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 10.
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
clie
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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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:
>>
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.
Than
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!
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
__
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)
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
...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.
>
> http://www.voip-info.org/wiki-Aste
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 wrote
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);
> Di
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
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
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 howeve
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:
>>> http://hostseries.co
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
ac
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
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. ;-)
How about .. dialplan.conf .;-)
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 card.
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: http://the-aster
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
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 s
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
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 oum
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)
> Aft
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.
on
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
1
2009/2/6 Mike
> 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
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 cal
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.
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
>
> ___
> -- Band
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 informatio
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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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]
context=m
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
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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 supportD
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 t
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 funkt
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 a
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 R
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
___
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,
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 so
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
e
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 onl
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 200
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 th
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
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 externno
Paul Chambers wrote:
> Josiah Bryan wrote:
>>
>> 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
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 1'
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 F
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 +
> Fr
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 12:4
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 Thu, 2/5/09, Ex Vito 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 didn't always work, ev
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 s
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
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
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 a
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 transfer
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.
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