Should a Linksys Sipura 2102 be configured with nat=yes even if it is on
the local network?
I have been having some troubles with a Linksys Sipura 2100 series, which
suffers from NO AUDIO after a few calls.. Because it is on the same subnet
as Asterisk it is configured with nat=no. When you think
Freeswitch was engineered from scratch by some Asterisk developers who
wanted to start afresh on a cleaner programming base. Asterisk is like
Topsy, She just growed and had to maintain backward compatibility.
The latest versions of Asterisk are reported to be much improved in that
respect.
On 7
Why doesn't this manager.conf code work on Asterisk 1.6.2 and 1.8.9? It
works perfectly on Asterisk 1.4
In Asterisk 1.6 it appears to disconnect as soon as events occur and in
1.8.9 it can't be read at all. Apparently it has some syntax issues with
1.8.9.
Is it possible to tell at a glance what
When I do a make menuselect cdr_adaptive_odbc is disabled.
What packages are required to enable it?
Even after executing apt-get install unixodbc libmyodbc odbc-postgresql
tdsodbc unixodbc-bin it is still disabled.
What am I missing?
/voipfc
--
[mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Frank Church
*Sent:* Monday, February 06, 2012 7:39 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] What packages are required to
getcdr_adaptive_odbc to be compiled in Asterisk?
When
Are there any ATAs that support IPv6 in the wild, given that IP4 address
are running out?
/voipfc
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:
match_auth_username=yes
Leandro
2012/1/19 Frank Church voi...@gmail.com:
Does Asterisk permit multiple registrations to the same host?
Each registration has a different username and password
The purpose is for billing, handling incoming calls is not important,
although it will be a bonus.
I guess I
Does Asterisk permit multiple registrations to the same host?
Each registration has a different username and password
The purpose is for billing, handling incoming calls is not important,
although it will be a bonus.
I guess I should also ask the converse, whether the receiving host can
accept
I am logging events from the AMI and the PeerStatus and Registry
events show that the privilege for them is System,All.
Can a lower set of privileges be used? All looks pretty high to me.
/Frank
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Of late I have noticed bruteforce attempts to register to a homebased
Asterisk server which is behind a router firewall and I need to know
whether the router on the firewall is the culprit here. It only stops
after the router is restarted.
The router is the HG521 from talktalk and the firewall
I have been seeing some attempts to register devices on my Asterisk
and I want to reconfigure it so that devices will be registered only
if they are from the correct address, ie 192.168.1.8/255.255.255.255.
I thought using a config like
deny=0.0.0.0/0.0.0.0
permit=192.168.1.8/255.255.255.255
On 6 August 2010 16:21, Bruce Ferrell bferr...@baywinds.org wrote:
On 08/06/2010 07:45 AM, Frank Church wrote:
I have been seeing some attempts to register devices on my Asterisk
and I want to reconfigure it so that devices will be registered only
if they are from the correct address, ie
On 7 August 2010 03:54, Bruce Ferrell bferr...@baywinds.org wrote:
On 08/06/2010 07:30 PM, Bruce Ferrell wrote:
On 08/06/2010 02:16 PM, Frank Church wrote:
On 6 August 2010 16:21, Bruce Ferrell bferr...@baywinds.org wrote:
On 08/06/2010 07:45 AM, Frank Church wrote:
I have been seeing
Is there a database of MAC address prefixes used the common VoIP
devices. I see the Linksys Sipura devices state with 00:0E.
Does the same apply to other Linksys VoIP equipment?
Is there some way VoIP equipment allow themselves to be identified by
requesting data from some ports?
--
More googling got me this page - http://www.freepbx.org/v2/wiki/DevicesTakeTwo
Very useful
Thanks
On 12 July 2010 16:41, Frank Church voi...@googlemail.com wrote:
Is there a database of MAC address prefixes used the common VoIP
devices. I see the Linksys Sipura devices state with 00:0E
What is the minimal module set required to run SIP with database CDR logging.
I compiled Asterisk from source and I obviously compiled more stuff
than I needed for VoIP and CDR logging to postgres.
Sometimes there is a long gap between Asterisk starting and devices
being able to register. sip
The DNS setup itself is fine. The sip module just seems to take too
much time to load. My modules.conf uses autoload=yes and it seems that
many unwanted modules are loaded before sip itself starts.
On 30 June 2010 13:52, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
I have been monitoring AMI events and realized that they don't have timestamps.
Is that standard behaviour, or is there some way to get them to
include timestamps?
I am on 1.4. Is it available on 1.6?
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On 30 March 2010 02:04, Mark Phillips g7...@g7ltt.com wrote:
They say confession is good for the soul. Perhaps they are offering a
phone in confessional service?
Unfortunately the business of the church often flies in the face of
the business of the Church.
On 03/29/2010 07:48 PM, Alex
On 29 March 2010 21:46, Frank Church voi...@googlemail.com wrote:
I have been asked by my church to recommend a VoIP system which can do
the following.
They do internet radio shows which are sometimes broadcast on radio.
They are looking for a system which does the following for about 5
thing for me. One of the distributions
with SugarCRM comes to mind here.
Sorry for cross-posting, but ready made and commercially supported
systems are not ruled out, if they come within their budget.
Regards
Frank Church
thing for me. One of the distributions
with SugarCRM comes to mind here.
Sorry for cross-posting, but ready made and commercially supported
systems are not ruled out, if they come within their budget.
Regards
Frank Church
Is there a way for a client to tell a server where it is registered to
remove the registration?
/voipfc
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Thanks.
Is there command is used for that?
I have checked the help show and there is no command like sip register
or sip unregister in the list.
Is it available on version 1.4?
On 11 March 2010 13:08, Kevin P. Fleming kpflem...@digium.com wrote:
Frank Church wrote:
Is there a way
and billsec is set to duration, or duration - 1.
Is that behaviour that has been observed before?
On 8 February 2010 17:00, Frank Church voi...@googlemail.com wrote:
The behaviour of my Asterisk appears to have changed suddenly without
any apparent cause.
The version is use is 1.4.27.1
When a call
The behaviour of my Asterisk appears to have changed suddenly without
any apparent cause.
The version is use is 1.4.27.1
When a call is not answered billsec is set to duration, and calls are
charged. I can't see any change I could have
made to cause this problem. Is it something already known in
I have a Linksys Sipura SPA2102 connected to Asterisk 1.4.27 and
sometimes it doesn't connect at all. I keep getting a busy signal when
I try to dial.
It appears to happen most often when both lines are registered.
The 2 lines on Linksys lines also use different ports. Does that mean
than it is
I have developed a minimal call shop billing system that includes an
Asterisk VM and I want it to be as small as to reduce the installation
size.
100Mb is good
On 2 February 2010 05:41, Frank Church voi...@googlemail.com wrote:
How small can an Asterisk system be, in terms of disk space
How small can an Asterisk system be, in terms of disk space utilized?
I am looking for just asterisk, with mysql, postgresql, or sqlite,
with PHP and Python.
After finishing the build and removing the tools, how small can the
whole system be?
100Mb, 200Mb?
Can packages be used to build the
I have installed an Asterisk 1.4 on Suse93 using a FritzCard.
Some calls are logged to the ISDN log, but Asterisk is not detecting
incoming calls.
I wonder whether some other device or process is preventing Asterisk
from gaining access to the isdn line?
Is there some way to ensure that only
On 26/11/2007, Per Jessen [EMAIL PROTECTED] wrote:
Frank Church wrote:
I have installed an Asterisk 1.4 on Suse93 using a FritzCard.
Some calls are logged to the ISDN log, but Asterisk is not detecting
incoming calls.
I wonder whether some other device or process is preventing
On 26/11/2007, Frank Church [EMAIL PROTECTED] wrote:
I have installed an Asterisk 1.4 on Suse93 using a FritzCard.
Some calls are logged to the ISDN log, but Asterisk is not detecting
incoming calls.
I wonder whether some other device or process is preventing Asterisk
from gaining access
Are there ATAs that allow different phone numbers from one network connection?
Such as supporting multiple IP addresses so that each RJ11 has a
different extension or some other way?
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I want to use a Fritz AVM ISDN card to create a switch which is
connected to 4 analogue extensions.
I believe I need a 4 port FXS module for that, are there any cheap but
reliable options out there?
Are there some guides that go through the whole process?
/voipfc
../../ast_trunk_manager_PlayDTMF.patch
Regards
On 10/13/06, Frank Church [EMAIL PROTECTED] wrote:
When I try to apply this patch - ast_trunk_manager_PlayDTMF.patch - I
receive the error below
missing header for unified diff at line 3 of patch
can't find file to patch at input line 3
Perhaps you used
Check the rpttimeout setting in sip.conf for the necessary extensions.
As you are not making normal calls to outside parties, that setting
may not applicable in your case.
On 10/13/06, Jason Adams [EMAIL PROTECTED] wrote:
Hey All,
Sometimes we are running into issues with calls randomly
to this was:
--
|--- app_senddtmf.c~2006-05-04 15:27:41.0 -0500
|+++ app_senddtmf.c 2006-05-04 15:29:21.0 -0500
--
File to patch:
Is there something else I am missing
On 10/12/06, Frank Church [EMAIL PROTECTED] wrote:
Hi Moises,
I have looked on that page
are never added to release branches, so you need to patch
1.2.12.1 adapting the trunk patch. Dont worry, is an easy patch.
Regards
On 10/11/06, Frank Church [EMAIL PROTECTED] wrote:
Hi Moises, Ignore my last reply about the presence of the DTMF in 1.4.
Do you have the source for patching
Hi Moises,
does the you mentioned earlier at
http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch include the DTMF
event, or is it for PlayDTMF and SendDTMF?
Looking through the actions on bug6082 it is hard to tell whether
the DTMF event patch is still in there when I last compiled that
branch
Hi Moises, Ignore my last reply about the presence of the DTMF in 1.4.
Do you have the source for patching the DTMF event?
There is no link to it on the bug6082 page, and I am not quite sure
how it can be obtained from SVN.
Regards
Richard
On 10/12/06, Frank Church [EMAIL PROTECTED] wrote
Moises, do you know if the DTMF event in bug 6082 made it into version 1.4?
When I last tried to compile that branch it needed the latest version
of make 3.81, which trunk did not, and caused me to wonder if it had
been committed to trunk.
The DTMF detection events in trunk did not also
On 10/4/06, Moises Silva [EMAIL PROTECTED] wrote:
I could be wrong here, but I think that you're looking for SendDTMF and
not PlayDTMF. getting it confuddled with PlayTones?
He is not confused. PlayDTMF is a manager command, not an dial plan
application, but included in the same module that
]
console = notice,warning,error,verbose,debug
Regards
On 9/15/06, Frank Church [EMAIL PROTECTED] wrote:
The program in question is an adaptation an AGI calling card program.
It is adapted for callback by setting by channelling the callback call
into the context used for the normal inbound leg
the channel and the dtmf numbers as parameters and send the DTMF
signals?
Frank Church
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ideas of what the problem might be?
On 9/14/06, Moises Silva [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF
Regards
On 9/14/06, Frank Church [EMAIL PROTECTED] wrote:
How can DTMF be sent down a channel?
I am thinking of method where say a channel id can
and it .
On 9/15/06, Moises Silva [EMAIL PROTECTED] wrote:
Frank. PlayDTMF and SendDTMF is the same as pressing keys at the
phone. Im not understanding well, can you please explain a practical
scenario of how do you expect it to work, and how actually works? :)
Thanks
Regards
On 9/14/06, Frank Church
Does this device allow connection to other phones besides Skype, like
Xten Xlite?
http://www.voipvoice.com/UConnect-2.html.
Compatibility with standard voip is not mentioned on their website?
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Bump?
On 9/7/06, Frank Church [EMAIL PROTECTED] wrote:
Does this device allow connection to other phones besides Skype, like
Xten Xlite?
http://www.voipvoice.com/UConnect-2.html.
Compatibility with standard voip is not mentioned on their website
people who might have tried it before.
On 9/7/06, Frank Church [EMAIL PROTECTED] wrote:
Bump?
On 9/7/06, Frank Church [EMAIL PROTECTED] wrote:
Does this device allow connection to other phones besides Skype, like
Xten Xlite?
http://www.voipvoice.com/UConnect-2.html .
Compatibility
When SendDTMF is used on a channel, which party is the DTMF being sent
to, the callee or the caller?
What is the syntax for using SendDTMF in an AGI command?
F Church
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