I know I should probably post this to the Trixbox forums, but am hoping
someone might have a quick answer here for me. Client with Trixbox 2.6.2
with a recompiled asterisk 1.4.23 and various patches for RPID. A meetme
conference with several people was in progress when the UPS died and the
ma
On Mon, 18 May 2009, Jeff LaCoursiere wrote:
>
> I know I should probably post this to the Trixbox forums, but am hoping
> someone might have a quick answer here for me. Client with Trixbox 2.6.2
> with a recompiled asterisk 1.4.23 and various patches for RPID. A meetme
>
Not the business - the list (is non commercial). Meaning if you want to
advertise your cool new service, do it on asterisk-biz. He knew that for
sure.
j
On Mon, 18 May 2009, Martin wrote:
What kind of business is non-commercial ? Please be so kind to explain
that to me ...
Martin
On
On Tue, 19 May 2009, Tim Nelson wrote:
Not all 18xx are to go out the analog lines on g1. Only, 1866 and 1800.
The second part makes sense. Now do the two groups below continue to go
into their own contexts and one included by the other or should they
both go into the same context?
You wo
On Tue, 19 May 2009, Jason Aarons (US) wrote:
> Can anyone recommend a codec pack with G723 for use under Vista? I have
> G723 prompts (about 70 prompts totaling 1MB) needing to be converted to
> G711 uLaw.
>
If the G.723 prompts are already on your asterisk server, just use
asterisk to conver
On Wed, 20 May 2009, John Regal wrote:
> Hi,
> I am attempting to make about ten calls simultaneously and intermittently
> get 'SIP/voipprovider is circuit-busy' followed by 'everyone is
> busy/congested at this time"
>
> I am not sure if this is related to my bandwidth to my voip provider, a
> c
On Wed, 20 May 2009, John Regal wrote:
> Thanks for the reply and apologize for the double post. My original post
> landed in another thread and thought it may have been missed...
>
> I questioned my voip provider before posting and they told me they have
> other asterisk customers that are maki
So you were fourteen and a military engineer?
j
On Wed, 20 May 2009, ContactTel Business wrote:
> Many years in telecom and computer world is around 100 year in real life..
> 10 years ago i was a millionaire in the dot com boom and 24 years old with a
> P2 300 computer.., 20 years ago i was mi
Lets start from the beginning. Why are using a network share for your
voicemail in the first place?
j
On Fri, 22 May 2009, Elliot Murdock wrote:
> Hello Matt,
>
> I do agree with you that NFS is that UNIX standard for network
> filesystems and that what should essentially be used. However, I
On Tue, 26 May 2009, Steve Howes wrote:
> On 26 May 2009, at 16:39, Jeff LaCoursiere wrote:
>> YMMV
>
> I think thats the problem :D sorry couldn't resist..
>
I did kind of mean that tounge-in-cheek :):)
j
___
-- Bandwidth
On Tue, 26 May 2009, randulo wrote:
> On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas wrote:
>> I run my analog telco over cat5, but that's in-house and definitely not 3km.
>> That sounds really far for current loop stuff.
>
> I was doing that too. I asked this same question a few years ago and
st things.
j
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: Tuesday, May 26, 2009 10:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: bal
On Tue, 26 May 2009, Danny Nicholas wrote:
> The best a native cat5 can run is 100 meters. Unless you like paying your
> telco huge bucks, you should go for some kind of SIP connection to your box.
>
He was asking about an analog telco connection - not an ethernet drop.
j
>
>
> _
>
> Fr
On Wed, 3 Jun 2009, Rob Hillis wrote:
> Alex Samad wrote:
>> I have been looking at a snom 300, which seems okay. the display goes a
>> bit haywire occasionally - not sure why yet.
>>
>> Are the 320 worth the extra money ?
>
> IMO yes, though it really depends on what you want from the phone.
>
>
On Thu, 4 Jun 2009, Rob Hillis wrote:
> Jeff LaCoursiere wrote:
>> We are still talking about a $175 phone. How about the Polycom IP 320?
>> $85 at 888voipstore. Can't go wrong with Polycom for voice quality.
>>
>
> True, Polycom's are brilliant for
On Wed, 3 Jun 2009, Tzafrir Cohen wrote:
> On Tue, Jun 02, 2009 at 04:40:53PM -0500, Erick Perez wrote:
>> I totally agree with you Jeff, however some of us do not actually sell
>> viagra over the phone.
>> This is a campaign to spread a message to the population about the health
>> prevention st
On Wed, 3 Jun 2009, Jim Dickenson wrote:
> I think it is a DID trunk. I am having problems getting the clients telco to
> tell me much about the T1. For sure 24 analog channels in a single T1.
>
> Would I be able to use this type of T1 with a Sangoma A102de?
> --
> Jim Dickenson
> mailto:dicken.
On Thu, 4 Jun 2009, Philipp Kempgen wrote:
> Joseph L. Casale schrieb:
>> I have a single server running asterisk 1.6.0.8 with a few sip voip providers
>> and a tdm card for redundancy. It has a caching name server and the sip
>> providers
>> are hard coded in the hosts file.
>>
>> When the inte
On Thu, 4 Jun 2009, Joseph L. Casale wrote:
>>> A persistent local DNS cache such as pdnsd[1] or djbdns[2] could help.
>>>
>>> [1] http://en.wikipedia.org/wiki/Pdnsd
>>> [2] http://en.wikipedia.org/wiki/Djbdns
>>>
>>>Philipp Kempgen
>>
>> I am guessing it fails to reverse lookup your internal
On Thu, 4 Jun 2009, Danny Nicholas wrote:
> Do you want a "live repository" or just a common gathering of the data? If
> LR then you should set up a deamon on each box to transfer records as they
> occur using something like the DBI functionality of PERL. If not, then just
> do a mysql dump pe
On Thu, 4 Jun 2009, Deepak wrote:
> Hi, we are experiencing a strange issue and I am hoping someone can point me
> to the right direction or help out with some pointers.
>
> We have asterisk 1.6.0.6 with a sangome a104DE card. We have basically 4
> T1's for a total of DAHDI 96 channels.
>
> We ha
On Thu, 4 Jun 2009, Deepak wrote:
> BTW,we are using an ODBC connection to Microsoft SQL Server.
> We are not using MySQL.
>
> Would that be a possible cause?
>
Good lord who designed this mess? Start over.
j
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-- Bandwidth and Colocation Provided
On Thu, 4 Jun 2009, Deepak wrote:
> It was not a conscious decision to use MS-SQL. We were "forced" to use
> MS-SQL since the "business" rules were on MS-SQL and other apps are using
> it.
>
> If you think it is not ODBC, then what in your opinion is causing the issue?
> I though you did mention
On Thu, 4 Jun 2009, Peder wrote:
> Is there a limitation to the number of variables you can set from a PHP agi
> script? I have a simple example and I can't get it to let me set more than
> 1. I am pretty sure I am just missing something, but I've searched all over
> an can't find the answer.
On Fri, 5 Jun 2009, Peder wrote:
> I've tried all of that and it still doesn't work right. I'm sure it's
> something dumb, but I just can't figure it out. I've even made it simpler:
>
> echo 'SET VARIABLE ISLOCALCONTEXT CONTEXT3\n';
> echo 'SET VARIABLE ISLOCALDID \n';
>
> and this produce
On Sat, 6 Jun 2009, Ayman Hendawy wrote:
> Dear sir
> I build a daughter card to interface the FXO module with blackfin537 stamp
> board, but unfortunately I can`t get Dial tone or any other signalling via
> fxo port.however the fxo module has been detected on the board.
> my daughter card work w
On Mon, 8 Jun 2009, David Backeberg wrote:
> On Mon, Jun 8, 2009 at 10:51 AM, equis software
> wrote:
>> Witch festival version are you talking about?
>>
>>
>> I need spanish(argentinian) voice...
>
> I don't know whether any free programs do spanish TTS. I can tell you that
> AT&T Natural voice
Has anyone set it up so that an inside call and an outside call get
different unavailable messages?
j
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To UNSUBSCRIBE or update options visit:
htt
On Tue, 9 Jun 2009, Philipp Kempgen wrote:
> Jeff LaCoursiere schrieb:
>> Has anyone set it up so that an inside call and an outside call get
>> different unavailable messages?
>
> Yes.
> AGI() -- optional
> if(...){...}else{...}
>
I appreciate the sarcasm - I su
On Tue, 9 Jun 2009, Steve Repo wrote:
On Tue, Jun 9, 2009 at 4:50 PM, Philipp
Kempgen wrote:
Thomas Kenyon schrieb:
Peder wrote:
Decent product, but their support and development are horrible. I showed
them that their SIP over TCP implementation was broken and their reply was
"use udp"
Had a fairly horrible lightning storm night before last, and four of eight
ports in a 1.4.20 machine stopped answering.
In the CLI:
budsw*CLI> zap show channels
Chan Extension Context Language MOH Interpret
pseudodefault en default
1
On Sat, 13 Jun 2009, Jim Gottlieb wrote:
> I'm evaluating using Polycom phones for our call center and I've set
> up my first phone (a SoundPoint 560) to give it a try.
>
> The phone is working and can successfully place and receive calls.
> But every minute, there's an error in the log file:
>
>
On Mon, 15 Jun 2009, Jim Gottlieb wrote:
> On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote:
>
>>> [hft0]
>>> type=friend
>>> username=hft0
>>> secret=mysecret
>>> context=outtrunk-office
>>> host=192.168.200.99
>>
On Sat, 20 Jun 2009, Erick Perez wrote:
> Jeff, indeed i was posting for posterity. Maybe someone will benefit in an
> outbound-only scenario that he/she will not need a supercomputer to pump a
> 20sec audio clip.
> Again, this was a public service. And indeed TV and radio was used. Unless
> you
On Wed, 24 Jun 2009, Grygoriy Dobrovolskyy wrote:
> 2009/6/24 Senad Jordanovic
>
>> Jay Fenton wrote:
>>> [ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ]
>>>
>>> Howler Technologies are proud to announce today the launch of
>>> their fully indemnified and highly optimised G.729A solution
On Thu, 25 Jun 2009, jonas kellens wrote:
> I feel a great preference for sticking to manually editing
> the .conf-files.
Then why did you ask for a GUI?
> But if I define in the contract that changes to the
> Asterisk-PBX need to be done by me, I force a maintenance cost towards
> the custome
On Fri, 26 Jun 2009, David Backeberg wrote:
On Fri, Jun 26, 2009 at 12:03 PM, drew einhorn wrote:
Have a bunch of old clunker boxes on the scrap heap that would
probably do the job as a server. But I'm not confident
they would be reliable enough.
Capable of supporting maximum of a dozen 2-li
On Sun, 28 Jun 2009, randulo wrote:
> On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote:
>> Hi,
>>
>> Has anyone tried it ?
>> Is there any available pricelist ?
>
> It is possible no one wants to answer this due to the NDA they had to sign?
>
Though they have written me back twice to say "coming
On Sun, 28 Jun 2009, Thomas Kenyon wrote:
> Jeff LaCoursiere wrote:
>> On Sun, 28 Jun 2009, randulo wrote:
>>
>>> On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote:
>>>> Hi,
>>>>
>>>> Has anyone tried it ?
>>>> Is there any
On Wed, 1 Jul 2009, Ken D'Ambrosio wrote:
> Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The
> VM box, itself, is beginning to show its age. Big-time. We're thinking it
> might be time to look for a replacement. I'd love to install Asterisk
> with an FXO card or somet
On Thu, 2 Jul 2009, Elliot Murdock wrote:
Hello!
Which RFC specifies the corresponding number of the formats?
Where in the Asterisk source code does it state the SDP formats?
Does Asterisk follow the formats of IANA?
(http://www.iana.org/assignments/rtp-parameters)
Thank you,
Elliot
Perha
s the offer from your device (what is it?). Where is the reply?
Please post the relevant section of your sip.conf.
j
On Thu, Jul 2, 2009 at 4:04 PM, Jeff LaCoursiere wrote:
On Thu, 2 Jul 2009, Elliot Murdock wrote:
Hello!
Which RFC specifies the corresponding number of the formats?
Where i
On Thu, 2 Jul 2009, Jeff LaCoursiere wrote:
On Thu, 2 Jul 2009, Elliot Murdock wrote:
Hello Jeff,
Yes, I use G729 all the time.
Here is the SDP extrace from Wireshark. I'll get more data as it
becomes available:
Session Description Protocol
Session Description Pro
Search the archives - we had a big discussion about this phone about six
months ago. If you make it work and want another one "I will give you
special price!".
j
On Tue, 14 Jul 2009, Cesar Gonzalez wrote:
> Has anyone played with this phone? i cant seem to get it to work
> properly, i manged
Howdy,
Getting ready to play with QoS settings. We have an asterisk 1.4.23
server running in a colo bunker in the US Virgin Islands under a large
radio tower. That tower has multiple "sector" radio/antenna pairs that
blanket a valley in 802.11a. The customers have directed dishes aimed at
On Tue, 14 Jul 2009, Cesar Gonzalez wrote:
> Jeff LaCoursiere wrote:
>> Search the archives - we had a big discussion about this phone about six
>> months ago. If you make it work and want another one "I will give you
>> special price!".
>>
>> j
>
se at the same time.
Cheers,
j
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: Tuesday, July 14, 2009 5:04 PM
> To: asterisk-users@lists.digium.com
>
Apologies for the off topic post... hoping someone knows if 800 number
portability in the states is legally enforced? One of my customers is
being told by their current "vanity" 800 provider that they own the number
and refuse to release it to their new carrier. I thought I understood
that i
On Thu, 16 Jul 2009, Don Kelly wrote:
> Changing toll-free RespOrgs (Responsible Organizations) is different from
> number portability.
>
> That said, the owner of a toll-free number has the right to change RespOrgs,
> so the question is "Who is the owner?"
The "owner" in this case is "CallSourc
On Mon, 20 Jul 2009, Oguzhan Kayhan wrote:
[snip]
> And for freepbx, i had the problem like i couldnt find how to group my
> users for outgoing calls.For some users i want to give only local call
> permissions for some including international calls etc.
> And there are no group selection or cre
On Wed, 22 Jul 2009, Olivier wrote:
> 2009/7/22 Steve Edwards
>
>> I finally found a reason TO run Asterisk as root.
>>
>> By default, ext[23] file systems "reserve" 5% of the filesystem for root.
>
> Do you imply this default can (and should) be changed ?
> Is it the same for other filesystems
What router did you install it on? Any stats on concurrent conversations
/ transcoding?
Cheers,
j
On Fri, 24 Jul 2009, David Cook wrote:
> Yeah, have it running on several units. It's really quite simple now.
>
> - Goto System -> Packages
> - Scroll down to "Update Package List" and wait a f
On Fri, 24 Jul 2009, David Cook wrote:
> Yeah, have it running on several units. It's really quite simple now.
>
> - Goto System -> Packages
> - Scroll down to "Update Package List" and wait a few seconds for that puppy
> to refresh.
> - You now should have a list of "installed" packages followed
In all my messing around with openwrt and dd-wrt this weekend I ran across
"Milkfish" which appears to be OpenSER scaled down to run on top of
openwrt and dd-wrt. The "voip" dd-wrt v24 actually has it built in
already.
We are about to launch an ITSP so have been testing a number of SIP
hardp
ncryption didn't work at all.
>> Maybe other versions (with more memory or faster CPU's) are better, but
>> my results were a disaster and i would not consider running Asterisk on
>> top of that.
>>
>> Joachim
>>
>> David Cook wrote:
>>
>
On Wed, 29 Jul 2009, Myles Wakeham wrote:
> I'm pretty new to this whole Asterisk system & VoIP thing, but being a
> programmer by trade the complexity didn't scare me off (at least not yet)...
>
> I have setup an Asterisk system for my home & home office. My wife & I
> run two separate business
Howdy,
Just installed a new switch in a new location (Ubuntu, 2.6.24-24 kernel,
zaptel 1.4.12.1 built from source, libpri-1.4.10.1 built from source,
asterisk 1.4.26 built from source, wanpipe 3.5.4 built from source,
Sangoma A104d with firmware that is probably a year old).
I plugged in an R
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Thompson
> Sent: Tuesday, August 04, 2009 9:36 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Anyone actively using RLT for mobile
> phoneforwarding?
>
>
> H
I have always been confused about how MWI is working with asterisk. If a
SIP device has the option to subscribe to MWI, should it? I think
asterisk sends NOTIFY messages to SIP clients if the sip peer entry has
"mailbox=". Is there any advantage then to leaving that out of the sip
peer entr
Howdy,
My first forray into using res_mysql.conf for realtime access of sip users
and extensions.
I have the following relevant section of extensions.conf:
---
[trunklocal]
exten => _NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[local]
include => trunklocal
include => trunkt
Meant to add that this is 1.4.26... :)
On Fri, 7 Aug 2009, Jeff LaCoursiere wrote:
>
> Howdy,
>
> My first forray into using res_mysql.conf for realtime access of sip users
> and extensions.
>
> I have the following relevant section of extensions.conf:
>
> -
On Fri, 7 Aug 2009, Jeff LaCoursiere wrote:
>
> Howdy,
>
> My first forray into using res_mysql.conf for realtime access of sip users
> and extensions.
>
> I have the following relevant section of extensions.conf:
>
> ---
>
> [trunklocal]
> exten => _
On Tue, 18 Aug 2009, Kevin P. Fleming wrote:
[snip]
>> Note: It is *mandatory* to configure an echo canceler for the
>> system's channels using dahdi_cfg unless the interface cards in use
>> have echo canceler modules available and enabled. There is *no*
>> default software echo canceler
On Wed, 19 Aug 2009, Dave Fullerton wrote:
> Tzafrir Cohen wrote:
>> On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:
>>
>>> Here's my $0.02. If you don't want an echo canceller, specify
>>> echocanceller=none,x-y and have dahdi_cfg print a warning (at any
>>> verbosity level) when
On Mon, 31 Aug 2009, ilker Aktuna wrote:
> Hi,
>
> My Trixbox 2.8.0.1 installation includes the following Asterik version:
> 1.6.0.9-samy-r27
>
> I am having some problems with it and I think they might be solved if I use
> the latest Asterisk version.
> Is it a good idea to update Asterisk in T
I actually use any
of the trixbox-only features. I've also been enamored with Ubuntu of
late, and have dumped CentOS. YMMV, but you might consider starting over
with a clean build of the linux of your choice, and doing asterisk +
addons + FreePBX from source.
j
>
> Thanks.
&
I'm only top posting to keep the flow going. Otherwise this would get
messy. ilker - you should consider bottom posting to not raise the ire of
others on the list.
This may be a silly question, but do you have "mailbox=" filled in with
the extension's number on the SIP extension page? If no
On Tue, 1 Sep 2009, James Lamanna wrote:
> Hi,
> I'm trying to configure 2 parallel sip trunks between 2 boxes.
> However I seem to have the problem that when making a call from Box 2
> to Box 1, it sometimes
> says authentication failed because it is using the username of the other
> trunk.
>
>
On Wed, 2 Sep 2009, li...@mgreg.com wrote:
> Hi All,
>
> As is obvious by my joining the list, I'm interested in learning more about
> Asterisk. I have downloaded the PDF manual (for version 1.4) and am
> beginning to go through it. What I'm looking for in the short-term, however,
> is a mor
On Wed, 2 Sep 2009, Antoine Patte wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Gordon Henderson wrote:
>> DNS.
>>
>> Run a caching DNS server on your Asterisk box, or a suitable device on
>> your network. (eg. the DHCP server)
>
> The network gateway has already a dns cache.
> Ina
On Fri, 4 Sep 2009, Leif Madsen wrote:
> tom wrote:
>> im confused:
>>
>> http://www.h-online.com/open/Starfish-PBX-Management-Tool-for-Asterisk--/news/114154
>>
>> whats that now? cloned - not cloned?
>
> It specifically says it's not a clone, however I can't see any way in which
> they
> justi
On Wed, 9 Sep 2009, hadi motamedi wrote:
> Thank you for your message . But I tried to find it on my server , as the
> followings :
> #find / -name sip.cfg -print
> But it didn't return any result . Can you please let me know where can I
> find it ?
You probably have not setup central provisioni
On Tue, 8 Sep 2009, Ira wrote:
> At 07:31 PM 9/8/2009, you wrote:
>> For the home user, 2xFXO + 6FXS, in a single slot small profile box is
>> ideal, but only able to offer 2xFXO + 4xFXS at the moment.
>
> Wow, I can't imagine ever using an analog phone on Asterisk. SIP
> phones are just so much
On Wed, 9 Sep 2009, David Backeberg wrote:
> On Wed, Sep 9, 2009 at 10:12 AM, B.Masoud @ SH wrote:
>> I don't want to bill the first 30 seconds, that's all, why is that so
>> strange??? My line does not support polarity reversal, so I don't want to
>> bill for ringing the line...
>> Do you sugges
On Wed, 16 Sep 2009, Doug Lytle wrote:
> Matt Riddell wrote:
>> Basically, the phones are displaying 79 on the screen (the number the
>> dial for pickup) - as you'd expect, but they'd like to see the CID of
>> the person who called in.
>>
>>
> There are patches against 1.4 that allow you to chang
On Wed, 23 Sep 2009, Tzafrir Cohen wrote:
> On Tue, Sep 22, 2009 at 07:43:51PM -0500, Martin wrote:
>> I do not know if fonebridge would work here since it sends/receives
>> the ~2 Mbps (for each circuit/port)
>> of data over ethernet ... constantly. That could choke the USB ...
>
> Ethernet has
On Fri, 25 Sep 2009, Giedrius Augys wrote:
> Hello,
>
>
> I have one problem and I need to disable dtmf (disable rfc2833, info and
> inband) on one (other peers must support dtmf) SIP peer . Is it possible?
> Workaround would be use g729 codec with dtmfmode=inband.
I don't think that would *d
On Sat, 26 Sep 2009, hadi motamedi wrote:
> Thank you for your reply . But I am seeking for PPPoE remote access that
> fits my case here . Can you please let me know if there is any solution in
> this regard ? (like PPPD)
It would be really cool if iaxmodem would actually answer an incoming
mo
On Sat, 26 Sep 2009, Alan Lord (News) wrote:
>
> Hmmm, has anyone tried SIP over a VPN?
>
> We are thinking of testing this but haven't yet...
>
> Al
>
I have a client with Sonicwall VPNs. Asterisk is at head office on
internal LAN, six external locations all have Linksys 2102 ATAs and
Polyco
On Sat, 26 Sep 2009, John A. Sullivan III wrote:
> We are using SIP over both IPSec and SSL VPNs very successfully with
> access controls in the tunnel ingress via the ISCS network security
> management project (http://iscs.sourceforge.net). There are a couple of
> issues.
>
> I'm not sure wha
On Mon, 5 Oct 2009, James Lamanna wrote:
> Hi,
> I noticed that Dahdi starting producing these error messages:
>
> ERROR[29250] chan_dahdi.c: No more room in scheduler
> ERROR[29250] chan_dahdi.c: Asked to delete sched id -1???
>
> during which time I could not send any calls or receive calls on
Anyone working on this? Would love to have a "click to talk" that would
operate with my Grandstream video phones.
j
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register N
On Fri, 9 Oct 2009, Michelle Dupuis wrote:
> Before we call each other liars and thieves, here is a link:
>
> http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>
> As with any open source, do your own homework on licensing, AND apply your
> own reasonable judgment.
>
> At this point I
On Fri, 9 Oct 2009, Gordon Henderson wrote:
> On Fri, 9 Oct 2009, Kevin P. Fleming wrote:
>
>> Gordon Henderson wrote:
>>
>>> All deskphnoes I've ever bought support g729 natively. They also all
>>> support G711. The wholesale termination services I use all support g729
>>> too. The only fly in
On Fri, 9 Oct 2009, Moises Silva wrote:
>>
>> I would be very surprised if that were true. Your phones speak many
>> codecs, but they negotiate with asterisk on registration which one they
>> will be using. They don't switch codecs based on the remote channel
>> (which they don't even know abou
On Tue, 13 Oct 2009, Gianluca Baù wrote:
Hello guys,
i have a question about a voip gateway we use.
I saw those values typing in cli:
core show translation
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc
g726 g722 slin16
g723 - - - -
Wish I could have made it :( Is there a possibility of a collection of
the talks/slides/handouts/videos/presentations for download? Even pay
for?
Cheers,
j
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AstriCon 2009 -
On Tue, 20 Oct 2009, Darrick Hartman wrote:
> John Todd wrote:
>> On Oct 17, 2009, at 7:47 PM, Michael Graves wrote:
>>
>>> I'm told that they will show up on the event site in about three
>>> weeks.
>>>
>>> On Sun, 18 Oct 2009 02:29:48 +0
On Tue, 20 Oct 2009, ABBAS SHAKEEL wrote:
> Hello
> I need some advice regarding the Asterisk server that are located at
> different locations.
>
> Three asterisk servers are here each at different location. Suppose A,B,C be
> the three servers respectively.
>
> Server A is connected to server B
On Tue, 20 Oct 2009, ABBAS SHAKEEL wrote:
> Yeah i do have considered that option but the problem is that in if i have
> four servers server ie A,B,C,D... all cannot be servers ands clients at the
> same time. Thats the reason I am wondering any other solution
>
Well you already suggested that
Working with a new client that has a ton of these phones, and in a new
installation the phone is registered, can place and receive calls with no
issues, but has a "locked" picture of a phone in the upper right corner.
Any Linksys experts know what this means? I have searched the admin guide
a
Want to make sure I understand why a caller might not hear "ringing" when
outbound calling.
A SIP phone is behind a firewall and is registered to an asterisk
server on a public network. Sometimes (but not always) when placing an
outbound call there is no ringing before the remote party answer
On Tue, 20 Oct 2009, Jimmy Godbout wrote:
> Can you send a picture of this ?
>
> Thanks
>
>> -Original Message-
>> From: j...@jeff.net
>> Sent: Tue, 20 Oct 2009 23:34:13 + (UTC)
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Linksys 962
>>
>>
>> Working with a
t; means its not online.
>
> Maybe on the phone a user pass has been set?
>
> best regards
>
> steve
>
> Jeff LaCoursiere schrieb:
>> Working with a new client that has a ton of these phones, and in a new
>> installation the phone is registered, can place and recei
On Wed, 21 Oct 2009, Danny Nicholas wrote:
> Not my cup of tea, but I think I'd be trying an ODBC connection to reduce
> some overhead here.
>
>
>
[snip]
Does that reduce overhead or add it? Seems that direct mysql-client code
should be more efficient than adding ODBC in the middle...
j
> Steve Edwards wrote:
>
>> Since I'm an "old-school" C programmer, I use emacs as my editor. I fire
>> up gdb (the GNU C (amongst other languages) debugger) in a window, give it
>> a command like "b main; r > through my program line by line, examining and changing variables at will.
>>
Bah. If
Sorry for the off-topic, but perhaps this will be of interest to other
asterisk based ITSPs.
We are starting service in a rural area where the ILEC has the rural
"monopoly". From what we have read in the FCC docs this does NOT exempt
them from number portability, but what does it take for us
On Sat, 31 Oct 2009, Cary Fitch wrote:
> Two more comments.
>
> Yes, to join the PSTN call distribution system you must have SS7.
>
> While rural ILECs are not exempt from number portability, there is a court
> injunction that saves them from having to transport the call out of their
> local rate
On Sat, 7 Nov 2009, jonas kellens wrote:
> vps*CLI> iax2 show channels
> Channel Peer UsernameID (Lo/Rem) Seq
> (Tx/Rx) Lag Jitter JitBuf Format
> 0 active IAX channels
> vps*CLI> core show channels
> Channel Location State
> Applica
On Thu, 12 Nov 2009, Cary Fitch wrote:
> I am not sure what the problems are and the reasons for the basic 64K modems
> used in VOIP are. I understand the compressed codecs that get the bandwidth
> down to 20-30 K. And perhaps the 64K units give much better potential audio
> than you would get
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