact network admin.
how to get rid of this. * is not behind NAT.
DIAX works fine
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xlite saying login timed out. contact network admin.
how to get rid of this. * is not behind NAT.
also, the grandstream SIP phone also seems to fail to register. IAX phones
are all ok.
DIAX works fine
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[EMAIL
1110
[EMAIL PROTECTED] root]#
my grandstream is also not registering to *.
- Original Message -
From: CW_ASN [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 02, 2004 9:14 PM
Subject: Re: [Asterisk-Users] Call recording
- Original Message -
From: Chandra [EMAIL
hi,
i can't seem to register my grandstream SIP to * server...
i have my grandstream IP as 192.168.0.11 want to register to * at
202.51.xx.xxx.
sip show peers says that my grand stream has unspecified IP but when i try
to dial a number it gets this error...
WARNING[5126]: File chan_sip.c, Line
hi,
I am trying to install ASTGUICLIENT and when i run the
AST_WINphoneAPP_0.8.pl it opens a window VICI Phone App -0.8 but i am
getting SQL Updater Down Mesasge. How can i solve this?
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### connect to asterisk manager through telnet
$t = new Net::Telnet (Port = 5038,
Prompt = '/.*[\$%#] $/',
Output_record_separator = '',);
#$fh = $t-dump_log(./telnet_log.txt); # uncomment for telnet log
$t-open($server_ip);
i got error in this line $t-open($server_ip);
my ip is
things easier. How has NuFone or Voicepulse or IaxTel guys have implemented
their asterisk box to add friends or peers?
Suggest.
Chandra
is there a installation guide? i didn't find any.
just the readme file.
- Original Message -
From:
Peter Brown
To: [EMAIL PROTECTED]
Sent: Tuesday, December 30, 2003 2:58
PM
Subject: Re: [Asterisk-Users] Asterisk
Config thru web interface or any GUI
Chandra
hi,
i have been using grandstream budgettone IP phones
and they work fine except that these phones times out after some hours.. i ahve
seen that the phones working ok are next day unregistered and sip show peers do
not show their IP and although these phones can make calls , they cannot be
if u are using NAT,
try adding
nat=yes
canreinvite=no
qualify=500
in your [1005]
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 19, 2003 9:31 AM
Subject: RE: [Asterisk-Users] SIP / X-ten Softphone
try adding username=1005
- Original Message -
From: Chandra [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 10, 2003 10:37 PM
Subject: * with RADIUS
hi,
i have been looking for implementations of asterisk with RADIUS which
would
ease for accounting purposes. where can i find more
last time i was experimenting IAXClient as a true
client from dial up and registering it with * behind NAT. i found that the same
account can b used by multiple ppl at the same time. is there any way to
restrict one user per account i/e if on user has already exist then the another
user
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