Hi All,
I am having some problems with Asterisk on static IP and Sipura-1001 on
dynamic IP. Is there any solutions to in the Asterisk configuration or
Sipura-1001 to re-register when the router change IP dynamic IP? Thanks.
Regards,
Kengie
___
--
On Fri, Jun 12, 2009 at 8:51 AM, Kengie Hokengiepa...@gmail.com wrote:
I am having some problems with Asterisk on static IP and Sipura-1001 on
dynamic IP. Is there any solutions to in the Asterisk configuration or
Sipura-1001 to re-register when the router change IP dynamic IP? Thanks.
If
Hi
My production Asterisk has these frequent errors when connected to the
ZTE type exchange...
Jun 9 10:48:24 NOTICE[1984] chan_zap.c: PRI got event: HDLC Overrun (7)
on Primary D-channel of span 1
followed by..
Jun 9 11:46:42 WARNING[1984] chan_zap.c: No D-channels available!
Using
2009/6/11 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new
Aastra SIP phones can be auto-provisioned when config files are stored in
a
specific TFTP subdirectory instead of TFTP root directory.
For
I am running Asterisk 1.4.21.2
For reception, I defined a simple queue with one SIP phone as the only
member.
When I receive an incoming call, I test QUEUE_WAITING_COUNT to see if it
is 0.
If it is 0, then I will playback a message to tell the caller to be
patient and then do a
Hello,
I was testing my asterisk for a while with 1.6 without much problem.
Now i am trying to install a new system with asterisk 1.4 but now i am
using a dual pri card instead of single pri.(TE220P)
What i want is to use both PRI ports as group.
Now i have zaptel.conf file created as follows
Hi all,
In about 4 hours from this writing, the G.722 conference bridge will
be brought up, the Talkshoe G.711 also, so you can call in via SIP,
PSTN or Skype (experimental)
http://vuc.me for all the gritty details
IRC #voip-users-conference anytime today
We'll also be talking about hosted PBX
On Fri, Jun 12, 2009 at 11:49:02AM +0300, Oguzhan Kayhan wrote:
Hello,
I was testing my asterisk for a while with 1.6 without much problem.
Now i am trying to install a new system with asterisk 1.4 but now i am
using a dual pri card instead of single pri.(TE220P)
What i want is to use
On Fri, Jun 12, 2009 at 8:13 AM, Tzafrir Cohentzafrir.co...@xorcom.com wrote:
On Fri, Jun 12, 2009 at 11:49:02AM +0300, Oguzhan Kayhan wrote:
Hello,
I was testing my asterisk for a while with 1.6 without much problem.
Now i am trying to install a new system with asterisk 1.4 but now i am
Hi nik600,
I had some trouble transferring calls with that version of Asterisk even
if I used the normal transfer via features.conf. Upgrading to 1.4.24
helped a bit (even if not completely). My advice is to upgrade to 1.4.24
or the latest.
Giorgio
nik600 wrote:
I'm experiencing some
On Fri, Jun 12, 2009 at 8:43 AM, Christopher Stamper
christopherstam...@gmail.com wrote:
On Thu, Jun 11, 2009 at 2:00 PM, Danny Nicholas da...@debsinc.com wrote:
Nerdvittles.com has a nice example of this, when they are up. They
used it for Phone trees for a school or something like
On Thu, Jun 11, 2009 at 2:00 PM, Danny Nicholas da...@debsinc.com wrote:
Nerdvittles.com has a nice example of this, when they are up. They used
it for Phone trees for a school or something like that. Took less than 30
minutes to put in my dialplan and use
Sounds like exactly what I am
According to app_dial.c these are the present values (1.4.25-rc2; I assume
these are the same for 1.6.1.1)
DIALSTATUS - This is the status of the call:\n
CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER |
CANCEL\n
DONTCALL | TORTURE |
Hi
I am in the process of installing a new box and using dahdi. I have a
tdm410 + hardware echo canceller.
I have just read in the read me for dadhi that VPMADT032 support has
been removed and unlike with the zaptel stuff i could just download and
install the firmware I can't with dahdi
what
Danny Nicholas schrieb:
According to app_dial.c these are the present values (1.4.25-rc2; I assume
these are the same for 1.6.1.1)
DIALSTATUS - This is the status of the call:\n
CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER |
CANCEL\n
On Fri, 2009-06-12 at 23:58 +1000, Alex Samad wrote:
what is the best way forward to recompile with hardware echo canceller
support.
No need to do anything special during compilation. For hardware echo
cancellation just put the option echocancel=yes in chan_dahdi.conf
Kengie Ho schrieb:
I am having some problems with Asterisk on static IP and Sipura-1001 on
dynamic IP. Is there any solutions to in the Asterisk configuration or
Sipura-1001 to re-register when the router change IP dynamic IP?
Whenever the ATA gets a different IP address from a DHCP server
On Fri, Jun 12, 2009 at 11:58:51PM +1000, Alex Samad wrote:
Hi
I am in the process of installing a new box and using dahdi. I have a
tdm410 + hardware echo canceller.
I have just read in the read me for dadhi that VPMADT032 support has
been removed and unlike with the zaptel stuff i
Thanks for the reply.
I may be mistaken for assuming that 'disposition' values in CDR were
actually from DIALSTATUS. My CDR table has entries for 'disposition'
including ANSWERED, FAILED, NO ANSWER.
JR
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi all,
I`m searching for a special solution to send text messages inside Sip info
packets, that are normally used for dtmf signalization. So far I’m able to
exchange sip Info messages between two softphones which are connected directly
together (only over a Switch).
By connecting both
IMO, the disposition in the CDR is set at hangup/fail time, not dial time.
I'm sure many others can elaborate.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal
Sent: Friday, June 12, 2009 10:03 AM
To:
John Regal schrieb:
I may be mistaken for assuming that 'disposition' values in CDR were
actually from DIALSTATUS. My CDR table has entries for 'disposition'
including ANSWERED, FAILED, NO ANSWER.
True.
and BUSY.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hello all, I have a TC400B Digium card in order to deal with transcoding and
I have some trouble using it, I have a timer synchronisation problem!
I would be very grateful if you have any idea to help me?
It seems that the card is not correctly synchronised to the system because
when I speak to
Asterisk as of this writing doesn't really support text messaging except for
asterisk-to-phone during a call. Some folks on this thread use other
programs in conjunction with asterisk to support this capability.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hey Everyone, once again - last time to publish this..
i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM
300,320,360 Devices.
In the combination with asterisk and the patton, there are occuring some
strange behaviour, due to the calling and answering everything works
good,
I mean..making a single trunk between a pstn or telco with 2 or more PRI's..
I mean instead of using 32 channels to use 64 or more..
I am trying to increase the capacity between my PSTN and asterisk actually.
There will be more than 35-40 concurrent calls so while creating a zap
trunk(or
Stefan Agethen schrieb:
Hey Everyone, once again - last time to publish this..
Hey, you posted this on Jun 8, 10 and 12.
i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM
300,320,360 Devices.
Which firmware version is on the phones?
I can't remember having seen any
JFYI and slightly off-topic:
All of the videos we recorded at AMOOCON open-source VoIP conference
(Rostock, Germany, May 4-5) are now available on the web site:
http://www.amoocon.com/
All of them are available in different qualities and formats,
including Quicktime 7, versions for the iPhone
Oguzhan Kayhan escribió:
I mean..making a single trunk between a pstn or telco with 2 or more PRI's..
I mean instead of using 32 channels to use 64 or more..
I am trying to increase the capacity between my PSTN and asterisk actually.
There will be more than 35-40 concurrent calls so while
On Fri, Jun 12, 2009 at 05:40:16PM +0300, Tzafrir Cohen wrote:
On Fri, Jun 12, 2009 at 11:58:51PM +1000, Alex Samad wrote:
Hi
I am in the process of installing a new box and using dahdi. I have a
tdm410 + hardware echo canceller.
I have just read in the read me for dadhi that
No divx hd? just kidding
OT: Odd how many video/audio standards there are, and the growing issue with
them? I recall when you had two choice Windows Media or RealPlayer. Now I have
to make 3-4 for everything from DivX to iPod to Walkman. For example my cell
phone can't play a H264/AAC due the
2009/6/12 Miguel Molina mmol...@millenium.com.co:
Oguzhan Kayhan escribió:
I mean..making a single trunk between a pstn or telco with 2 or more
PRI's..
I mean instead of using 32 channels to use 64 or more..
I am trying to increase the capacity between my PSTN and asterisk
actually.
There
Jason Aarons (US) schrieb:
OT: Odd how many video/audio standards there are, and the growing issue with
them? I recall when you had two choice Windows Media or RealPlayer.
There is only one format[1] of choice: .mov :-)
It's amazing how formats natively supported by QuickTime play
smoothly
I just wish my HTC Touch Pro cell phone or my PlayStation3 could play .mov
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen
Sent: Friday, June 12, 2009 5:40 PM
To: Asterisk Users Mailing List -
On Sat, Jun 13, 2009 at 06:57:11AM +1000, Alex Samad wrote:
any chance of getting digium to host a digium debian repo (sort of how
virtulbox doit), that way they could have a fully build package ?
Or resolve the issues that made this patch necessary in the first place.
--
On Sat, Jun 13, 2009 at 01:40:48AM +0300, Tzafrir Cohen wrote:
On Sat, Jun 13, 2009 at 06:57:11AM +1000, Alex Samad wrote:
any chance of getting digium to host a digium debian repo (sort of how
virtulbox doit), that way they could have a fully build package ?
Or resolve the issues that
Hi All;
Did anyone tried Fedora Core 10 with Asterisk 1.4.25.1? I am facing a problem
that it is not able to detect the g729 (although I have another machines
running fedora core 9 and it is fine).
Any help?
Regards
Bilal
___
-- Bandwidth
I'm not sure if the kernel timing HZ has anything to still do with
things anymore. You might need to recompile your kernel with HZ=1000
-Jon
lf...@leurent.eu wrote:
Hello all, I have a TC400B Digium card in order to deal with
transcoding and I have some trouble using it, I have a timer
For our internal fax machines, I'm checking if the faxes are going to
branch offices. If they are, I want to capture and email them to the
branches. I've set up extension 8447 to test this.
A fax machines is connected via an SPA 2102 on 173. Any calls from 173
are sent to:
[outbound-fax]
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