On 24-01-14 00:37, Marek Cervenka wrote:
can someone confirm that mp3 is unsupported? is patch available?
Iirc mp3 was in asterisk-addons in asterisk 1.4 and iirc in later
versions of asterisk you can enable format_mp3 in make menuselect.
what about patch for Opus?
uncle google doesnt
On 16-01-14 21:37, Gergely Kiss wrote:
Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as plan B).
On 17-01-14 01:57, Dan Austin wrote:
Patrick Lists wrote:
On 16-01-14 21:37, Gergely Kiss wrote:
Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol
Hi Steve,
On 15-01-14 18:53, Steve Edwards wrote:
On Wed, 15 Jan 2014, Patrick Lists wrote:
Would you mind sharing where you get the per country IP ranges from?
I confess I 'brute forced' it by entering '/8s' into ARIN's web page and
noting if the block had been assigned to a 'foreign' NIC
Hi Steve,
On 14-01-14 10:39, Steven Howes wrote:
On 14 Jan 2014, at 02:19, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:
Thanks for your feedback Paul. The not having outbound trunks is going to be a
challenge.
Why? it’s what contexts were invented for.
Yes that is indeed what
Hi Steve,
On 15-01-14 02:44, Steve Edwards wrote:
On Tue, 14 Jan 2014, Patrick Lists wrote:
...I guess I'll cook up some dialplan logic that records IP addresses,
keeps track of the amount of failed password attempts etc. and block
the offending IP addresses...
A few iptables rules can
Hi all,
I'm looking into adding the ability to call me at m...@mydomain.org on my
Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow
this kind of access as securely as possible?
Thanks,
Patrick
--
_
--
On 14-01-14 02:36, Paul Belanger wrote:
On Mon, Jan 13, 2014 at 9:24 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
Hi all,
I'm looking into adding the ability to call me at m...@mydomain.org on my
Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow
this kind
On 01/03/2014 03:56 PM, Jonas Kellens wrote:
Hello,
I am getting the following error when compiling dahdi :
[snip]
`/usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux'
make: *** [all] Error 2
I have the right kernel sources installed :
[root@sip dahdi-linux-complete-2.7.0.1+2.7.0.1]#
On 12/15/2013 09:55 PM, CDR wrote:
I have had the issue for years. The problem is that Asterisk
developers are removed from the business. We desperately need simple
way to eliminate transcoding when unnecessary. Transcoding brings a
server to its knees. It is a very simple new setting in
On 12/14/2013 01:29 AM, Martin wrote:
If I need to use SIP, from where to get the suitable firmware for
these Cisco IP Phones 7942G?
Be careful, not all versions of SIP firmware work with asterisk. I do
have 8-3-1 (cmterm-7941_7961-sip.8-3-1)here and it works just fine with
my 7961.
Probably feeding the trolls but here it goes.
On 12/04/2013 04:19 PM, CDR wrote:
Digium is 100% lost in the map. If they would come up with a Paid
version of Asterisk, one that would use the .NET framework in Windows,
something simple to install, they could go public on the product.
IIRC
On 12/03/2013 06:35 PM, Russ Meyerriecks wrote:
This is why we love release candidate feedback! Thanks! I've managed to
mis-tag rc4 and missed all of Oron's commits.
Cutting a v2.7.0.2 and a (correct) v2.8.0-rc5 today.
Thanks. I'll give rc5 a spin when it arrives and report back if I find
On 12/02/2013 04:19 PM, Russ Meyerriecks wrote:
This is fixed on the dahdi-linux master branch and will be included in
the next release:
More info:
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=summary
On 12/02/2013 10:09 PM, Bakko wrote:
Hello,
during DAHDI 2.7.0.1 compilation on CentOS 6.5 64bit, I have this error:
[snip]
This was discussed earlier today and Russ pointed to the fixes:
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=summary
Hi,
I just looked at 2.8.0-rc4 and noticed the udev rules/apps change which
are now supposed to be part dahdi-tools. After make, make install and
make config it seems the dahdi.rules are not installed. I couldn't find
a reference to it in the Makefile either. Did I miss something or has
the move
On 11/26/2013 12:24 AM, Doug Lytle wrote:
Bryant Zimmerman wrote:
Hey all
I believe I found the bug in Asterisk 11.xxx If someone can help me
verify it.
Actually,
I wouldn't consider it a bug. I've know for years that you need to
answer a channel before you play back audio or strange
On 10/28/2013 07:29 PM, Eddie Mikell wrote:
All,
The users in our organization are well, quite frankly, sick of phone
service that is being provided. The choppy phone calls, and drop outs
are detrimental to our sales force.
I've tried about everything I can think of.
Moved the asterisk
On 09/25/2013 09:22 AM, Endri Stefani wrote:
Hi
Greeting to all you out there.
I am new at asterisk, I have been working with PLMN platforms
telecommunication for 5 years with NSN and Huawei.
We have recently built an asterisk PBX with Trixbox and connected it to
our MSS using Digium E1
On 09/25/2013 01:57 PM, Endri Stefani wrote:
Hi Patrick,
If I use latest stable asterisk will I be able to change dialplan by changing
pridialplan in chan_dahdi.conf?
AFAIK yes.
You may also want to check out Asterisk The Definitive Guide (4th
edition is the latest). Paperback version:
Hi Kristian,
On 09/20/2013 03:17 PM, Kristian Kielhofner wrote:
I've been spending some time looking at some of the significant
changes Apple has made to Facetime in iOS 7. I'm far from an Apple
fanboy but some of them are pretty interesting:
- multiplexing everything over a single UDP port
-
On 08/27/2013 08:04 PM, Gergo Csibra wrote:
Hi,
is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card with CAPI in my first
installation (1 BRI), and a hfc 4 port
On 08/19/2013 08:10 PM, Eric Wieling wrote:
One of Asterisk's dirty little secrets is that it does not show the source IP
when a device or hacker tries sending a call without registering. The
rejection message in the logs do not show the IP of the attacker. Yes it
sucks, yes it has been
On 08/19/2013 08:55 PM, Steve Edwards wrote:
On Mon, 19 Aug 2013, Ira wrote:
[2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c:
Failed to authenticate device
390sip:3...@xx.xx.xxx.xxx;tag=2762c06e
xx.xx.xxx.xxx is my public I.P.
What kind of filtering are you doing?
On 08/19/2013 09:29 PM, Eric Wieling wrote:
Actually, you can try enabling the security logging destination in
logger.conf. I believe that may contain the info, but it is new in Asterisk 11. 1.8 and
earlier does not have this.
Thanks I'll give that a try.
Regards,
Patrick
--
On 07/25/2013 11:51 AM, bilal ghayyad wrote:
Hello;
If our Digium Telephony Card does not support echo cancellation like
(1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome
the echo?
Use the free OSLEC echo canceller software module or Digium's commercial
HPEC echo
On 07/18/2013 03:56 PM, jacob.e.mi...@l-3com.com wrote:
I am attempting to setup my server to use Lua for the dialplan
(extentions.lua), but I am unable to get the asterisk configure script
to find the installation of Lua on my box. I have downloaded the Lua
sources from the www.lua.org site,
On 07/10/2013 06:46 PM, Chris Gentle wrote:
[snip]
and then others can connect via SIP. For some reason, when the
speaker says words with S's and F's, they almost sound distorted. Not
quite static but you can tell the quality has been affected. May just
be a side-effect of 8,000 Hz. Just
On 07/08/2013 01:46 PM, Giles Coochey wrote:
Just a note that I did a little work to extend FreePBX distro with some
extra Fail2Ban which deals with some drive-by SIP registration attempts.
My regex is poor to middling, but the steps detailed here:
http://www.coochey.net/?p=61 manage to stop
On 07/06/2013 03:35 PM, Bruce B wrote:
Thanks Patrick.
Do the encrypted config files safe guard against hard resets such as
Web Recovery mode - aka holding down 1 # sign at startup? My
main purpose is to lock the sets due to contract terms so I'd rather not
see user steal the phone and break
On 07/06/2013 08:15 AM, Bruce B wrote:
Hi everyone;
Is it possible to provision lock Aastra phones to provider so that no
soft or hard reset can unlock them?
Iirc you can use encrypted configs using an app called anacrypt and lock
them down. The admin guide (3.2.2) has more details in
On 07/04/2013 05:32 PM, 杨华杰 wrote:
Hi
I just bought some digium analog cards and I would like to build an IVR
system for my customers.
However I am googling and googling , I didn't find any blog and
instruction for beginners like me. So I come here for help. Any tips or
blogs will help.
On 06/11/2013 04:44 PM, Jonas Kellens wrote:
[snip]
Ok thanks.
Any idea how I can resolve this ?
Even if there *can* be more than 1 digit, in case there is only 1 digit
it should go faster.
Would it help if they pressed for example 1 followed by the # key?
If not then, as Eric mentioned,
On 06/09/2013 06:35 PM, Nick Khamis wrote:
Anyone?
Sangoma has a multiplexer:
http://www.sangoma.com/products/stm1mux-fiber-multiplexer/
Which you could then use with:
http://www.sangoma.com/products/a116-16-span-t1e1j1-board/
And there is this card:
On 06/06/2013 05:55 PM, Olivier wrote:
Hi,
I need to rebuild a system which has 4 BRI ports and is connected to
Point-to-multiPoint lines, in a country where telco often drop lines
for energy savings.
I think the dropped D-channel issue should be handled by a very recent
DAHDI. If there are
On 06/03/2013 06:47 PM, Matthew Jordan wrote:
On 06/02/2013 08:36 PM, Patrick Lists wrote:
Hi,
Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I
know for example if a conf ended normally or if someone gave a wrong
conf number or pin?
Thanks,
Patrick
Hi,
The dahdi-firmware package seems to be missing in the asterisk-current
repo on http://packages.asterisk.org
-- Finished Dependency Resolution
Error: Package: dahdi-linux-2.6.2-1_centos6.x86_64 (asterisk-current)
Requires: dahdi-firmware
Can this please be fixed.
Thanks,
Hi,
Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I
know for example if a conf ended normally or if someone gave a wrong
conf number or pin?
Thanks,
Patrick
--
_
-- Bandwidth and Colocation Provided
On 05/13/2013 01:14 PM, Salaheddine Elharit wrote:
hi
You can download a tarball of the release here:
http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz
At least give the link to the latest release which is not 2.6.2-rc1 but
2.6.3-rc1:
Hi Carlos,
On 04/28/2013 10:56 PM, Carlos Alvarez wrote:
We have a new customer with a lot of old phones like the 9133i. They
won't register, and we see some very strange behavior with them. If
the SIP peer exists, they simply fail silently, with no error in the
CLI or the messages log.
On 04/04/2013 09:54 PM, Joseph wrote:
+1.7044972383
If that number is his actual number, maybe create a script that calls
him 10 times an hour, every hour between 00:00 - 07:00am and plays
screaming monkeys every time he picks up (or his voicemail kicks in).
Regards,
Patrick
--
On 04/03/2013 02:48 PM, Marshall Henderson wrote:
Hi Tzafrir-
I know where to find the DAHDI source, but I was more asking where to
actually find which chipsets are supported within the source. Any thoughts?
Have you checked the PCI IDs in the source?
Regards,
Patrick
--
On 04/03/2013 08:34 PM, Marshall Henderson wrote:
Hi Patrick- Yes, I did find the list of PCI IDs (I think). Do these look
right (from wctdm.c):
static DEFINE_PCI_DEVICE_TABLE(wctdm_pci_tbl) = {
{ 0xe159, 0x0001, 0xa159, PCI_ANY_ID, 0, 0, (unsigned long)
wctdm },
{ 0xe159,
On 03/14/2013 11:04 PM, mohsen feyzzadeh wrote:
Hi all
I installed
DAHDI Version - 2.6.1
DAHDI Tools Version - 2.6.1
libss7-trunk
Asterisk 11.0.1
from source on Fedora 12 x86_64.
In case the 12 in Fedora 12 was not a typo, you do realize that Fedora
12 has been end-of-line for years and has
On 03/11/2013 04:18 AM, bilal ghayyad wrote:
I am not mixing. I need this for LAB testing.
How? This PCI passthrough, how to enable it on virualbox?
It's in the VirtualBox manual.
Regards,
Patrick
--
_
-- Bandwidth and
On 03/11/2013 12:53 PM, termo termosel wrote:
Hi,
I have Ubuntu and Asterisk 11.2.1 in a boot USB. When I put it in
desktop computer, asterisk starts without problem but if I insert the
same USB in a laptop computer Asterisk doesn't start. Is it possible
because different microprocessors?
On 03/11/2013 07:07 PM, Asghar Mohammad wrote:
HI Bilal,
i am using chan_mobile for call termination, you can use it but you need
to tweak chan_mobile.c it is broken from a long time.
let me know if you want give it a try.
If you could send the patches you made to chan_mobile to this mailing
On 03/12/2013 12:07 AM, Andrew Yager wrote:
Hi,
I'm trying to find (with some desperation now) a decent web based or
application based UI that integrates with an Asterisk based PBX and is
designed for a Serviced Office environment.
Key features we're looking for:
Don't know if it covers your
On 02/08/2013 06:35 AM, Ding Peng wrote:
Hi, everybody,
Where can I get the manual or user guide of latest asterisk version,
1.11.x?
I want to know the syntax and usage of all the supported functions or
something like that in the latest version.
You can find one on the O'Reilly website.
On 01/24/2013 10:37 AM, Zyumbilev, Peter wrote:
Hello,
I need to setup system of aroud 60 DECT phones with asterisk.
So far I found
http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710
However is there some cheap base station(similar to GSM cell) so
On 01/24/2013 09:44 PM, Richard Kenner wrote:
[snip]
When I use alaw, the path from Asterisk to the Alcatel is completely
clean, but the other way has a set of clicks that kind of sound like
old-fashioned audio noise.
[snip]
It's been ages since I experienced that but things to check that come
On 01/24/2013 11:57 PM, Richard Kenner wrote:
- jitterbuffer settings (try on/off)
I added
jbenable=yes
and get lots of:
[Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put:
DAHDI/i1/2128518396-6c7 received frame with invalid timing info:
has_timing_info=1, len=0,
On 01/22/2013 08:54 AM, Sakharam Thorat wrote:
Can anybody send me Detailed process to configure Asterisk in CENTOS ??
Detailed description highly appreciated.
Start by reading the Asterisk book, check asterisk.org and Google around
to see if your question has already been answered.
On 01/17/2013 09:05 PM, Joe Ruffolo wrote:
Hi all! In need of some serious help. We currently run Trixbox and Cent
Os on a 2u server for our small business’s phones system.
Afaik Trixbox is no longer maintained and their forum are hardly active
anymore so it may be a bit of a challenge to get
On 01/02/2013 07:11 PM, Carlos Alvarez wrote:
The number of questions posted here that are easily answered with a
search or which are far too basic and open (how do I make Asterisk work)
is very high these days, and that does kill a list. A lot of us are
interested in helping people who help
On 01/02/2013 06:20 PM, Steve Totaro wrote:
I became a list member way before any such rule and never had to click
through and agree to these update ToS.
I am grandfathered in.
Just looked it up. I see my first post back in April 2003, yours in
September 2003 and Jon in March 2003. Wow you
On 01/02/2013 09:46 PM, jon pounder wrote:
On 01/02/2013 03:22 PM, Patrick Lists wrote:
On 01/02/2013 06:20 PM, Steve Totaro wrote:
I became a list member way before any such rule and never had to click
through and agree to these update ToS.
I am grandfathered in.
Just looked it up. I see
On 12/30/2012 04:26 PM, Ron Wheeler wrote:
I participate in a lot of lists and top posting is now the norm since
people want to see quickly if the message is worth reading.
Isn't it a bit of a stretch to extrapolate your experience with your
lists to top posting being the norm? I am
On 11/13/2012 12:11 AM, Phil Reynolds wrote:
[snip]
It turns out to be a known issue:
https://issues.asterisk.org/jira/browse/ASTERISK-19532
... and can be fixed by applying the patch at:
https://issues.asterisk.org/jira/secure/attachment/43441/xmpp_no_crash_with_ejabberd.patch
I will file
On 11/13/2012 07:05 PM, Michael L. Young wrote:
[snip]
Is it an omission that this fix has not been applied to the 11 tree?
From the looks of ASTERISK-19532 it seems that the fix has only been
applied to 1.8 and 10.
If you click on the link for ASTERISK-19532, there is a tab in the Activity
On 10/16/2012 08:50 AM, Sebastian Arcus wrote:
I've just bought an OpenVOX B200p ISDN card - and if I remember
correctly from last time I used one of these - it is possible to use
either DAHDI or mISDN with it. Are there any factors to consider when
choosing which software to use? Is one better
On 10/15/2012 09:07 AM, sudeep melekar wrote:
[snip]
i m completely new to asterisk
so any help would be appreciated
If you are totally new to Asterisk I recommend you first read the
Asterisk book and go through the wiki. Both have sections how to install
the various Asterisk components.
On 10/12/2012 11:17 PM, Philip Bennefall wrote:
From what I gather, it costs extra for each channel even for direct
Skype to Asterisk calls. Since my plan was to use this for business
purposes, I'd need at least something like 30 channels which would be
way out of my monthly budget
On 10/04/2012 10:00 PM, Phil Daws wrote:
Hello:
I am investigating the possibility of using LDAP for storing certain Asterisk
configuration parameters.
I have examined res_ldap.conf and see where mailbox can be defined from
AstAccountMailbox but I do not see where the password can be stored
On 10/05/2012 11:51 AM, Shanavaz E A wrote:
Hi,
No replies until now. Some one please help... There must be some people
who are using it...
Thanks
No idea but since Asterisk is making you money why don't you hire an
experienced Asterisk consultant to get it resolved.
Regards,
Patrick
--
On 10/05/2012 02:10 PM, Benoit Panizzon wrote:
Hello
We had this situation:
Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk
Server was abused to call a large number of expensive destinations.
I'm sorry to hear that. In the Asterisk source there is a doc that
On 09/27/2012 08:15 AM, Shanavaz E A wrote:
[snip]
Patrick, can you please give the steps to configure fax with iaxmodem
and hylafax. Is it free to use?
It's been years since I set it up so I don't know exactly how to
configure it anymore. But I do remember that I found some howto/docs via
On 09/28/2012 03:01 AM, Patrick Archibald wrote:
Hi,
Is there a way to move 100 .call files in to
/var/spool/asterisk/outgoing/ at once and have Asterisk call at
maximum 10 at a time?
Afaik that is not possible. Wouldn't it make more sense to move call
files in batches of 10 to outgoing/?
On 09/26/2012 05:53 PM, Mark Robinson wrote:
Hello.
I have asterisk 1.8.18 which connects to ISDN PRI. All phones are sip,
Aastra 6757i. Everything works as expected.
We also have a FAX machine. We need to be able to use that FAX machine
to send or receive faxes. We are planning to have a
On 09/25/2012 11:18 PM, Logan Bibby wrote:
MyISAM would be best, in my opinion. The features that cause the little
bit of performance overhead in InnoDB wouldn't be necessary for CDR storage.
Iirc InnoDB is ACID compliant so might be preferable if MyISAM is not.
More information here:
On 09/14/2012 05:26 AM, Raj Mathur (राज माथुर) wrote:
[snip]
Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai.
Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
Your DAHDI and Asterisk versions are old so for starters I would update
everything to the latest
On 09/14/2012 10:25 AM, A J Stiles wrote:
[snip]
It could be nothing more than a dry solder joint on one of the RJ45s. For the
sake of five minutes' work with a soldering iron, that's got to be worth
eliminating.
Wouldn't that void your warranty? I would take it up with Digium support
and
On 01-09-12 04:14, Vladimir Mikhelson wrote:
[snip]
* Ability to send host name or other CN not equal to the phone IP in
TLS negotiation
Afaik you usually put alternative CNs in SubjectAltName in the
certificate. Have you tried that?
Regards,
Patrick
--
On 08/30/2012 09:45 AM, Gopalakrishnan N wrote:
Hi,
I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host,
I am not using any virtualbox, still i struck in loading the modules.
Please do not top post.
Install strace and then start asterisk with the command:
# strace asterisk
On 27-08-12 08:25, Gopalakrishnan N wrote:
This is really tuff working with OpenSuse. I am clueless how to sort our
this.
Maybe switch to a different distribution? I have used CentOS and RHEL
for years without any problems and as far as I know both debian and
ubuntu should work without
On 27-08-12 14:08, DHAVAL INDRODIYA wrote:
Hi All,
i would like to know if anyone has done or having idea regarding PRI
terminations in asterisk.
i have a requirement where i need to support 80 PRI in one machine i
have found a machine which have 10 PCI slots available
now i am thinking of
On 25-08-12 14:31, Stefan at WPF wrote:
Hello all,
I need some help understand the values of the CHANNEL function, e.g.
txploss // local packets loss
rxploss // remote packets loss
txjitter // local jitter
rxjitter // remote jitter
My main problem in understand is that a
Hi Hans,
On 24-08-12 10:13, Hans Witvliet wrote:
Hi all,
After making a nice demo-setup for one of our innivationmanagers, he
came up with a completely different stratagy ;-(
Well if you could create it then obviously it's no longer innovative so
they had to come up with something else :-)
On 22-08-12 20:04, Giuseppe Longo wrote:
Just a little questions, what's the difference between asterisk 1.8
and asterisk 11?
Iirc you can check the ChangeLog in the Asterisk 11 sources.
Regards,
Patrick
--
_
-- Bandwidth
On 14-08-12 08:29, Gopalakrishnan N wrote:
If I change autoload=no then asterisk is starting, but post to that
loading modules even chan_sip.so asterisk hangs. Its strange, only in
OpenSuse I am facing this. In CentOS, Ubuntu its working fine, same
Asterisk version with same hardware.
Please
On 10-08-12 10:12, SamyGo wrote:
Oh, I see - check if your country blocks the SIP port 5060 ? try
changing the default poert from 5060 to something else like and
then try this.
I think your ISP is blocking the SIP.
If that is the case, setup an IAX connection and see if that works.
On 26-07-12 12:40, Chris Bagnall wrote:
On 26/7/12 11:08 am, Ishfaq Malik wrote:
I'm thinking about deploying TTS onto our asterisk servers and was just
wondering which ones people use and like...
We've tried Festival, Cepstral and Ivona.
I didn't know about Ivona so thanks for mentioning
On 20-07-12 09:15, neo nortan wrote:
dear
i am a neewbi for asterisk, plz tell me or if any link is there where i
can understand how asterisk, freepbx, web-meetme, dahdi all these tools
works and how they are related.
plz help me.
http://www.asteriskdocs.org/
Regards,
Patrick
--
Hi,
The flowroute website mentions that they set callerid on outbound calls
based on the presence of (in order of preference):
P-Asserted-Identity, Remote-Party-ID or From:.
I've been trying to make outbound callerid work via flowroute to no
avail. Does anyone have an extensions.conf /
On 10-07-12 18:29, Alex Balashov wrote:
SIPAddHeader() comes to mind. :-)
Yup I got that far :) I tried things like (with correct name number):
exten = _1ZX,1,SipAddHeader(P-Asserted-Identity: Global
Minties Corp sip:19995551212@AST_BOX_FQDN)
But that did not work as flowroute
On 10-07-12 18:49, Warren Selby wrote:
You can't* set the outbound name. That's defined in the national caller
id name database that the receiving phone company dips into. As far as
I know, Flowroute does not add entries to this database, nor do they dip
it when you receive a call to pass the
On 10-07-12 18:48, Danny Nicholas wrote:
Check your users.conf - this looks like an override issue to me.
Thank you for your feedback Danny. users.conf is default and has not
been touched.
Regards,
Patrick
--
_
--
On 10-07-12 18:47, Ira wrote:
At 09:20 AM 7/10/2012, you wrote:
I've been trying to make outbound callerid work via flowroute to no
avail. Does anyone have an extensions.conf / sip.conf snippet howto
make this work? This is for Asterisk 1.4.44.
This is a section of code I use to choose
On 10-07-12 19:48, Warren Selby wrote:
On Tue, Jul 10, 2012 at 12:34 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl
mailto:asterisk-l...@puzzled.xs4all.nl wrote:
Thank you for your feedback Warren. I removed the outbound name but
still get random numbers VOIP CALLER on outbound
On 10-07-12 20:42, Carlos Alvarez wrote:
I'm currently trying to decide on which GUI-enabled version of Asterisk
to use for one particular installation, where we will need good
telecommuter support. We've made it so easy for people to work remotely
that the customer is downsizing their real
On 09-07-12 12:00, Chandrakant Solanki wrote:
Hi
@Patrick, are you using which AMR source, will you please provide me
link, I also tried with 1.8.11 but didn't found success.
I am using sourceforge one
http://sourceforge.net/projects/aterisk-amr/files/
I used the one from sourceforge:
On 09-07-12 07:42, Andrew Colin wrote:
Hi Patrick
Can you possibly guide me as to how you got it working.
I am running the same version of Centos and asterisk.
Did you use a specific kernel?
I did not use a specific kernel version. I got the latest versions from
misdn.eu and installed those.
On 03-07-12 09:38, Andrew Colin wrote:
Hi Guys
Has anyone got this working on Centos 6?
I set it up on a CentOS 6.2 x86_64 box with Asterisk 1.8.11 and the
latest mISDN/mISDNuser/isdn4k-utils/lcr stuff and made some test calls
between a Polycom 650 and a GSM calling into the ISDN line
On 06-06-12 11:41, Thorsten Göllner wrote:
Where can I find such ip-lists, please?
http://www.ipdeny.com/
Regards,
Patrick
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Hi zoa,
On 31-05-12 17:39, joachim wrote:
Ellow,
We released zoiper for Android today, available for free here:
https://play.google.com/store/apps/details?id=com.zoiper.android.app
SIP and IAX is supported, should work quite well, unfortunately it is
really hard to test all android and
Hi Khalid,
On 18-05-12 20:50, khalid touati wrote:
Hi Patrick,
it seems like you have the magic ball, I think what you described is
exactly what happened:
After we tested the server+ link and we were able to have simultaneous
calls (as expected), and knowing that this server was not touched
On 16-05-12 17:10, gincantalupo wrote:
Hi Larry,
thank you for your answer.
This is same test I did. After this I lowered again to 4800...result:
iaxmodem receives at 9600 b/s.that's why I cannot solve the puzzle.
I put that line (Class1RMQueryCmd: !24,48,72) in config.IAXtty
On 05/13/2012 03:07 AM, khalid touati wrote:
My Issie is finally fixed and I can make calls, I received actually from
digium the fix, I'll try to give as much details as I can to make sure
people who find this thread understand pb and solution.
Problem: not able to dial calls using BRI from
On 11-05-12 05:44, khalid touati wrote:
Thank you for your reply Patrick!
for the first situation, I did try asterisk 1.6.2.6 and dahdi 2.3 but
with no success.
Can anyone suggest a combination that works till a patch is released?
Unfortunately not and I don't have a Digium BRI card to test
On 10-05-12 21:10, khalid touati wrote:
Hi All,
I did downgrade to asterisk 1.6.2.6 and dahdi 2.3 which is working with
other BRI line (in our NL office), but I get this type of errores:
-- Called G1/0788744550
[May 10 20:07:20] ERROR[27479]: chan_dahdi.c:12280 dahdi_pri_error: ACK
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