i am using a x lite phone.
regards
upendra
On Fri, Jul 13, 2012 at 10:29 AM, James Sharp ja...@fivecats.org wrote:
Different phones use different methods. What kind of sip phones do you
have?
On Jul 13, 2012, at 12:17 AM, upendra uppi...@gmail.com wrote:
Hi,
i wanted to make
From my experience with xlite, the soft phone itself must be configured for
auto answer. There is no way for the dialplan to control this.
On Jul 13, 2012, at 2:35, upendra uppi...@gmail.com wrote:
i am using a x lite phone.
regards
upendra
On Fri, Jul 13, 2012 at 10:29 AM,
It is known as intercom feature. Some soft and hard phone support this
feature. You can use following context for it.
[intercom-context]
Exten = 9155,1,SIPAddHeader(Call-Info:
sip:pbx.your-company.de\;answer-after=0)
Exten = 9155,2,Dial(SIP/155)
Fow more details, use following link
hi,
thats fine , i am using now xlite-4 , not able to find the option to enable
it, let me know .
regards
Upendra.
On Fri, Jul 13, 2012 at 1:03 PM, James Sharp ja...@fivecats.org wrote:
From my experience with xlite, the soft phone itself must be configured
for auto answer. There is no way
Hi guys!
I have a some non standard problem when I register my asterisk into My
SIP Provider .
The trouble is: my asterisk stay behind router with port forwarding, who
have Public IP (55.55.55.55 - for example), asterisk have a private IP
(192.168.1.2)
From My SIP Provider cabinet I see:
13.07.2012 13:00, Elliot Murdock ?:
Hello,
Which tools are recommendable for monitoring VOIP, bandwidth, server
alarms, etc.?
Thanks,
Elliot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
hi,
oh k thanks then i will re-install the xlite 3.
Regards
Upendra
On Fri, Jul 13, 2012 at 2:42 PM, Bharat Lalcheta
bharatlalch...@gmail.comwrote:
From X-lite 4 version, auto answer feature removed. So use old version
if you have or try some other softphone
On Fri, Jul 13, 2012 at 1:51
Hi,
I am trying to write dial plan for sip to auto answer (auto attend) the
incoming call to the sip phone.
- If i call from sip1 to sip2 then sip2 should automatically answer the
call and play some sound file.
I am trying to do this but as new to the asterisk dial plan configuration ,
so not
try with SipAddHeader(uri=answer-after=0)
check syntax for Addheader
Regards,
Zohair Raza
On Fri, Jul 13, 2012 at 1:42 PM, upendra uppi...@gmail.com wrote:
Hi,
I am trying to write dial plan for sip to auto answer (auto attend) the
incoming call to the sip phone.
- If i call from sip1
Hi,
thanks , i need to put this in the sip context...
regards
Upendra.
On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:
try with SipAddHeader(uri=answer-after=0)
check syntax for Addheader
Regards,
Zohair Raza
On Fri, Jul 13, 2012 at 1:42 PM,
In dialplan
http://www.voip-info.org/wiki/view/Asterisk+cmd+SipAddHeader
Regards,
Zohair Raza
On Fri, Jul 13, 2012 at 1:50 PM, upendra uppi...@gmail.com wrote:
Hi,
thanks , i need to put this in the sip context...
regards
Upendra.
On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza
2012/7/13 Nikolay G. Petrov r...@dir.bg
Hi guys!
I have a some non standard problem when I register my asterisk into My
SIP Provider .
The trouble is: my asterisk stay behind router with port forwarding, who
have Public IP (55.55.55.55 - for example), asterisk have a private IP
Kevin P. Fleming kpflem...@digium.com writes:
I've just looked into this a bit, and I don't see how using connect()
would actually solve the problem. If we receive a UDP datagram from a
SIP endpoint, we could use socket() and connect() to create a socket
specifically for sending to (and
Raj Mathur (राज माथुर) r...@linux-delhi.org writes:
Precisely. In fact, if a packet from 192.168.2.n is received on /any/
interface, the response will always go out from the 192.168.2.X
interface. (Barring some weird routing/iptables configuration, of
course.)
This is only the case for
13.07.2012 15:01, Leandro Dardini ?:
2012/7/13 Nikolay G. Petrov r...@dir.bg mailto:r...@dir.bg
Hi guys!
I have a some non standard problem when I register my asterisk
into My SIP Provider .
The trouble is: my asterisk stay behind router with port
forwarding, who have
Il giorno 13/lug/2012 14:00, Nikolay G. Petrov r...@mail.bg ha scritto:
13.07.2012 15:01, Leandro Dardini пишет:
2012/7/13 Nikolay G. Petrov r...@dir.bg
Hi guys!
I have a some non standard problem when I register my asterisk into My
SIP Provider .
The trouble is: my asterisk stay behind
On 12-07-13 06:00 AM, Elliot Murdock wrote:
Hello,
Which tools are recommendable for monitoring VOIP, bandwidth, server
alarms, etc.?
Nagios (http://www.nagios.org/) can be configured to monitor pretty much
anything you want. The (much) harder part is deciding what's relevant to
monitor,
13.07.2012 16:25, Leandro Dardini ?:
Il giorno 13/lug/2012 14:00, Nikolay G. Petrov r...@mail.bg
mailto:r...@mail.bg ha scritto:
13.07.2012 15:01, Leandro Dardini ?:
2012/7/13 Nikolay G. Petrov r...@dir.bg mailto:r...@dir.bg
Hi guys!
I have a some non standard problem when I
On 29.06.2012 09:00, Michelle Konzack wrote:
Hello Armin Schindler,
Am 2012-06-28 09:52:30, hacktest Du folgendes herunter:
On 27.06.2012 18:46, Michelle Konzack wrote:
Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?
Which PCI-ID is that?
I do not know, because I have not
On Thu, Jul 12, 2012 at 3:55 PM, Ellen Apolinar
ellen.apolinar...@googlemail.com wrote:
Hello mailinglist,
I want to connect Asterisk with OpenBTS and make a call with a mobile phone.
I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone
You also have to send the alert info you particular phone needs to make it
autoanswer.
On Jul 13, 2012 4:53 AM, upendra uppi...@gmail.com wrote:
Hi,
thanks , i need to put this in the sip context...
regards
Upendra.
On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza
On 12-07-13 08:37 AM, Mike wrote:
On 12-07-13 06:00 AM, Elliot Murdock wrote:
Hello,
Which tools are recommendable for monitoring VOIP, bandwidth, server
alarms, etc.?
Nagios (http://www.nagios.org/) can be configured to monitor pretty much
anything you want. The (much) harder part is
Smokeping with sip probe is quite nice
Sent from BETA iOS6
On Jul 13, 2012, at 4:54 PM, Paul Belanger paul.belan...@polybeacon.com wrote:
On 12-07-13 08:37 AM, Mike wrote:
On 12-07-13 06:00 AM, Elliot Murdock wrote:
Hello,
Which tools are recommendable for monitoring VOIP, bandwidth,
Thanks whoever is running an auto response ticket system!
Look forward to getting more spam from you!
Sent from BETA iOS6
On Jul 13, 2012, at 4:54 PM, Paul Belanger paul.belan...@polybeacon.com wrote:
On 12-07-13 08:37 AM, Mike wrote:
On 12-07-13 06:00 AM, Elliot Murdock wrote:
Hello,
Hi,
its not working for me ! let me know anyone having sample dialplan so
that i can use for test 1 sip call answer.
regards
Upendra
On Fri, Jul 13, 2012 at 9:57 PM, Jared Baxley jared.bax...@gmail.comwrote:
You also have to send the alert info you particular phone needs to make it
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