We've just had the problem where our DNS server went down, and * started
to act funny.
Is the best solution to install a local DNS server on the * box, and
have no other DNS servers ? - this is an internal app, no need for any
external DNS resolution at all.
Julian.
Hello list,
I need to record all calls. So I'm using application Monitor. Works
good until someone transfers a callee to another internal extension.
Example:
A calls B
A set B on hold
A calls C
A transfers B to C with SIP transfer (SIP REFER - with phone funktions
and not Asterisk attended
Hi list,
I am still a newbie and struggling with tweaking the dial plan to my
requirements. I have tried googling for this specific problem, and apologies if
I have overlooked the obvious answer already. If you could please be so kind as
to point me in the right direction, that would be most
Hi Julian,
maybe /etc/hosts can help you...it is faster to setup.
Julian Lyndon-Smith wrote:
We've just had the problem where our DNS server went down, and * started
to act funny.
Is the best solution to install a local DNS server on the * box, and
have no other DNS servers ? - this is an
--- On Thu, 2/5/09, Ex Vito ex.vitor...@gmail.com wrote:
App nvfaxdetect() works fine for that purpose on both Zap
and mISDN.
See http://www.voip-info.org/wiki-NVFaxDetect
Thanks. I setup a system with both nvfaxdetect and the built-in fax detection
because the built-in detection alone
Put faxdetect = none in the misdn.conf and you'll be fine.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Vieri
-- Sent: 06 February 2009 12:40
--
You should get away very easily in the nscd case - there's no config, just
start it. Beware of any negative caching though - failed lookups that stick
and changes that take a bit longer to be recognised by the server.
Good luck!
Best regards
Jan
Date: Fri, 6 Feb 2009 12:47:58 +
From:
Use Set(CALLERID(num)=99) instead of using CALLERID(all).
Remember to set this BEFORE you Dial.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Vieri
-- Sent: 06 February
Huh, sorry, buut.. kmdl rpm requires dahdi-linux:
rpm -i ./dahdi-linux-kmdl-2.6.18-92.1.22.el5-2.1.0.3-59.RHL5.i386.rpm
error: Failed dependencies:
dahdi-linux = 2.1.0.3-59.RHL5 is needed by
dahdi-linux-kmdl-2.6.18-92.1.22.el5-2.1.0.3-59.RHL5.i386
And the rpm, builded with 'kmdl_userland
Paul Chambers wrote:
Josiah Bryan wrote:
snip
Problem is that its crashing for seemingly no reason at all, no errors
on the console, no logs (that I can find), nothing in /var/lib/messages
- its puzzeling! Management is screaming like banshees, calls are
dropping like flies, and all hell
Thanks for the suggestions. Modifying the sendmail command in
voicemail.conf sounds like the most straightforward method, however, I
will first try using 'record' in the dialplan instead of calling
voicemail. This is so I can control the naming of the recorded file. I
will simply run my
Thanks but it still doesn't work.
I did:
-- Executing Set(SIP/4053-b23c5280, CALLERID(num)=99) in new
stack
before Dial(), of course.
I've read somewhere that the misdn debug message:
-- P[ 1] -- TON: Unknown
may mean that the carrier did not recognize the caller id I set. Is
James Moore wrote:
Notice that one of the prohibited items is:
# Phone Services - includes 800 or 900 phone services and audio text
services, prepaid phone cards, and prepaid phone services.
https://payments.amazon.com/sdui/sdui/about?acceptableuse
Google Checkout started with these
You're quite right. We'll need to see your misdn.conf file to check the
settings.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Vieri
-- Sent: 06 February 2009 13:49
Hi,
I understand SRTP and SSIP (encryption for RTP and SIP) is not part of
Asterisk trunk at this very moment. What can I add (not necessarily freely,
I am willing to pay) to Asterisk to accommodate the customers who do need
that level of security? Anything I can put in front of Asterisk to
please in really need help
i looking for an solution some like this.
Laptop/modem(home) - PSTN - Asterisk/or other software using an E1 30
channels with TE122B card or other - enterprice network services
for data connection.
i'm not sure if Asterisk cmd PPPD do that. or if i need a emulate a
Enrique schrieb:
My quiestion is if i can use asterisk to authenticate my users of radius on
Start a new thread please.
http://www.urbandictionary.com/define.php?term=Thread%20Jacking
Your question is not related to extensions ending with #...
Philipp Kempgen
--
AMOOCON 2009, May 4-5,
Hello,
We made small stress-test for H323.
Test shows that H323 protocol is heavyweight compared with SIP.
More details: http://wiki.kolmisoft.com/index.php/H323_pass-through_test
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
Hi
I want to do a dial in server in Linux and I want to use a TE120 digium card
connected to PSTN via E1.
And the users should connect to my network through Linux server, I need help
with this.
In the card documentation seed that supportData Modes: SyncPPP (both Fixed
and Dialup), Frame
Julian Lyndon-Smith wrote:
We've just had the problem where our DNS server went down, and * started
to act funny.
Is the best solution to install a local DNS server on the * box, and
have no other DNS servers ? - this is an internal app, no need for any
external DNS resolution at all
Gunnar Schaller wrote:
Hello list,
I need to record all calls. So I'm using application Monitor. Works
good until someone transfers a callee to another internal extension.
Example:
A calls B
A set B on hold
A calls C
A transfers B to C with SIP transfer (SIP REFER - with phone funktions
Paul Chambers wrote:
I'd recommend dnsmasq. I've been running it for a few years, and it
works very well for me. Besides DNS, it optionally supports DHCP
(integrated with DNS) and TFTP. A basic (i.e. normal :) configuration is
easy to set up, though there's plenty of depth if you need to
Thanks, Ja-Aage and Giorgio - I'll have a go at implementing your
suggestions.
Julian.
Jan-Aage Frydenbø-Bruvoll wrote:
Hi,
nscd (name server caching daemon) is a part of most Linux
distributions as well - maybe that'd help you if your DNS server is
unstable.
Best regards
Jan
On Fri, Feb 06, 2009 at 12:10:52PM +, Jan-Aage Frydenbø-Bruvoll wrote:
Hi,
nscd (name server caching daemon) is a part of most Linux
distributions as well - maybe that'd help you if your DNS server is
unstable.
The NSCD caches names from the name switch. Those names are not only
I'm trying to set caller ids on outgoing calls.
I have a quad BRI B410P card connected to my telephony provider.
I know the list of DID numbers the provider assigned to my company.
If I don't set the caller id then the callee always sees the same top-level
number.
If I set the caller id to a
Hi,
nscd (name server caching daemon) is a part of most Linux distributions as well
- maybe that'd help you if your DNS server is unstable.
Best regards
Jan
Julian Lyndon-Smith wrote:
We've just had the problem where our DNS server went down, and * started
to act funny.
Is the best
On 6 Feb 2009, at 15:35, Enrique wrote:
Hi
I want to do a dial in server in Linux and I want to use a TE120
digium card connected to PSTN via E1.
And the users should connect to my network through Linux server, I
need help with this.
In the card documentation seed that supportData
Hi,
just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same
zaptel/libpri/mISDN/add-ons.
It crashes when transferring a call.
Anybody tried it with success?
Thank you
Giorgio
___
-- Bandwidth and Colocation Provided by
If it's of any help, here's my misdn.conf:
[general]
misdn_init=/etc/misdn-init.conf
debug=0
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
ntkeepcalls=no
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh
[default]
Hello I need someone to install Chan_SCCP for me and get it working on
Elastix with Cisco 7937
Interested party please msn me on sam__tam AT hotmail DOT com or email me
back
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Geoff Lane wrote:
On Thursday, February 5, 2009, Mark Michelson wrote:
I've tried it and you're correct. So it looks like the docs need a
bug report - any idea how I go about that?
If you're using the 2nd edition of the book, check the preface, page xix for
contact information.
Thanks
Giorgio Incantalupo wrote:
Hi,
just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same
zaptel/libpri/mISDN/add-ons.
It crashes when transferring a call.
Anybody tried it with success?
Thank you
Giorgio
___
-- Bandwidth and
On Thu, 2009-02-05 at 22:09 +, Geoff Lane wrote:
Thanks. For info, *TFOT says:
PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to
either SUCCESS or FAILURE. If Caller ID is received on the channel,
PrivacyManager() does nothing.
I've tried it and you're correct. So it
Giorgio Incantalupo wrote:
Hi,
just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same
zaptel/libpri/mISDN/add-ons.
It crashes when transferring a call.
Anybody tried it with success?
Thank you
Giorgio
If you're having crashes occur when transferring a call, you
2009/2/6 Mike l...@virtutel.ca
Hi,
I understand SRTP and SSIP (encryption for RTP and SIP) is not part of
Asterisk trunk at this very moment. What can I add (not necessarily freely,
I am willing to pay) to Asterisk to accommodate the customers who do need
that level of security?
Hi,
My queue used to work fine until I upgraded to 1.6. I am getting the
message:
No application 'AgentCallBackLogin' for extension (default, 31001, 1)
After some rearch I learnt that AgentCallBackLogin is removed in 1.6.
Any one has a configuration that works in place of AgentCallBackLogin in
Is there a way to restrict connection to my asterisk server to users based
on their IP addresses, and not just password. I have some hackers who
connect to my server to make illegitimate solicitation calls to people. I
had to shutdown the server for now until I find a solution. ANY HELP?
Thanks.
Check out this alternative:
http://hostseries.com/agentcallbacklogin-alternative/
Regards,
Robert Broyles
oumar ndiaye wrote:
Hi,
My queue used to work fine until I upgraded to 1.6. I am getting the
message:
No application 'AgentCallBackLogin' for extension (default, 31001, 1)
After
You should be able to do some sort of iptable magic to restrict incoming
activity to specific IP addresses. It depends on your flavor of Linux.
Google linux hardening.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of oumar
The deprecation of Agent Callback login was announced in 1.4.
Robert Broyles wrote:
Check out this alternative:
http://hostseries.com/agentcallbacklogin-alternative/
Regards,
Robert Broyles
oumar ndiaye wrote:
Hi,
My queue used to work fine until I upgraded to 1.6. I am getting
On Thu, 2009-02-05 at 22:01 +0100, Philipp Kempgen wrote:
Mark Michelson schrieb:
Actually, jumping to priority n + 101 is a thing of the past
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most
purposes.
In the same
Another (??) link to check out.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackL
ogin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony
Francis
Sent: Friday, February 06, 2009
Danny Nicholas schrieb:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackL
ogin
http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk:
Anyone have much luck with these on ATA's? I have a few sites that use
them succesfully with multi-port Audiocodes boxes, but just connected ten
machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb
switched network that is barely utilized, then out a T1 on a Sangoma
On Thu, 2009-02-05 at 22:01 +0100, Philipp Kempgen wrote:
Mark Michelson schrieb:
Actually, jumping to priority n + 101 is a thing of the past
And in addition extensions.conf is a thing of the past. ;-)
snip
How about .. dialplan.conf .;-)
A bit of hopefully happy news - the Linksys 2102 has a feature called
modem pass through mode which can be accessed by prepending *99 to the
call. Anyone ever used this? Sounds like that might help with faxing as
well...
j
On Fri, 6 Feb 2009, Jeff LaCoursiere wrote:
Anyone have much
Robert Broyles wrote:
Check out this alternative:
http://hostseries.com/agentcallbacklogin-alternative/
Regards,
Robert Broyles
I like what he came up ,with however it doesn't replace the agent
callback login systems use of being able to make an agent press a key to
accept a call,
Why don't you use followme if you want to do that?
In fact, you can have followme, plus the local agents as mentioned in
the previous alternative that I mentioned.
--
Regards,
Robert Broyles
Anthony Francis wrote:
Robert Broyles wrote:
Check out this alternative:
Anthony Francis schrieb:
Robert Broyles wrote:
Check out this alternative:
http://hostseries.com/agentcallbacklogin-alternative/
---cut---
[agents]
exten = 1050,1,Set(AGENT_SIP=${DB(agent_sip/1050)})
exten = 1050,n,Dial(SIP/${AGENT_SIP})
---cut---
I like what he came up ,with however it
Philipp Kempgen schrieb:
Anthony Francis schrieb:
Robert Broyles wrote:
Check out this alternative:
http://hostseries.com/agentcallbacklogin-alternative/
---cut---
[agents]
exten = 1050,1,Set(AGENT_SIP=${DB(agent_sip/1050)})
exten = 1050,n,Dial(SIP/${AGENT_SIP})
---cut---
I like what
Philipp Kempgen schrieb:
Anthony Francis schrieb:
I like what he came up ,with however it doesn't replace the agent
callback login systems use of being able to make an agent press a key to
accept a call, very important when people are logging in via cell phone
and you don't their voice
Philipp Kempgen schrieb:
SIP (and other protocols) should probably have a Voicemail
header which tells the other party if it's ok to answer the
call by automated means.
Allow-Automated-Answer: voicemail, queue
or some such.
SIPAddHeader(Allow-Automated-Answer: no);
Hmm, this is all very interesting.
Looks like using a Macro and the 'M' Dial() option is the way to go for
now if you need the answer confirmation.
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Look at example #2, and adapt it for your needs.
--
Regards,
Robert Broyles
Philipp Kempgen
...except that Macros are now deprecated and will most likely be removed
in 1.8.
Robert Broyles wrote:
Hmm, this is all very interesting.
Looks like using a Macro and the 'M' Dial() option is the way to go for
now if you need the answer confirmation.
Rob Hillis schrieb:
...except that Macros are now deprecated and will most likely be removed
in 1.8.
Robert Broyles wrote:
Looks like using a Macro and the 'M' Dial() option is the way to go for
now if you need the answer confirmation.
Use U() and Gosubs then!
Philipp Kempgen
--
Le 06.02.2009 22:02, Jan-Aage Frydenbø-Bruvoll a écrit :
A banale example (which does not work):
[outgoing]
exten = _+.,1,Goto(outgoing,00${EXTEN:1},1)
exten = _00.,1,Verbose(International call 00 - Vyke)
exten =
_00.,n,Dial(SIP/vyke/$EXTEN,30,tr)
Hi,
I'm trying to run Asterisk 1.4.23.1 on a small ARM linux board (TS-7800).
Everything compiles fine, but on startup Asterisk always crashes while
loading chan_sip.
If chan_sip is removed, it starts up fine, but I really need SIP to work.
Any ideas?
Thanks.
-- James
Hi!
First thought: try to debug it. Use debugging options while building, if
there are any, and I believe there are.
Then run asterisk with gdb:
gdb asterisk # optionally try asterisk with full path
Then in gdb:
gdb set args your_options
e.g.:
gdb set args -c
Good luck!
Thanks all for your responses.
I am not sure I know every thing AgentCallBackLogin is capable. I don't know
either if I have to have all the functions offered by AgentCallBackLogin.
All I need is a way to allow call takers to login and before they can take
calls. How is this done today in 1.6.
You guys... grr...
I'm still on 1.4 svn ... not ready to even think about 1.6 OR 1.8(when
it's released) for production right now. :-)
--
Regards,
Robert Broyles
Rob Hillis wrote:
...except that Macros are now deprecated and will most likely be removed
in 1.8.
Robert Broyles wrote:
Hmm,
One of my user was asking, can he use VPN to access asterisk ?
What does it mean ?
And its possible ?
How ?VPN
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asterisk-users mailing list
To UNSUBSCRIBE or update options
Hello David,
VPN means 'Virtual Private Network'.
You can have more information about them here:
http://en.wikipedia.org/wiki/Virtual_private_network
And about VPN and Asterisk; yes, of course. If the VPN is working,
there should be no problems using Asterisk inside it (I have one or two
hi
is it possible to set up in the dialplan (on in sip.conf, or something
else) the hostname of the outgoing uri call?
This is my scenario:
- CCM integrated with Asterisk via h323
- SIP user registerd to Asterisk
- Asterisk is behind NAT
- Asterisk ip is 10.10.10.2
- SIP user view Asterisk as
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