> > Asterisk 1.4.42
> >
> > Set alwaysauthreject=yes in [general] section of sip.conf.
> > Restarted asterisk
> >
> > However when I attempt to register I still get:
> > [2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from
> > '' failed for '121.98.1.1' -
> Wrong
> > password
> > [2012
Thanks for the reply however it is not possible to get the public IP address
using the SIP_HEADER function (see my original post).
We have many devices connecting from hundreds of dynamic external IPs.
--
_
-- Bandwidth and Col
Asterisk 1.4.42
Set alwaysauthreject=yes in [general] section of sip.conf.
Restarted asterisk
However when I attempt to register I still get:
[2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from
'' failed for '121.98.1.1' - Wrong
password
[2012-08-08 21:12:42] NOTICE[15689] chan_sip.
Thanks.
exten => s,n,Set(foo=${CHANNEL(peerip)}) ; Doesn't return anything
exten => s,n,Set(foo=${CHANNEL(recvip)}) ; Doesn't return anything
exten => s,n,Set(foo=${SIPCHANINFO(peerip)}) ; Returns public IP when
calling from a SIP device
Strange that CHANNEL doesn't return anything.
--
___
We are looking to further secure our Asterisk installation by inspecting the
IP address that a SIP INVITE comes from and performing some logic to
determine whether the call should proceed. The purpose of this is to prevent
calls to certain expensive destinations if the SIP message is coming from a
Asterisk 1.4
We are experiencing an issue on transfers where no progress tones are heard
by the caller:
1. Call from 1593 (SPA525G 0026998D2) to 1595 (SPA922 000B820AF).
1595 answers
2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4A). 1595 hears
MoH.
3. 1597 starts ringing and 1
On Nov 8, 2011, at 9:55 AM, isr...@gmail.com wrote:
There is a bug which blocks call progress message 8 which was fixed
but I don't remember in which version
Try upgrading to latest 1.6 version
Before we opened for the day today I updated to 1.6.2.20 and that
seems to have solved the ca
I recently switched to a PRI from analog lines. For reasons out of my
control, my vendor had problems getting the PRI to interface so they
set it to use T1-CAS instead. The lines are working just fine for
inbound and outbound calls, except I get no call progress sounds. So
no ring, busy, et
> On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote:
> > Are there any plans to include the ISAC codec in Asterisk? Is it
> possible or
> > even desirable? Is ISAC open source (nothing indicates it is from the
> WebRTC
> > website http://www.webrtc.org)?
>
>
Are there any plans to include the ISAC codec in Asterisk? Is it possible or
even desirable? Is ISAC open source (nothing indicates it is from the WebRTC
website http://www.webrtc.org)?
--
_
-- Bandwidth and Colocation Provided b
We have multiple customers running on a single Asterisk 1.4 installation and
therefore require a large number of pickup groups. There seems to be a
limitation of 64 call groups. Can anyone suggest how we work around this?
For example is this limitation removed in a later version, is there a patch,
>
> On Wed, May 04, 2011 at 09:56:40PM +1200, CB wrote:
> > I am attempting to install Dahdi on a virtual machine running Centos
> 5.5 and
> > having various problems.
> >
> > yum install kernel-devel gcc make gcc-c++ libxml2-devel
> > Loaded plugins: faste
I am attempting to install Dahdi on a virtual machine running Centos 5.5 and
having various problems.
yum install kernel-devel gcc make gcc-c++ libxml2-devel
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
* base: mirror.optus.net
* extras: mirror.optus.net
* rpmforge: fr
On Apr 15, 2011, at 12:50 PM, satish patel wrote:
We had echo issue before so we replaced old PRI card with Sangoma
A102D HWEC. Now my question is i set rx/txgain level 0.0 default do
i need to touch this value or default is best. I have read on google
and people say it should around 14844
Could anyone recommend some documentation regarding Asterisk 1.8 and the
realtime architecture? Specifically I want to know if it is possible to set
a priority label or to use n as a priority for realtime extensions in
Asterisk 1.8? My understanding is that is not possible with Asterisk 1.4 and
I w
On Sep 15, 2010, at 6:10 PM, Al lists wrote:
> we tried to use fxtune but looks like it wont work with Sangoma
> card, ( please correct me if i'm wrong)
> Echo is really bad and also we have background noise on all lines.
> We tried both mg2 and oslec echo canceler.
I've only used Sangoma wit
On Mar 10, 2010, at 10:27 PM, Chris Owen wrote:
>> This normally works fine but occasionally when someone picks up the
>> call other phones don't seem to realize the call has been answered
>> and will continue to ring. On at least once occasion I saw a call
>> that went to voicemail and al
On Jan 24, 2010, at 12:22 PM, Karl Fife wrote:
> --I can adjust the volume easily without looking AND without fat-
> fingering
> some DTMF tones--very good haptics. With the Snom you have to look
> and
> guide your fingers to the volume buttons or you'll inadvertently
> beep some
> DTMF's.
On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote:
> Would be a cause of static for inbound/outbound and ext to ext calls?
>
> Its voip both in and out.
>
> We swapped, phones, cordes, switches etc…..
>
> Typically a reboot of the phone resolves the problem…person also
> swears there is nothing on
I've done gain tuning as per the info I've found online. I've got my
RXGain set so my volumes list as about 14,800 (using a milliwatt test
number and ztmonitor -vv). However listening to the line now, this
sounds too loud to me. The person speaking sounds fine, but I've now
got a large amou
I'm not sure where to check next, so I'm reaching out to those that
know this stuff better than I.
I've got Asterisk up and running, but I've still got an occasional
audio issue. Once in a while (maybe 1 out of every 20-30 calls), the
audio becomes heavily distorted, but only on the local si
On May 14, 2009, at 11:34 AM, Doug Lytle wrote:
> SIP phones send a completed dial string, analog phones send 1 digit
> at a
> time. With the timeout values, it no more digits are recieved by
> the 2
> second timeout, the dial plan continues.
Ok, that was the part I didn't understand that m
On May 13, 2009, at 10:50 AM, Doug Lytle wrote:
>> I have a Sangoma A400 card with two FXS ports. They work fine,
>> however as I have analog phones connected, I have no way of telling
>> the phone I am done dialing. Pressing # works fine, but then Asterisk
>>
>
> That's what the digit and respons
I have a Sangoma A400 card with two FXS ports. They work fine,
however as I have analog phones connected, I have no way of telling
the phone I am done dialing. Pressing # works fine, but then Asterisk
passes that # over to the POTS line, and about every 5th call, for
some reason that is cau
On Oct 29, 2007, at 5:35 PM, Kyle Sexton wrote:
> Does anyone know who really makes this phone:
>
> http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/
>
> Large pictures are at the bottom:
>
> http://www.hybsys.bg/img/ipph/IP5000_1.jpg
> http://www.hybsys.bg/img/ipph/IP5000_2.jpg
I don't know who
I wish to set up two PSTN calls and then connect them similar to Jajah (is
this called 3pcc?). The PSTN interconnect is handled by a third party SIP
provider.
I can do this using the manager or call files. An example (using php) would
be:
fputs($oSocket, "Action: login\r\n");
fputs($oSocket,
On Sep 26, 2007, at 10:58 AM, Ricardo Carvalho wrote:
> All phones have firmware version 1.1.1.14; we are testing new
> stable version 1.1.4.18 but by now we found that some phones freeze
> sometimes - version 1.1.1.14 seems more stable.
I'm not sure which firmware I'm running on my GXP2000 (
On Aug 7, 2007, at 1:28 AM, Olivier wrote:
> How can can you stack sangoma cards such as http://www.sangoma.com/
> datasheets/p_a200-specs in a given PC enclosure ?
> It seems to me that it introduces mechanical constraints that seem
> difficult to comply with as space between cards is set by R
On Jul 4, 2007, at 9:59 AM, Steve Kennedy wrote:
>> Oh, so anyway, who was guy "Eng" you named the country after?
>
> And who was America named after ?
Amerigo Vespucci
-chris
___
--Bandwidth and Colocation Provided by http://www.api-digital.c
On May 27, 2007, at 5:29 PM, Paul Aviles wrote:
Is there a way to remote reboot a Cisco 7940 or 7960 phone via some
kind of command? The idea is to force a reboot automatically after
changing one of the configuration files.
As long as you have telnet access turned on in the config file, you
On May 9, 2007, at 3:45 PM, Gavin Henry wrote:
http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-
express-p-393.html
But it will be 3 PCI slots.
Just to clarify in case you didn't already realize it. It doesn't
actually *use* 3 PCI slots, it just occupies the physical space of
On Apr 18, 2007, at 6:50 AM, Rob Hillis wrote:
We've had the very occasional problem with the phone locking up,
but nothing overly serious.
Are you using DHCP on the GXPs that are locking up?
I have one and it would lock up almost every night requiring the
power to be pulled in the morning
On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote:
When a call comes in I want to ring an extension that happens to be
loud speaker. The users can the press *8 to answer the call. Is
there a SIP device that I can connect to Asterisk as an extension
that can accomplish something like
On Mar 15, 2007, at 12:32 AM, shadowym wrote:
Hard to expect the business community to take Asterisk seriously
when this sort of stuff happens IMHO. I can't understand how 3 of
4 hard drives could just suddenly fail simultaneously. There must
be more too it.
It is drifting off topic, bu
On Mar 7, 2007, at 9:58 AM, Steve Totaro wrote:
Might as well since it is free after rebate.
Just as a heads up, that rebate, like most of the others for Vonage
based items, requires Vonage activation in order to actually get the
rebate.
-chris
I'm having a strange issue. My voicemail is working fine, however,
any time I try to access it via one of my analog phones that are
connecting to Asterisk via a Mediatrix 1124... the voicemail system
complains I've entered the wrong password.
There is about a 15 second pause between when I
On Feb 15, 2007, at 1:12 AM, jameson asterisk wrote:
Can anyone provide a recommendation based on user experience?
Feel free to suggest an alternative gateway if one stands out.
I've been working with the Grandstream GXW-4108 (the 8 port version
of the 4108), and it was a rough start, but I
On Jan 26, 2007, at 10:39 AM, Drew Gibson wrote:
You can get the option numbers and values from the source html of
the web page. (I am assuming the GXW-4108 works the same as other
Grandstream products)
I'll try that out, thanks!
I did see a thread on another forum mentioning the HTML sour
On Jan 25, 2007, at 5:38 PM, Leif Neland wrote:
A Grandstream GXW-4108 with 8FXO cost $400, almost the same as a
TDM404B fully populated 4FXO card.
I'm currently testing a GXW-4108... my verdict is still out. I've had
some problems, some minor, some major.
In the minor department, it does
On Jan 10, 2007, at 6:33 PM, M.Hockings wrote:
But it would have been slick to be able to use the old analogue
cell phones as we have several unused here and I am sure they would
be cheap or free to pick up more.
I don't know about in Canada, but around here (NJ, USA), there has
been a co
On Jan 5, 2007, at 12:02 PM, Erick Perez wrote:
When you have a bunch of analog phones that you want to connect to
asterisk, but those analog phones have no transfer button, what are
the options to allow the phones to transfer a call?
You can setup a dial rule to do transfering based on keypre
On Jan 2, 2007, at 10:15 PM, blackwater dev wrote:
I have trixbox working how I want. How do I now (cheaply as
possibly) get a phone number so people can call it from any
number? I am just doing a prototype so just want it done cheaply
so I can demo it to my supervisors.
I just went thr
Has anyone used either the 8 port or 4 port FXO device from
Grandstream? (GXW-4108 or 4104).
They seem to be the lowest cost multi port FXO devices that I can
find, so I'm getting ready to buy the 8 port version. I just want to
see if there are any opinions on the device before I commit to
On Dec 13, 2006, at 7:42 PM, Hadley Rich wrote:
Google does :)
http://www.google.com/search?q=something+site:lists.digium.com
Sweet... I live off of google, and for some reason trying a site
specific search from google just didn't cross my mind.
Thanks!
-chris
Is there a searchable archive of this list? Did I overlook something
obvious? I can find the archives, but short of downloading all the
monthly gzips and building my own searchable database, it seems my
only other option is to go month by month looking at subjects and
hope to stumble on wha
On Dec 11, 2006, at 8:58 PM, Tim Panton wrote:
It looks like there might be enough info on these pages to get you
going:
Thanks for the links! Hopefully I can get somewhere with the info.
If you need a hand with the SNMP side, drop me a mail
I'm pretty new to SNMP, so I may take you u
I recently purchased a Mediatrix 1124 from an auction of a company
that went out of business. It came with nothing other than the unit
itself.
In digging thru the Mediatrix web site, and various google searches,
it looks like it only supports SNMP setup, and only with their
software (or t
Has anyone been able to get Asterisk to work with Verizon's VoiceWing
service? I'm in the process of testing Asterisk to see if it will fit
the needs of my company. Since I already have Verizon's VoiceWing
VoIP service, I figured if I can tie into it, that would let me
evaluate service goin
On Dec 6, 2006, at 8:13 AM, Paul wrote:
Also, I should have mentioned that many of these providers advertise
"business" plans on their website. How can anyone honestly advertise
phone, fax, email hosting, web hosting, etc. to the business community
without 24/7 support?
People should also keep
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