Erase your caller directory, happened to me because the default ring on
directory was 1 (silent)
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Sent: Thursday, April 07, 2005 12:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Me too...
:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor
Harrison
Sent: Tuesday, April 12, 2005 11:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: polycom phones
On Apr 11, 2005 11:49 PM, Gre
I've spent may hours to play with HTB QoS settings on the firewall, but with
absolutely no effect. In fact, this is normal, because the time required to let
a data packet going through the ADSL line will break the voice jitter. The only
right way to handle this issue is to modify the MTU on the
What version of mpg123 do you use ? You must have a 0.59r one, else you
will ear a strange noise, and the nothing. But no warning message.
For test, you can try this version:
ftp://ftp.proxad.fr/pub/Distributions_Linux/Mandrakelinux/official/10.2/i586/media/main/mpg123-0.59r-23mdk.i586.rpm
This i
Hello all,
sccp transfer question!
Tell at somebody it has turned out to make a transfer of a call
between 3 sccp (HARD) phones.
|---| |---| |---|
| A | ---> | B |-(#)->| C |
|___| |___| |___|
sccp sccp sccp
And if that has turned out as? If that is possible an exa
The command you may play with, is CallingPres. The values that did work
for me, with a zaphfc an with Swisscom (telco), are:
- 0 - hide callerID
- 32 - show callerID
There is a quite good explanation you to calculate the presentation on:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd
On Thu, 2005-05-05 at 17:20 -0500, Daniel Bingham wrote:
> Reviewing the IP-600 and IP-500 further, the spec sheets have me a
> little confused. The IP-500 states it supports three lines, and the
> IP-600 six lines. What is confusing me is that on the IP-600 spec sheet
> (http://www.polycom.com/c
> -Original Message-
> From: Adam Goryachev [mailto:[EMAIL PROTECTED]
> Sent: Thursday, May 05, 2005 10:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Re: Problems with TDM400P card
>
>
> On Thu, 2005-05-05 at 12:11 -0400, Andrew Kohlsm
On Thu, 2005-05-05 at 12:11 -0400, Andrew Kohlsmith wrote:
> On May 5, 2005 11:13 am, Mike Mueller wrote:
> > > Couple this with the fact that the driver now seems to pull 100% CPU
> > > every 5 seconds or so and it didn't before and I think we have a good
> > > case for there being something weird
Hi,
I've been trying to get music on hold going on one of our servers:
Upon dialling extension 005, it plays:
-- Executing WaitMusicOnHold("SIP/parssyd1-4dbe", "30") in new stack
-- Started music on hold, class 'default', on SIP/parssyd1-4dbe
However, no music in the background
MPG123 i
can anyone tell what is the "unknown RTP codec 72" means and how to fix it.
I'm using xlite to call PSTN line and the message just pop up on my console but the call can be connected. What am I going to do?__Do You Yahoo!?Tired of spam? Yahoo! Mail
Other notes:
The clever integrator of this application will save themselves some
lookup $ by caching the responses from the database into their own
database, along with a datestamp. Perhaps if an entry is >90 days
old, the system will re-lookup the entry in the Accudata database but
otherwis
[cross-posted to -biz and -users since it could fall into either category]
Interesting new product that has been introduced that I think some
would be interested in here (at least, those users in the United
States and perhaps Canada): CNAM delivery via IP lookup.
The problem: inbound calls on ma
Hey All,
Our upstream provider requires the use of H323 and after several months
(6!) of having problems with OH323 I've decided it might be worth biting
the bullet and getting a cisco device that can gateway up to
approximately 50 calls from SIP to H323.
Would a 2500 or 2600 series do the job?
calls get setup with the config below but in the stats all calls are failed.
BYE are sent from uac to asterisk but after no more than 1 response is
receieved. All calls show to be failed in stats printed by sipp (uac).
isn't the uas part of sipp supposed to received 200 messages after BYE has
be
Sorry, I just fixed it by myslef. It is an issue of incompatible
codec. I am wondering why option "t" in dial() is not able to make it
work.
Any advice??? Many Thanks.
On 5/6/05, VoIP Newbie <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> I could register * to a provider. However, I failed to make out
On Thu, 2005-05-05 at 21:56 -0500, Jon Gabrielson wrote:
> On Thursday 05 May 2005 05:28 pm, Joseph wrote:
> > It has 1-FXS and one 1-Life Line (it is pass through type)
>
> I've seen the pass-through term used alot and
> I'm not quite for sure what that means. What is the
> difference between a
Hi all,
I could register * to a provider. However, I failed to make outgoing
calls through the provider. Please help and advise how to get it work.
m2*CLI> sip show registry
HostUsername Refresh State
sip_proxy:5060 abc105 Registered
on 5/3/05 21:46, Matt Riddell at [EMAIL PROTECTED] wrote:
> Would there be any chance of creating a GPL exception for them if we
> donated them?
>
> I have rather a few songs, mostly in the trance/psytrance genre but also
> dub and DnB.
>
> Ideas?
What we do is find local artists from our nearb
Hi David,
First, thanks for the reply to my questions about the Snom 360. I may have a
few followup questions when I get a little more time.
As for the 360 getting the configuration directly from Snom's servers, I find
that very backwards. What if your phones have no gateway to the internet?
I use cdr_pgsql on asterisk-head to store my CDR records. The issue is
that regardless to what I set amaflags to in iax.conf, the data that
gets written to the cdr.amaflags is always 3. What am I missing?
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Pass through has the same functionality as a modem with a "line" and a
"phone" connection. Line is where you plug in the dialtone, the dial passes
through the "phone" connection unless the card picks up (like a modem does).
I have a X100P clone that is setup as a passthrough. I've never seen a pas
On Thursday 05 May 2005 05:28 pm, Joseph wrote:
> It has 1-FXS and one 1-Life Line (it is pass through type)
I've seen the pass-through term used alot and
I'm not quite for sure what that means. What is the
difference between a passthrough type and a regular
FXO. What can you do with one that y
> P3 1Ghz under Tao Linux 1.0 (2.4 Kenrnel) cvs-stable w/ X101P
>
> --- Results after 66 passes ---
> Best: 1.024461 -- Worst: 1.024420 -- Average: 1.024447
>
> And on our new gateway box...
>
> P4 3.0 Ghz under Tao Linux 4.0 (2.6 Kernel) cvs-stable w/ TE405P
>
> --- Results after 106 passes -
On Thu, 5 May 2005, Colin Anderson wrote:
> I like easy questions!
>
> Add to /etc/rc.d:
>
> modprobe wctdm
> ztcfg -vv
> su /usr/sbin/safe_asterisk &
>
> where is the user that you normally run Asterisk under.
> Omit the su if you are running Asterisk as root.
> safe_asterisk is a watchdog
I know I may be asking the obvious, but shouldn't we get tother to
build a matrix (prehaps on the wiki) showing boards / servers that
work against cards.
Unless it is already in place, in which case would one of you be kind
enough to point me in the right direction?
Thanks
David
On 5/5/05, Natha
> I need an rc script that will autorun the module for the wctdm and then
> follow that by running asterisk. A first look at google tells me to try this
> list.
Take a look at 'make config' in both the zaptel and asterisk src directories.
It should install the scripting necessary.
_
On Thu, 5 May 2005, Charlie Watts wrote:
> Manjit Riat wrote:
> > Out of curiosity what's the reason? Why would they not sell phones to
> > asterisk users? Do they not trust asterisk or their phones to work
> > with each other?
>
> My guess: They don't want to compete with the folks that OEM Po
on 5/5/05 11:42, Derek Whitten at [EMAIL PROTECTED] wrote:
> nufone has been rock solid
>
> http://www.nufone.net
Rock Solid Are you using switch-1.nufone.net? We have seen congested
messages quite a bit this week. ...especially for international calls.
--
Stu
On Wed, 4 May 2005, Andrew Kohlsmith wrote:
> > When the span was "0", I NEVER got that message. I haven't heard any
> > complaints from the other office mates that use the PRI for voice, but the
> > error just bothers me.
> >
> > What is the real difference between "0" and "1" on the span timing?
It is more then a cisco 7940
On Fri, 6 May 2005, David John Walsh wrote:
Dan
I've had a snom 360 "on the bench" for about 10 days now. I too have
become dispondent with Cisco's licencing structure
The snom 360 provides the majority of the high end Cisco features you
would use 99% percent of the ti
I like easy questions!
Add to /etc/rc.d:
modprobe wctdm
ztcfg -vv
su /usr/sbin/safe_asterisk &
where is the user that you normally run Asterisk under.
Omit the su if you are running Asterisk as root.
safe_asterisk is a watchdog script that restarts Asterisk if it craps out.
The & Ampersand r
Dan
I've had a snom 360 "on the bench" for about 10 days now. I too have
become dispondent with Cisco's licencing structure
The snom 360 provides the majority of the high end Cisco features you
would use 99% percent of the time, it doesn't have the flexibity of
the on-demand buttons around the s
Hi, I have an annoying issue whereby when my fax machine hangs up (after
receiving a fax or any call) when on a FXS interface (I have a TDM400P),
the default incoming ring kicks in. I guess the voltage drop when the fax
hangs up is causing asterisk to think there is an incoming call.
Any ideas
P3 1Ghz under Tao Linux 1.0 (2.4 Kenrnel) cvs-stable w/ X101P
--- Results after 66 passes ---
Best: 1.024461 -- Worst: 1.024420 -- Average: 1.024447
And on our new gateway box...
P4 3.0 Ghz under Tao Linux 4.0 (2.6 Kernel) cvs-stable w/ TE405P
--- Results after 106 passes ---
Best: 1.023967 --
Hello
Although not stictly a asterisk issue, any help would be apreciated.
Firstly a few notes on the snom 360, which I have had on a test bed
for the last week. Its a great phone, with a good user interface,
both physically and its web based one.
At its lastest firmware it does have a few quir
I use GalaxyVoice and they'll let you BYOD. $25 a month all you can eat
US and Ca.
Mark
Steve Maroney wrote:
Yeah, I agree, Nufone as well as Voicepulse Connect hasn't giving me much
troulbe compared to Broadvoice but Broadvoice seems to be the only
"Unlimited" server that supports byod.
Thank yo
Hi,
On 5/5/05, Niksa Baldun <[EMAIL PROTECTED]> wrote:
> Assuming your h.323 phones are registered with gnugk, you need to
> instruct gnugk to forward certain numbers to Asterisk. In OH323 (which I
> am using) you would need to add something like:
>
> [register]
> gwprefix=0
> gwprefix=1
> etc.
>
I've been wanting to write a Cisco XML "web service" that would allow
the user to view currently parked calls as a dialing directory that
had the pickup extensions as numbers.
I've been trying to figure out the best way to either access the
parking lot contents or maintain the contents through hoo
I need an rc script that will autorun the module for the wctdm and then
follow that by running asterisk. A first look at google tells me to try this
list.
Thanks,
--
Jeff Ramsey
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http://li
You don't need to have multiple connections defined. (I think you can
IAX trunk multiple conversations over ONE IAX connection (one data
stream) but you can run multiple streams without problem (if you had
concurrent multiple connections you may be able to gain some efficiency
using IAX trunkin
Before the issuing the dial command issue the setvar command like this:
exten => 1234,1,SetVar(CDR(accountcode)=value)
exten => 1234,2,Dial(SIP/whatever)
this should take care of it.
On 5/5/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> Yes, but how do I do it automatically like when a SIP call is
Hi, anyone know of a default driver option that could be set so that an
incoming pstn call on my fxo interface would be forwarded/bridged to 1 or
more fxs interfaces and vice versa.
Would be nice to have a fail-open option, whereby a phone connected to the
fxs interface could still make/receiv
Quote from the site when you click on the "download SIP software" link:
SIP
Software - Certified VoIP Resellers can download from the Polycom Resource
Center
NOTE: At this time, end-user customers can not download software. Please
work directly with the Polycom Certified VoIP Reseller you purchas
I Had the same Problem as you did...
I used the following from the list as a template and Setup up my dial Plan
Accordingly...
http://lists.digium.com/pipermail/asterisk-users/2004-September/062564.html
Hope it helps.
Dave
Chris wrote:
>I haven't gotten to keys yet.
>The documentation out
I will try it tomorrow. After I got it to work I didn't pay much
attention to the other configurations.
Chris
- Original Message -
From: "Tim Pushor" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, May 05, 2005 5:14 PM
Subje
Apologies for asking more questions so quickly after my last one. A few
more questions about the Polycom phones:
Searching the list I found a few references like this:
"I would also like to figure out how to make the phone *ring* when
you're already on another line, but haven't had a chance to
It's easy.NAT has to have redirection. However IAX uses UDP 4569.
If you include Trunk=yes in your IAX.CONF you will only need one trunk
configured.
You can also do "TCPDUMP -i eth0 udp and port 4569"
This will show you the traffic. You should see bidirectional traffic when the
SI
Yes, but how do I do it automatically like when a SIP call is made?
Plus, on some DIDs I don't know which phone will answer until after it has
answered.
-Matthew
C F wrote:
> do:
> SetVar(CDR(accountcode)=value)
>
> On 5/5/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
>> Scenario #1:
>> SIP UA
On Thu, 2005-05-05 at 16:51 -0500, Jon Gabrielson wrote:
> The AG-168E has an FXO port?
> The only seller I can find seems to think it is just a single FXS port.
> http://www.iaxtalk.com/product_info.php?products_id=30
>
> You wouldn't happen to have another link with more info would you?
>
>
>
Thank you to both Chris and Tim
I could not get my head around this .. after seeing the examples it now
makes sense what needs to be done. I will give both a whirl tonight.
I do like the RSA key idea.
One question is this, will I need multiple accounts on the Static IP
machines so the Dynamic m
I have most agents using Alaw and/or ulaw and a handful of agents using
gsm.
Thanks,
- Daniel
On May 5, 2005, at 1:12 PM, Mehdi Chouikh wrote:
If use Alaw or ulaw as codec, i think it's
enough.
But if you need to make transcoding to a hard codec like g729, g723,
you have to look other cpu.
reg
mattf <[EMAIL PROTECTED]> writes:
> They would also be competing indirectly with themselves(Cisco and Avaya
> license their technology in their phones). But the real reason comes down to
> support. I've been following the Polycom/Asterisk thing for 2 years now and
> it really comes down to the fac
Is there a way to have the IAXY dialout with a call file?
I tried:
Channel: IAX2/606/5068012
Context: smvoice-dialout
Extension: smvoice
Priority: 1
RetryTime: 2
WaitTime: 20
MaxRetries: 0
And it did not work.
Is there a way to have the iaxy call out from the outgoing spool directory?
606 is the ex
On 5/5/05, JD Austin <[EMAIL PROTECTED]> wrote:
> If I get a standard business line from qwest, plug it into my FXO card
> and get the call forward busy service,
> will that allow me to handle more than 2 inbound calls?
Yeah sure, and how will qwest make money? In most cases you have to
pay per ch
do:
SetVar(CDR(accountcode)=value)
On 5/5/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> Scenario #1:
> SIP UA 1 -> Asterisk -> PSTN
> CDR shows account code of SIP UA 1; as expected, works great.
>
> Scenario #2:
> PSTN -> Asterisk -> SIP UA 1
> CDR shows no account code.
>
> How can I
Its scattered and hard to find. Once I understood the relationship
between user and peer (and friend, somewhat), it was pretty clear.
Actually I find RSA authentication much easier than managing
usernames/passwords (but I will have about 10 boxes that need to be able
to talk to each other).
I
Reviewing the IP-600 and IP-500 further, the spec sheets have me a
little confused. The IP-500 states it supports three lines, and the
IP-600 six lines. What is confusing me is that on the IP-600 spec sheet
(http://www.polycom.com/common/pw_item_show_doc/1,1276,1820,00.pdf),
under the IETF SIP bu
When I get an incoming pstn call to an FXO port on my Adtran 600 Channel
Bank, I see this in the CLI. What causes it?
-- Starting simple switch on 'Zap/13-1'
May 5 17:20:29 ERROR[9329]: callerid.c:262 callerid_feed: fsk_serie made
mylen < 0 (-16)
May 5 17:20:29 WARNING[9329]: chan_zap.c:573
Matthew Boehm wrote:
Tomasz Chmielewski wrote:
Matthew Boehm wrote:
first, let me know, if you can dial yourself? (i.e. PSTN -> * - that
(zap) card)
Yep. I sure can.
so everything seems OK.
I guess we would need:
- /etc/zaptel.conf
- /etc/asterisk/zapata.conf
- the construction of the extension
I haven't gotten to keys yet.
The documentation out there doesn't seem to be very good.
Chris
- Original Message -
From: "Tim Pushor" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, May 05, 2005 4:06 PM
Subject: Re: [Asterisk-Users
Tomasz Chmielewski wrote:
> Matthew Boehm wrote:
>
> first, let me know, if you can dial yourself? (i.e. PSTN -> * - that
> (zap) card)
Yep. I sure can.
-Matthew
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http://lists.digium.com/ma
My current LAN
===
100Mbps LAN (with 15 windows
- ---
| Switch |---< |
DSL Router at 128kbps | -->
DNIS is dialed number identification service. With toll free services,
the carrier will send DNIS digits (to your PBX) to identify which toll
free number was called. The DNIS sent from the network to your PBX may be
4 digits long for example. I suspect that you will need to build an
extension ma
The AG-168E has an FXO port?
The only seller I can find seems to think it is just a single FXS port.
http://www.iaxtalk.com/product_info.php?products_id=30
You wouldn't happen to have another link with more info would you?
Thanks,
Jon.
On Thursday 05 May 2005 01:33 pm, Joseph wrote:
> Indee
Hi Salina. If you are new in GNU/Linux like operation systems, and
Asterisk, there is much that you have to read and learn, not
impossible but difficult i think. However is worth to do it. It would
be nice that you tell us what have you done so far, and what do you
already know in order to not wa
Nope. Just tried it. ForkCDR doesn't mess with accountcode.
-Matthew
kritikus Araklidas wrote:
> Could you use ForkCDR.
>
> Regards.
>
>
>
>> From: "Matthew Boehm" <[EMAIL PROTECTED]>
>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>>
>> To:
>> Subject: [Asterisk-Users]
Matt wrote:
When I try to make a call out my PRI I get 'all-circuits-are-busy-now'
from allison... and I get:
Look at the value of ${HANGUPCAUSE}. Use something like
Noop(HANGUPCAUSE=${HANGUPCAUSE})
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On Thu, 2005-05-05 at 17:09 -0400, Daryl G. Jurbala wrote:
> Huh? The last time I dealt with DNIS (admittedly, years ago) the
> provider sent the digits to ME via DTMF to tell ME what number was
> dialed to terminate on that line (you knowDialed Number
> Identification Service).
>
> Unless DN
Tomasz Chmielewski wrote:
Eric Wieling aka ManxPower wrote:
Tomasz Chmielewski wrote:
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I
can call myself and talk to asterisk :)
Zap/0 is not a valid Zap channel.
Tomasz Chmielewski wrote:
Eric Wieling aka ManxPower wrote:
Tomasz Chmielewski wrote:
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I
can call myself and talk to asterisk :)
Zap/0 is not a valid Zap channel.
Matthew Boehm wrote:
Before you jump ahead, yes I do have chan_zap.so loaded..
Call Flow: Asterisk 1 --IAX2--> Asterisk 2 ---> PRI
-- Accepting AUTHENTICATED call from 22.22.22.22:
> requested format = ulaw,
> requested prefs = (),
> actual format = ulaw,
> host pre
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Thursday, May 05, 2005 6:20 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] 1800 DNIS and asterisk (HOW TO?)
>
> Hi
> I have applied for Qwest 1800 ter
This is going to seem like a stupid question but its been bugging me.
According to documentation I have read. The current stable release of
Asterisk is 1.07
(which is available for download from ftp.asterisk.org in a tarball).
However I should
also be able to download the current stable versi
- Original Message -
The Grandstream HandyTone 488 has an FXO port.
I've never used it though.
I could be wrong, but I seem to remember reading up on the HandyTone and
deciding that it doesn't really act like a true FXO, as in calls come in and
go straight to Asterisk like an FXO, and
> I think it's a server/connection issue with the LiveVoip server.
Perhaps. I'm still testing, but since moving to their NYC server,
audio quality has been very good to excellent in the past days. Not
too bad, after all, but still too early to draw a conclusion.
--Luki
> If I get a standard business line from qwest, plug it into my FXO card
> and get the call forward busy service,
> will that allow me to handle more than 2 inbound calls?
Yes, I'm doing that right now with Alltel. Alltel's implementation
is CO-based (programmed by their techs). If you ask them
Personally, if I owned both boxes and had full control of the dialplan
on both, I'd stay away from passwords. (but be careful what I say, I'm a
hack)
I have a bunch of boxes connected together via IAX and authenticating
via RSA. The entries in iax.conf are simple, and dialing across the
connec
Well having heard of the praises for polycom phones and everyone loving them
and having trouble finding them I think I'll sign the "We'll only
sell Polycom equipment". Let me look into that for more info..
-Original Message-
From: Charlie Watts [mailto:[EMAIL PROTECTED]
Sent: Thursday, M
There is a thread in dev or biz about this too. A guy got referred some
motherboards by Digium he can't get easily in Australia.
I'll second that the DL380 G2 seems to work, I'll know more in a few months.
The common thread seems to be either a serverworks chipset or more
specifically, a chipset
Hi Team:
Somebody knows how to configure some extension for monitoring on live a
group of other extensions.
Regards.
Kritikus.
_
Express yourself instantly with MSN Messenger! Download today - it's FREE!
http://messenger.msn.cli
On Thu, 5 May 2005, Peter Svensson wrote:
> On Thu, 5 May 2005, Vikram Rangnekar wrote:
>
> > what i noticed is that when i pull any one end of the E1 (breaking the E1
> > connection) I get multiple RED ALARMS on the zap channels I understand this
> > is ok and should happen if the E1 link brea
On Thu, May 05, 2005 at 12:11:51PM -0400, Andrew Kohlsmith wrote:
> On May 5, 2005 11:13 am, Mike Mueller wrote:
>
> > What was that? No buffering? That means its tx/rx ISR should have priority
> > over those servicing interfaces with buffering. Is that happening?
>
> It's one of the primary rea
Joseph wrote:
>
>The AG-168 supports IAX2 and the FXO port is "pass though" type.
>The difference is that SPA-3000 answer the phone and rings asterisk (the
>phone at this moment has been answered the ringing party is incurring
>the charges before asterisk answered the phone), the AG-168 is ringin
Could you use ForkCDR.
Regards.
From: "Matthew Boehm" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
To:
Subject: [Asterisk-Users] Account Code in all cases?
Date: Thu, 5 May 2005 13:52:29 -0500
Scenario #1:
SIP UA 1 -> Asterisk -> PSTN
CDR shows acco
I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the
new box, I've installed a generic ebay X100P. I don't have my livevoip or
voicepulse accounts set up yet on the new box (can both boxes be registered
at the same time?). I've set up one IP phone (SPA841) with the new bo
Before you jump ahead, yes I do have chan_zap.so loaded..
Call Flow: Asterisk 1 --IAX2--> Asterisk 2 ---> PRI
-- Accepting AUTHENTICATED call from 22.22.22.22:
> requested format = ulaw,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (ulaw|alaw|gsm
8192 samples in 8194 sample intervals 99.975586%
8192 samples in 8194 sample intervals 99.975586%
8192 samples in 8194 sample intervals 99.975586%
8192 samples in 8194 sample intervals 99.975586%
8192 samples in 8194 sample intervals 99.975586%
8192 samples in 8194 sample intervals 99.975586%
8192
They would also be competing indirectly with themselves(Cisco and Avaya
license their technology in their phones). But the real reason comes down to
support. I've been following the Polycom/Asterisk thing for 2 years now and
it really comes down to the fact that they do not want Asterisk users
call
When I try to make a call out my PRI I get 'all-circuits-are-busy-now'
from allison... and I get:
May 5 16:02:05 DEBUG[1337]: Call from user '200' is 1 out of 0
May 5 16:02:05 VERBOSE[1337]: -- Executing Macro("SIP/200-36c5",
"dialout-trunk|1|3232166") in new stack
May 5 16:02:05 DEBUG[133
Hi
I have applied for Qwest 1800 termination to a T1 ISDN PRI. They tell me
that I will have to program a predefined DNIS number on my switch.
According to them unless asterisk returns that DNIS number no call will
get through.
How do I program the DNIS, is it through zaptel.conf or some other
I have something similar. Both of my servers are behind a firewall and
NAT. You will need to allow UDP 4569 through the firewall for IAX2. If you
have NAT you will need to redirect 4569 to the internal server.
I would suggest using AMP and then looking at IAX_ADDITIONAL.CONF to see
Hi:
Is it possible for two asterisk boxes behind a nat to
initiate iax call over the internet?
I can understand the case when one has a public ip and
one behind a nat. In such a case, the nat asterisk
would register with the public ip asterisk and IAX
channel is always open to initiate a call. But
Manjit Riat wrote:
> Out of curiosity what's the reason? Why would they not sell phones to
> asterisk users? Do they not trust asterisk or their phones to work
> with each other?
My guess: They don't want to compete with the folks that OEM Polycom
hardware. Lots of commercial phone system vendor
OK. So I am testing Monitor() and res_monitor.
exten => 3044,1,Monitor(wav,mytest,mb)
exten => 3044,2,Dial(SIP/3044,15)
exten => 3044,3,Voicemail([EMAIL PROTECTED])
exten => 3044,103,Voicemail([EMAIL PROTECTED])
If I use "wav49" I get a nice, proper duration file with sound. (meaning I
can he
Its all there specified as you recommend
On 5/5/05 20:39, "Tim Thompson" <[EMAIL PROTECTED]> wrote:
> Make sure you have something along the lines in your Zapata.conf file as
> well.
>
>
>
> [EMAIL PROTECTED]
> callerid="Tim Thompson"<311>
> channel => 21
>
>> -Original Message-
>> F
Make sure you have something along the lines in your Zapata.conf file as
well.
[EMAIL PROTECTED]
callerid="Tim Thompson"<311>
channel => 21
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dan Goscomb
> Sent: Thursday, May 05, 2005
On Thu, May 05, 2005 at 12:33:50PM -0600, Joseph said:
> The difference is that SPA-3000 answer the phone and rings asterisk (the
> phone at this moment has been answered the ringing party is incurring
> the charges before asterisk answered the phone), the AG-168 is ringing
> the asterisk directly,
David Brodbeck wrote:
> I find this game kind of infuriating. If you have problems, they
> tell you to buy a different motherboard. But they don't supply a
> list of "approved" ones that they'll support.
One fellow at Digium suggested to me that the HP/Compaq D380 works well.
And it comes with a
Any one using voip.net with asterisk?
Jerry
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