14 feb 2010 kl. 03.25 skrev C F:
Excellent and very informative article, Thanks Olle.
You're welcome.
I ran thru lots of my dialplans now quickly to see if I have a catch
all exten anywhere. I couldn't find any that are accessible
unauthenticated, I always declare all fixed length
My dialer works perfectly , but whenever I dial a number manually from xlite
and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as
soon as I press any key from xlite
What could be the issues ?
I tried the SAME VOIP from another center and Its Ok there.
I tried the Same
Hello all,
this may be slightly offtopic :-)
I have some Cisco 7940 phones with SIP firmware, connected to an
Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g (HorstBox Pro with custom
extensions.conf).
On some of the phones, two lines are configured, one for business, one
for private calls.
When
Hello Scott,
First, I want to thank you for your good help.
I need to handle all the failure situations of voip calls. Sometimes, the
source of failure are the ISP and the government theirselves who inspects
traffic with powerful firewalls and sometimes the problem comes from the
client who does
Thank you.
Good tip.
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Appears to be unresponsive.
Doug
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Problem is that the port 80 you are talking about is a TCP port. Voip
(iax and rtp) use UDP
Yes true. HTTP uses 80 TCP port.
I mentioned port 80 as example (even if it can be used for SIP signaling:
SIP supports also TCP). For RTP, UDP must be used. We can use another well
known UDP port.
From a technical point UDP and TCP ports are separate, a server
listening for TCP requests on port 80 wont see any UDP traffic on that
port unless it explicitly opens a UDP socket. Tunneling in on UDP port
80 might be possible if the routing rules that are in place dont
specify to allow only TCP
i had configured voicemail, here is the config files voicemail.conf, sip.conf,
extensions.conf, zaptel.conf, zapata.conf
voice mail is working when ever call is received, extension 2000 receives it
and if not answered in 20 secs, message is stored in
voicemail no problem in that. after
On Sunday 14 February 2010 06:43:37 Doug Lytle wrote:
Appears to be unresponsive.
It appears that somebody's bot was hammering the site. This has since
subsided, so the site is back to normal.
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Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig
I have a server that is receiving a disconnect during recording of long
incoming messages. The connection is via a SIP gateway and when the gateway
sees no RTP for 5 mins, it hangs up the call. I enabled
transmit_silence_during_record but I see no RTP being sent from Asterisk to
the gateway
Any help ?
On Sun, Feb 14, 2010 at 3:03 PM, Global Meds gm.cu...@gmail.com wrote:
My dialer works perfectly , but whenever I dial a number manually from
xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets
DC as soon as I press any key from xlite
What could be the
Hello,
My host just had a faulty power supply and therefore, my Asterisk server was
down for 7 hours.
It was a Sunday so no one was making calls, however if it happened during the
week, I'd have problems.
I was trying to find a whitepaper or advice on how to set up two Asterisk
servers to
I’ve been googling “asterisk redundancy” but all I’ve found is questions,
and no real answers.
Is this any help Dan?
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
Chris
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In 1.6, a SIP invite checks for valid [user] then TO:
domain=abc.com,context, then if not present [general]
context=incoming. This is fine and I think I understand the reasoning
behind the new method. But we have lost the ability to route calls
based on 'from' ip address.
No, 1.6.x should
On Sun, Feb 14, 2010 at 11:42 AM, Dan Journo
d...@keshercommunications.com wrote:
Hello,
My host just had a faulty power supply and therefore, my Asterisk server was
down for 7 hours.
It was a Sunday so no one was making calls, however if it happened during
the week, I’d have problems.
I agree that better hardware is needed.
I'm looking into buying my own servers and getting a rack in a data centre.
I'll impliment a redundancy solution at the same time.
Thanks for the links.
Dan Journo
Kesher Communications Ltd
-Original Message-
From: Steve Totaro
On Fri, Feb 12, 2010 at 12:54 PM, James Lamanna jlama...@gmail.com wrote:
Hi, I have a PRI problem where it appears that my system is not
responding to SETUP messages on a channel.
It seems to be retransmitting a significant number of RELEASE messages
to clear a call that is most likely
to be
2010/2/14 Oliver Nittka o...@nittka.com
- dialplan app SendText() only works on a connected channel, AFAIK, and
I'm not sure if the 7940 with SIP firmware honours it at all.
Have you tried to call the phone with auto-answer ?
--
On Sun, Feb 14, 2010 at 2:30 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sat, Feb 13, 2010 at 09:25:01PM -0500, C F wrote:
Excellent and very informative article, Thanks Olle.
I ran thru lots of my dialplans now quickly to see if I have a catch
all exten anywhere. I couldn't find any
On Sun, Feb 14, 2010 at 3:26 AM, Olle E. Johansson o...@edvina.net wrote:
14 feb 2010 kl. 03.25 skrev C F:
Excellent and very informative article, Thanks Olle.
You're welcome.
I ran thru lots of my dialplans now quickly to see if I have a catch
all exten anywhere. I couldn't find any that
strip_ampersands(${EXTEN})?
On Sun, Feb 14, 2010 at 10:56 AM, C F shma...@gmail.com wrote:
On Sun, Feb 14, 2010 at 3:26 AM, Olle E. Johansson o...@edvina.net wrote:
14 feb 2010 kl. 03.25 skrev C F:
Excellent and very informative article, Thanks Olle.
You're welcome.
I ran thru lots of my
Olivier schrieb:
2010/2/14 Oliver Nittka o...@nittka.com
- dialplan app SendText() only works on a connected channel, AFAIK, and
I'm not sure if the 7940 with SIP firmware honours it at all.
Have you tried to call the phone with auto-answer ?
I'm afraid that's not possible with the 7940.
Hello Scott,
Thank you for your kind support.
All your ideas are helpful.
I will check the OpenVPN solution first. then, I will see if Skype and IAX
may help.
Best Regards.
Abdelkader Mosbah.
♫ *Please discover scientific miracles of CORAN:*
http://www.55a.net/
On Sun, Feb 14, 2010 at 2:57
On Sun, 14 Feb 2010, Kyle Kienapfel wrote:
strip_ampersands(${EXTEN})?
(sip.conf)
[general]
allow-characters= all
disallow-characters =
[example-did-provider]
allow-characters= [:numeric:]
--
Thanks in advance,
While I like these solutions, they should never be substituting a good
secure dialplan.
On Sun, Feb 14, 2010 at 3:04 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Sun, 14 Feb 2010, Kyle Kienapfel wrote:
strip_ampersands(${EXTEN})?
(sip.conf)
[general]
allow-characters
14 feb 2010 kl. 21.04 skrev Steve Edwards:
On Sun, 14 Feb 2010, Kyle Kienapfel wrote:
strip_ampersands(${EXTEN})?
(sip.conf)
[general]
allow-characters= all
disallow-characters =
[example-did-provider]
allow-characters
Olivier schrieb:
Thanks for the suggestion anyway, I'm going to test this just out of
curiosity :-)
And that's what i get in the CLI:
Got SIP response 501 Not Implemented back from XXX.XXX.XXX.XXX
Well, I guess I should really give chan_sccp another shot ...
Thanks anyway!
-- o
--
i am getting a problem, when a call is received by an sip extension it receives
the call no problem in that. but if somebody calls again on that no busy tone
is displayed.i think its a signal problem. so plz tell me
how to have disconnect signals enabled in line.
Your Mail works
On Sun, Feb 14, 2010 at 11:22:12AM -0800, Kyle Kienapfel wrote:
strip_ampersands(${EXTEN})?
You forget other potentially harmful characters.
@:,/|
And maybe others.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
Can we debug or get the log file when we press any key in Xlite and which
send to asterisk and the output we get ?
On Sun, Feb 14, 2010 at 3:03 PM, Global Meds gm.cu...@gmail.com wrote:
My dialer works perfectly , but whenever I dial a number manually from
xlite and press a Key like 6055
Interestingly .Note this :
Phone Number : 7274507674 Room ID: 6055
When I dial this number through Xlite and asterisk , on pressing any key ,
line get disconnect.
When I dial this number through Skype, its perfect.
Phone Number : 2127773456
When I dial this number through Xlite and asterisk
I'm using insecure=invite with two different dial plans, it it working with
one dial plan but not with the other.
What other parameters could influence insecure=invite
In sip.conf below insecure=invite is working OK
[pstn-1270]
type=friend
secret=spa3k
username=voice-1270
mailbox=369
Wow, can't believe I missed that.
Thanks so much!
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How to capture keys entered through soft phone and what action asterisk is
taking on that key ?
Any Log of that ?
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To
On Mon, Feb 15, 2010 at 03:29:55AM +0530, cool dude wrote:
i am getting a problem, when a call is received by an sip extension it
receives the call no problem in that. but if somebody calls again on that no
busy tone is displayed.i think its a signal problem. so plz tell me
how to have
On Mon, 2010-02-15 at 06:55 +0530, Global Meds wrote:
[cut]
FYI: You may not be aware but with your 'Global Meds' name you are going
to be ending up in lots of spam filters. I've just had to fish your
posts out of a Quarantine.
--
Would any setting in extension.config caused this strange hebaviour?
--
Joseph
On 02/14/10 17:35, Joseph wrote:
I'm using insecure=invite with two different dial plans, it is working with
one dial plan but not with the other.
What other parameters could influence insecure=invite
In sip.conf
ohh Didnt notice !!
On Mon, Feb 15, 2010 at 6:55 AM, Global Meds gm.cu...@gmail.com wrote:
How to capture keys entered through soft phone and what action asterisk is
taking on that key ?
Any Log of that ?
--
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--
voice mail is working when ever call is received, extension 2000 receives it
and if not answered in 20 secs, message is stored in
voicemail no problem in that. after creating voice mail if some one again call
at that no this time even bell dosent ring, busy
tone is heard, but when i restart
On Mon, Feb 15, 2010 at 06:41:14AM +0200, Tzafrir Cohen wrote:
On Mon, Feb 15, 2010 at 03:29:55AM +0530, cool dude wrote:
i am getting a problem, when a call is received by an sip extension it
receives the call no problem in that. but if somebody calls again on that
no busy tone is
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