Actually, Digium Support has been quite responsive in recent weeks, as
noted on this list 2 weeks ago:
http://lists.digium.com/pipermail/asterisk-users/2008-April/209457.html
We strive to be as responsive as we can, and have had some success on
this front recently. Please give us a
Hi Olle -
Actually, there's a large difference between an IAX2 trunk and an IAX2
connection.
The IAX2 trunk multiplexes multiple media streams in one UDP packet,
therefore you can call it trunking. In order for this to work, you
need to enable a zaptel timer source in your system.
any of you guys have used FOP for drag and drop transfer on 30 40 phones
environment?
At one point, I used it for about 35 phones (25 users). I had to
really do some adjusting to the size of the buttons, but it worked
well. I thought it was very useful, as it showed MWI status, and was
great
how stable is that?
The version I used is probably a couple of versions old now, and it
was pretty reliable then. I imagine it would has probably at least
stayed as stabled if not improved a bit.
Mmmm. Me talk well english! At the risk of being redundant and
wasting list resources,
Any suggestion for a headset (cord and cordless) for IP601?
Any good (and economical) ones from Polycom or Platronics?
I don't know about cordless, but for corded, I've had great success
with Plantronics H91N's.
- Noah
___
-- Bandwidth and
Hi James -
The VLAN used by the phone can be configured in several ways:
1. Hard-code it on the phone. Not recommended if you have lots of phones.
2. Auto-discovery using CDP. Requires Cisco or older HP switches.
3. Auto-discovery using DHCP. Disabled by default in SIP 2.1.x.
We use
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: Thursday, 13 March 2008 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IP650 console with expansion modules
Hi Bill -
I just replaced an IP 601 with a new IP 650. We
Hi Bill -
I just replaced an IP 601 with a new IP 650. We have 2 expansion
modules attached. The lights on the expansion modules light up if
a users gets an INBOUND DID call, but the lights don't light up if
the user makes an OUTBOUND call.
Sip: 2.1.1.0052
Has anyone seen this?
Hi Daniel -
Thank you for guide most things become cleare. No I dont need the dial tone.
When I pickup XLITE to dial a number I hear dialtone and after I enter
number nothing happens, this behaviar was strange for me, exactly
becase you said I have analog phone in mind :)
The only thing
Hi Bilal -
1) If I pressed 1 twice (11), so it runs the step
related to first 1 and then it runs the step related
to second 1, so it does buffering for my input and run
two steps, how can I make it run only the step related
to first entered digit 1 and does not do buffering
(so
Hi -
Thank you all for answers. As I understand s - i and others is device
specific.
I will not need them in my SIP configuration.
The s extension is not zap-specific. You can use it for any type of
device. It's just the generic extension that a call will go to when
no other matching
Hi Steve -
I could be mistaken, but I think this has to be done physically from
the phone. I don't think you can do this with central provisioning or
from the web interface.
- Noah
On Wed, Mar 5, 2008 at 3:20 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
I setup a number of remote phones on
Hi Bilal -
I have the following configuration in my iax.conf
files at asterisk box1 and box2 (two asterisk):
At box1:
[user1]
disallow=all
codec=g729
codec=GSM
At box2:
[user2]
disallow=all
codec=g729
codec=GSM
If G729 is no more available at box1, so how can I let
Hi All -
Anyone know if the callpark feature is in ABE?
Is there a comprehensive list of the differences between ABE and the
open source version? I've only seen a bullet-point chart which has no
real detail.
Thanks,
Noah
___
-- Bandwidth and
Hi Mick -
I've had a functioning Asterisk system (1.2.18), which I haven't
reconfigured in any way, that is just now refusing to forward calls. I
only have Polycom phones. When I use the phone's forward feature
(forwarding the phone with extension 204 to extension 206, which used to
work
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Noah Miller
Sent: Thursday, December 13, 2007 21:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.2.18 and Polycom
phones notforwarding anymore
Hi Mick
Hi Kevin -
I'm trying to decide between the foneBRIDGE2 ($1135) and foneBRIDGE2-EC
($1610).
Would we really suffer
without the onboard echo cancellation?
Each situation is different, but I have a client that had significant
problems with echo on their PRI. Asterisk's software EC (any of
Hi Alex -
Is there a way to dynamically alter the sip.conf properties of a SIP peer
in runtime without doing a SIP reload?
realtime (i.e. database)?
- Noah
___
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asterisk-users
Hi -
I guess that's my question. Is this the standard method of doing faxing?
Just point the PRI DIDs to a TDM and hang fax machines off of the ports?
I've never used a TDM880 for this purpose, but I've used multiple
TDM400's in this capacity (PRI - TE4XXP - TDM400 - Fax Machine),
and it works
Hi Arun -
I've configured my asterisk and voicemail all works fine but I want to
restrict call time to voicemail that is when user calls voicemail he can
use voicemail system only for a max of 5 min that is after five minutes
asterisk should disconnect the call.
Do you mean that you want the
Now I'm confused. Laura from Digium said the AA50 was and I quote not
user editable when it comes to the config files. Which is it? Can I edit
the config files, or not?
Particularly of concern/interest to me is.. can I put my aastra phone config
files on the flash and access them from
Hi All -
Has anyone had a chance to use the Asterisk Appliance yet? Any
thoughts or reactions? I have a couple of clients waiting on the
Zaptel version, but maybe somebody has used the VoIP-only version?
Thanks,
Noah
___
--Bandwidth and Colocation
Hi David -
Last I checked, the replacement with the new firmware is only for those who
bought the card in the last year (i.e. the card is still under warranty).
Those of us who were early adopters cannot enjoy the improvements of the
upgraded firmware without buying all new cards.
IEEE802.3af uses same 4 wire as data.
thats what Polycom uses.
the way i'm seeing it we are better off with poe switch(looking at the
price).
802.3af has two different modes:
Mode A: uses the same 4 wires as 10/100 ethernet, typically done by
PoE endpoints like switches
Mode B: uses the
Hi Mike -
It seems like since we got FIOS
installed (including switching to fios phone lines which are supposed to be
the same on our end) i am having massive problems with asterisk not hanging
up dead calls for days, even weeks if i dont catch it. It slowly builds up
randomly not ending a
Hi Vieri -
I'm trying to set a rule to dial out through multiple
Zap groups so that, say, g0 is the cheaper POTS lines
group
and must be used first. However, if g0 is busy or
disconnected then try dialing out g1.
My g0 group is made up of 4 analog lines connected to
a 4-FXO card. I
Hi Bilal -
The question here is: how asterisk will be able to
receive calls at two network cards where each network
card has a different IP address.
Maybe we need to know if asterisk is doing a hear on
the ports only without caring for IP or it is doing a
hear only on the IP:port?
If you
Hi Satish -
I have configure asterisk with 100 SIP PHONE ( SNOM ) but now
thing is that my boss need phonebook feature find extention number by Pbook
so i have read about it there is a feature in asterisk but it is with
voicemail now i have IP SIP phone of SNOM so how to fine
Hi Bilal -
What is folks? Where I can find it about VPN solution?
We have a language misunderstanding here. Folks means people. Tim
meant that he knows that some people have VPN's working with Asterisk.
As far as which VPN: feel free to choose - SSL-based, IPSEC, PPTP -
whatever you're
You really need to update to a later version of asterisk (and zaptel).
There have probably been somewhere close to a thousand bug fixes
since 1.2.10. If you still have this issue with the latest version,
please collect as much information as possible (exact cli messages,
turn on
You have to first uninstall your Asterisk1.2 like this--
First you have to stop your asterisk...using--
1. killall -9 asterisk or killall -9 safe_asterisk, whichever you are
using.
In my experience, you don't need to do this step. In fact, you can
keep the old asterisk running,
Hi Dave -
question, can the IP address receiving the incoming call be used in
extension logic to determine call handling procedures, or maybe a better
way to ask is can asterisk provide information as to the IP address on
which a request was received?
If you have control (or influence)
Hi Philipp -
Since the list was switched over to API-Digital almost
every message I get is older than a week. Coincidence?
Is anyone else having trouble?
Well, this is now the third active thread on this subject, but I guess
you won't see this message for a while. Has anyone dissected the
Hi Zeeshan -
I have to install an Asterisk PBX for a customer and he wants something like
logic supply's fanless computers. Can anybody advise about how good will
they work, are they compatible with the Asterisk system? I'll also be
installing a sangoma 4 port FXO card in it.
Have you
Hi Norman -
To add to what Edgar said, yes, use linux-ha. It works nicely in
combination with DRBD. DRBD uses a dedicated network interface on
each box with a crossover cable between the two. It does a block
level copy of the entire filesystem, so you have two machines that are
identical.
Hi Arun -
Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents
this asterisk box is connected to another asterisk box using 5 IAX trunk to
load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my
cli start flooding with message: Maximum trunk data space exceeded
If you're using the Snom transfer button, you don't need to do
anything in features.conf. In extensions.conf, just make sure that
the dial() command used to call the snom phone uses the 't' flag.
THIS IS INCORRECT!
The options t and T are for DTMF based transfers. You do not need any
I'm looking for 24 or 48 port IEEE802.3af POE injector.
Any recommendation?
Yes. For the price of one of those multi-port injectors, you can come
close to the price of a new Netgear or 3Com PoE switch. The injectors
typically add power to the unused pairs (mode B PoE). This means you
can't
You have to first uninstall your Asterisk1.2 like this--
First you have to stop your asterisk...using--
1. killall -9 asterisk or killall -9 safe_asterisk, whichever you are using.
In my experience, you don't need to do this step. In fact, you can
keep the old asterisk running, compile and
Hi Satish -
you are right but can u explain me i have SNOM SI 120 phone with transfer
button on it but what entry i will do on asterisk feature.conf and what
configuration and button will use for transfer call
If you're using the Snom transfer button, you don't need to do
anything in
Hi Norman -
On 7/18/07, Norman Franke [EMAIL PROTECTED] wrote:
I've been evaluating Asterisk for a while, and things seem to be
going very well. The issue of redundancy and automatic fail-over is
now on my mind. I searched the archives and googled for solutions,
but didn't really come up
Hi Don -
Yesterday I upgraded from asterisk 1.2.21.1 to 1.2.22. We are running
Zaptel-1.2.17.1. After upgrading all of our calls from the phone company
(DIDs trunks with wink start on a channelized T1) were not coming in.
I believe you'll need to upgrade Zaptel to the latest version if you
Hi Stefan -
What I want to accomplish:
- calls within the LAN are re-invited (RTP goes from endpoint to endpoint)
- asterisk detects when a call is going beyond the local LAN (over the NAT),
and then stays in the middle.
I'm wondering if this is hard to do and how I'm supposed to configure
Hi Satish -
I am going to install 2 port pri card on asterisk but i dont
know how to incomming call goes in to IVR and how to route call outside base
on pattern match means if some one call on mobile phone then use PRI 1 and
if call on landline phon call route through pri 2
Hi Ira -
In the end my issue would seem to be I did something out of order,
though I thought I did it right, and as much as I tried the software
only gave meaningless to me messages. I didn't understand and I don't
think I've seen it said before that I should run .\configure every
time I do
I have a strange comportment of the MOH system on my asterisk.
When i respond to a call and after fews second i set this call in hold
mode the correspondent listen the music fine.
When i re-take my correspondent at T0 instant the music is paused. And
when i re-hold him at T60 (60
Hi Mark -
I'm having a tough time figuring out how to do something. If I have an
operator (which could potentially be in their own context) and an
internal-only context, is it possible to make it so the operator can
call the internal-only context but *NOT* transfer calls to it?
Sort of.
Hi -
I have a strange comportment of the MOH system on my asterisk.
When i respond to a call and after fews second i set this call in hold
mode the correspondent listen the music fine.
When i re-take my correspondent at T0 instant the music is paused. And
when i re-hold him at T60 (60 second
Hi Matt -
Right... you dial *67 to block, however WE are the phone provider and
need to set the appearance value so that when our customers dial *67
we correctly block their caller-id from going out.
Have you tried explicitly setting the CID variables to NULL strings?
- Noah
Hi Matt -
No I haven't. Shouldn't I be able to set the appearance to like '4'
or '5', etc?
I can do that on the PRI's I've had experience with. I found that on
most landlines, this will show up as Unavailable or something
similar, but on most cell phones it will show that number.
- Noah
Hi Florin -
On Linux, it looks like Ekiga is a good candidate. But how about Windows?
There's a windows version of Ekiga, too. It's at the very bottom of
the download page.
- Noah
___
--Bandwidth and Colocation Provided by
Hi Robert -
I am putting some scripts together to allow a local admin to add
extensions, then to reload the extensions, something like:
asterisk -r -x extensions reload
Are registered extensions forced to reauth?
Nope.
Are active calls disrupted?
Nope.
Reloads are safe to do in the
The 430's have two line appearances. I'm trying to get the second line
registered to a different extension but for some reason it's not
allowing me to do this. The first line will register fine but the second
line never seems to register no matter how I swap the device ID's and
permissions
Hi Arun -
I need help in configuring a auto dialer system using Asterisk. I'm holding
my customers number in MySQL want to fetch 10 numbers one time and dial if
gets connected and answered by customer wants to play a sequence of message
I've tried here is my code to place calls but in this I
Hi Matt -
What do I need to do to set the outbound appearance on a call so that
it shows up as Unavailable or Private?
In most cases, I think you'd need to arrange this with your provider.
If you want to do it on a call-by-call basis (in the US), dial *67
before you dial the number. If you
i am using tdm400P in my office. i tested that TDMF generated by asterisk
is so bad. the sound is very soft and quality is so bad. i am using
asterisk 1.2.18. most of time, the # key can not be detected correctly.
Does anyone has that problem?
please give me a hit for that problem!
Hi Arun -
using php script and Asterisk manager I'm dialing numbers and once gets
connected send to an exten in my dial plan that plays an automated message
but some time without answering even it goes to my exten. How can I handle
early media in Asterisk that is I want only when user answer
Is DTLS available for Asterisk on any Linux distro?
Nope.
I've read that the reSIProcate SIP stack has DTLS support.
- Noah
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asterisk-users mailing list
To UNSUBSCRIBE or
It is recommended
to stop asterisk b4r doing make install of new version .
It should work to do a make install of a new version while the
previous version is running. At least, I've never had any issues
doing it.
- Noah
___
--Bandwidth and
Hi Giorgio -
I'm testing attended transfer with 3 SIP phones. I noticed about 10% of
my transfers make the call drop and I get this on my log:
Some questions:
1. What asterisk version are you using?
2. What are your SIP devices?
3. Who is your SIP provider? (Judging by your CLI output, I'm
Hi Gary -
What I want to do is take one of my SIP devices to my office (which is ALSO
behind another NAT) and try to connect with my home Asterisk box with it.
For port forwarding, my AsteriskNOW box has a static IP on the inside of my
NAT and I've configured the LinkSys router to
Verizon wants to pretend the service doesn't exist
I don't know, they're advertising it:
http://www.verizonbusiness.com/us/voice/local/compare.xml#isdnbri
- Noah
___
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Hi Giorgio -
1 - my asterisk version is 1.2.18
2 - my SIP devices are SNOM phones
3 - no SIP provider is involved...they are connected to my
Asterisk...this is the strangest thing.
This happens sometimesI think it could be a network overload...can
it be?
Well, that's possible, but
Hi Again Brent -
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXX?
I think you've got it right already. What do you need to do?
If you wanted to get more specific and identify ONLY NANP, you may
have to break it out into more than just one rule:
Hi Brent -
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXX?
I think you've got it right already. What do you need to do?
- Noah
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Hi Eve -
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as you can imagine there will be
congestion when some of
Hi Bilal -
In other words, how can I let the ued codec for the IP
Trunk between my Asterisk and the other IP PBX to be
g729 and not g711? Ofcourse, I am assuming that the
other side also supporting g729.
You can have multiple allow lines, i.e.
allow=g729
allow=ulaw
Be
Is it just me? After the mail list server upgrade, the average delivery
time for messages to the users list is between 4 and 5 days. The Dev
list seems fine!
I'm getting new messages within a matter of minutes. I dunno.
- Noah
___
--Bandwidth
Eagerly waiting for v1.4.x to mature a bit before getting serious about
it.
Is it ready for production yet? If that's too general, where is it in
terms
of stability compared to where 1.2.x is now. Anyone running it
successfully
in production environment and if so what sort of
Hi Bilal -
If I need to do a trunk between Asterisk and another
SIP softswitch (so Asterisk will send a SIP calls to
that softswitch), then I have to configure this on the
sip.conf file
Yes.
And is it the same
when I configure iax trunk?
Not exactly the same, but very close. Here's a
Hi Ronaldo -
I have a IAX trunk between two asterisk servers, both with dynamic IP
and both have a DNS name associated with it.
In the iax.conf file I configure the host parameter with the DNS name
of the servers. Everything works fine until one of these servers get a
new IP, so the other can't
3. Can you post some of the CLI errors you mentioned?
iax2_trunk_queue: Maximum data space exceeded
and once this start it never gets stopped so I've to kill the
asterisk and restart the whole box. Instead of restart whole box if
I just try to restart the asterisk my agents not able to
Hi Carlos -
HI, my problem is with internal sounds of asterisk.
for example when calling voicemail, no system recordings are being
played back. However, when running asterisk in a debug mode, i see the
call coming through to the system and the system playing back the wav
files promptly.
Most small/medium companies have a T1 for all their phone needs.
Internally there is a need for some analog lines.
* Fax Machine - FXS
* Security System (most ask/demand two lines) FXS
* Paging - FXO
* Dialup systems
I think he's asking why both T1 and FXS/FXO need to be on a single card.
Hi Arun -
I've two boxes connected over IAX2 trunk before IAX I was using SIP trunk
and they were working fine b'coz of bandwidth issue I changed from SIP to
IAX now I'm facing a strange problem after some time on the cli of my
asterisk box I see lots of messages of IAX2 trunk and b'coz of that
Hi Steve -
It's definitely ftp. I have given the phone a static ip. When I set it to
dhcp it just
hangs and cannot get an IP. I can ping the phone and see the web config page
so it is on the network.
Any more suggestions.
I once saw this same sort of behavior on a Polycom 501. It got
Hi JR -
Has anyone gotten the polycoms or the linksys phones to accept oprtion
66 on the dhcp request for the address of the tftp config server?
Yes. I've gotten this to work successfully using Polycom phones with
DHCP from Cisco routers and firewalls (I generally don't use ISC's
DHCP).
Hi Marco -
We bought some Linsys WRTP54G-NA boxes which have WIFI, 4-port, 2 SIPs...
The two SIP ports work on A* if you call one line to talk to the other in
the same box.
When we pick up a line, dial to another phone via the A* server, this will
ring at the other end... But, when you pick
Hi David -
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
Hi Olivier -
Our last trial was so conclusive (every call was affected), we step back to
previous situation without HPEC.
We will do our best to help to solve this (gathering audio captures for
instance) though it will be very hard for me to convince our customer to
try.
Being able to to
I am looking for hardware sip phone with very good sound quality. Can anyone
recommend ?
I use to have Grandstream Budge-Tone 100 but I feel that the sound is not
very
satisfactory and volume too soft
Polycom, Snom, Cisco, Aastra
___
--Bandwidth and
Hi Matthew -
Can any one suggest if asterisk-users is the best mailing list for questions
on Digium Iaxy (S101I) hardware, or a different one if not?
Digium support is the best resource for this. How old is the affected IAXy?
- Noah
___
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of around 6) i wouldnt be able to pull that off.
The world is a big place, and I suppose there's room for all kinds.
In these parts, the
I have a requirement of running 10 PRI's (300 Channels). I still have to
decide on hardware and cards. Can you suggest some. As per my
understanding it will be tough to go beyond 150.
Alex Balashov wrote:
On Mon, 14 May 2007, Kapil Dhawan said something to this effect:
I want to try Asterisk
Our last trial dates from 1.2.17 days (3 weeks ago).
My question is : are those HPEC audio clipping issues fixed with 1.2.17.1 ?
It's not about the Asterisk version, it's about the HPEC version.
According to other posts on the list, HPEC version 8.2 does not have
the clipping issues, but the
Hi Bilal -
Well, I understood now that Nortel has some digital
phones that can be used with astrisk, but the
question: what are the card models that should be
installed on Asterisk server? Digium? What these
models?
If you use the Citel Portico gateway, you don't need any telephony
card,
Hi Ritesh -
Does anyone know if there is a known list of telemarketers?
Something like http://whocalled.us/ with an easier access?
We could all benefit if there was such a thing :-)
If there is enough interest, I could put up a database that everyone can
benefit from.
I just need some
Hi Zvonimir
I have an issue that hopefully you can help me solve. I've got the sangoma
a101 card and installed it on freebsd but I according to the manual I should
be see when running dmesg PCi0 vendor. something that tells me sangoma
it's being recognized by the system. Now this is the 2nd
Hi Nitesh -
Thanks everyone... The GXV-3000 IP Video Phone works with Asterisk 1.2
using H.263 Video Coder.
I had to update both phones firmware with new one...
Out of curiosity - do you like the phone? I've looked for reviews,
but I haven't found any that rate the phone's functionality.
Many thanks to you all. I think that I can do this for IAX based VPN set
up between two Asterisk servers
in remote offices.
If it helps at all, I read a study that said that SSL VPN's can
actually help with jitter problems. So it might be preferable to
implement something with OpenVPN (uses
We are running 2.1.1 with 3 expansion modules. Presence in turned on in
sip.cfg, hints are set up on the extensions and buddy watch is enabled on
about 40 extensions mapped to the expansion modules. We have 3 paging
groups: Building A with 20 phones, Building B with 20 phones and Both
with all
1. A patch allowing capture audio streams in a way that will allow [us] to
debug (and presumably fix) the problem was mentioned by Kevin. -
Anything new about it ?
I couldn't find it in Zaptel 1.2.17.1 nor 1.4.2.1 changelog.
I believe it was introduced in Zaptel 1.2.17.1. From the description
Just wondering if anyone has tried using Asterisk 1.2 on CentOS 5. Is it
worth considering for a Production install yet?
I'm not quite there yet. I'll probably do the upgrade within 2 weeks, though.
Did they fix that
spinlock.h Kernel problem?
Looks like it's fixed:
Hi Steve -
[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten = s,1,Set(CALLERID(number)=${ARG1})
exten = s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4)
exten = s,3,Set(ARG2=1${ARG2})
exten = s,4,Dial(${TRUNK}/${ARG2},,m)
exten = s,5,Congestion()
exten = s,105,Busy()
This macro
I've heard of a device that acts as a failover for a PRI line so you
can plug a PRI into two different devices and have the PRI failover if
one device fails. Unfortunately nothing like this is commercially
available today.
Sounds like the ISDNguard:
Hi Eric -
How do you handle transfering vmail from one user to another when they're on
separate servers?
I'm using the single vmail server, mounted NFS partition for this right now.
I'd love to be able to have them standalone so they're survivable when the
WAN collapses, but I haven't figured
Hi Bilal -
What is the difference between TDM11B and TDM04B? Why
the price of TDM11B cheaper than the price of TDM04B?
The TDM11B has one FXS port and one FXO port. The TDM04B has four FXO
ports. More FXO modules on the TDM04B means a higher price.
- Noah
Russell from Digium is working on a piece of code for monitoring
'states' or 'status' across multiple servers (mainly to monitor do I
have a voicemail for collection on another server).
Do a google on the -dev list and you'll find some posts to reply to.
Hey Thanks Dean! That sounds like a
Hi Matt -
I have two Trixbox servers that are connected over the Internet via an IAX2
connection. We are experiencing very poor sound quality. I have tried many
different codecs gsm, ilbc, g729, g711 and all seem to have the same
problem. (All though g729 seems to work the best but still
Hi Joseph -
Thanks, I think you are on the right track.
When no Sip adapters were connected to asterisk it took me over one
minute from the time I typed reload to the time I've seen anything on
the screen.
When, I connected the all the sip devices and eliminated some entries in
sip.conf and
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