On 12/11/09 21:21, Jonathan Thurman wrote:
>On Fri, Dec 11, 2009 at 7:52 PM, Joseph wrote:
>[snip]
>> Thank for suggestion.
>> Well, it is not that cheap but the problem with their equipment is luck
>> support and decent manual.
>
>I actually find the Quick-start guide that comes in the box the m
On Fri, Dec 11, 2009 at 7:52 PM, Joseph wrote:
[snip]
> Thank for suggestion.
> Well, it is not that cheap but the problem with their equipment is luck
> support and decent manual.
I actually find the Quick-start guide that comes in the box the most
useful, if you aren't doing anything strange.
On 12/12/09 04:02, Jeff LaCoursiere wrote:
[snip]
>> Thank for suggestion.
>> Well, it is not that cheap but the problem with their equipment is luck
>> support and decent manual.
>> Whatever I google about AudioCodecs everybody seems to be straggling with
>> the setup; I don't think this should
On Fri, 11 Dec 2009, Joseph wrote:
> On 12/11/09 14:05, Jonathan Thurman wrote:
>> On Fri, Dec 11, 2009 at 12:44 PM, Connor Spiess wrote:
Joseph
>>>
>>> You could also check out the Audio Codes gateways if the Grandstream
>>> doesn't work out for you. They make FXO/FXS
>>> gateways. They w
On 12/11/09 14:05, Jonathan Thurman wrote:
>On Fri, Dec 11, 2009 at 12:44 PM, Connor Spiess wrote:
>>>Joseph
>>
>> You could also check out the Audio Codes gateways if the Grandstream doesn't
>> work out for you. They make FXO/FXS
>> gateways. They were reliable boxes for us but this was to a non
Hi All,
I've been knee deep in T38 faxing for a couple of weeks now, trying to
find a version of Asterisk that would pass through T38 with an
Audiocodes Mediant 1000 and MP203 ATA. I had problems with 1.6.0.x
through 1.6.1.10. Tested 6 different versions. Either it just would
not work or fail b
Hi All,
My client makes manual sales calls to prospects. He is often sent to
voicemail on the prospect's side. If he finds himself having to leave a
message, he would like to be able to press a key and let a pre-recorded
message play into the prospect's vmail box. This is so he can maintain
consist
"restart when convenient"
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, December 11, 2009 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-
Un-top-posting...
> On Wed, 9 Dec 2009, Michelle Dupuis wrote:
>
>> However, I have a cron job that tries to restart asterisk and gets
>> this
>> error:
>
>> No such command 'restart gracefully' (type 'help restart gracefully'
>> for other possible commands)
> On Behalf Of Steve Edwards
> Sent: F
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last "incoming" label defined in those trunks' contexts in
sip.conf.
My ITSP insists on insecure=very in the trunk context; is this the cause?
John
2009/12/11 Noah Miller :
>> I assume if all the SIP trunks are to the
James Lamanna wrote:
> Has terminating T.38 to PSTN found its way into the asterisk 1.6 mainline yet?
> I remember seeing an app_gateway floating around at some point a while
> ago, but I never had any luck with it.
It has not, no.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technol
On Fri, Dec 11, 2009 at 12:44 PM, Connor Spiess wrote:
>>Joseph
>
> You could also check out the Audio Codes gateways if the Grandstream doesn't
> work out for you. They make FXO/FXS
> gateways. They were reliable boxes for us but this was to a non-asterisk PBX
> over MGCP. I mention them cause
Hi,
Has terminating T.38 to PSTN found its way into the asterisk 1.6 mainline yet?
I remember seeing an app_gateway floating around at some point a while
ago, but I never had any luck with it.
Thanks.
-- James
___
-- Bandwidth and Colocation Provided b
Looks like single quotes did the trick. No idea why...but the error is gone
from my log
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, December 11, 2009 4:23 PM
To: Asterisk Users
On 12/11/09 14:44, Connor Spiess wrote:
>>Looks very interesting, but they are all either FXS or FXO; it would be
>>practical if they could make it 2xFX0 and or 2xFXS or modular design.
>>Cisco had such unit but they discontinued it.
>>
>>--
>>Joseph
>
>You could also check out the Audio Codes gat
On Wed, 9 Dec 2009, Michelle Dupuis wrote:
> However, I have a cron job that tries to restart asterisk and gets this
> error:
> No such command 'restart gracefully' (type 'help restart gracefully' for
> other possible commands)
Did you find a solution -- inquiring minds want to know...
--
Th
On Fri, Dec 11, 2009 at 10:25:33AM -0700, mir shahnawaz wrote:
> Hi there,
>
> I am trying to configure chan_dahdi.conf for TDM404E. Should I
> separate channels for dialing out and recieveing calls on this card or
> should I leave it random so that outgoing and incoming call get first
> available
-Original Message-
From: Joseph [mailto:syscon...@gmail.com]
Sent: Friday, December 11, 2009 1:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ATA FXO
On 12/11/09 12:52, Connor Spiess wrote:
>>
>>-Original Message-
>>From: Joseph [ma
On 12/11/09 12:52, Connor Spiess wrote:
>
>-Original Message-
>From: Joseph [mailto:syscon...@gmail.com]
>Sent: Friday, December 11, 2009 11:37 AM
>To: asterisk-users@lists.digium.com
>Subject: [asterisk-users] ATA FXO
>
>>I'm looking for a reliable ATA FXO/FXS adapter.
>>
>>Linksys 3102 -
-Original Message-
From: Joseph [mailto:syscon...@gmail.com]
Sent: Friday, December 11, 2009 11:37 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ATA FXO
>I'm looking for a reliable ATA FXO/FXS adapter.
>
>Linksys 3102 - a lot of echo problem + two of them died within a
I'll check it out, but Grandstream HT503 doesn't have a good introduction on
voip-wiki web-page:
http://www.voip-info.org/wiki/view/HT-503
--
Joseph
On 12/11/09 19:37, jonas kellens wrote:
>Grandstream HT503
>
>On Fri, 2009-12-11 at 10:37 -0700, Joseph wrote:
>
>> I'm looking for a reliable ATA
Hi!
Are you sure you are getting Astrisk out of the media path? I guess
reinvite must be allowed. Then it should work without transcoding
licenses.
Maybe you should take a look at the SIP DEBUG info to see what codec
Asterisk is trying to negotiate with the trunk. You could disallow
alaw and ulaw
Grandstream HT503
On Fri, 2009-12-11 at 10:37 -0700, Joseph wrote:
> I'm looking for a reliable ATA FXO/FXS adapter.
>
> Linksys 3102 - a lot of echo problem + two of them died within a year (not
> reliable)
> Sangoma USBFXO - problem installing drive in Gentoo.
>
> I've tried two Chines units
> I assume if all the SIP trunks are to the same host/port, Asterisk
> cannot distinguish which trunk is active when an incoming call is
> made- it will dump all incoming calls to the context specified in the
> last trunk entry of sip.conf
No. SIP uses authentication (well, I guess you can not us
I have – but I don’t see why that would be required for pass – through?
The codec purchase should only be required if I wanted to leave voicemail in
G729 or MOH.
If my end points support G729 and I am advertising it in the invite, and
negotiating it with the 200OK, I don’t see why its not allow
Have you paied for and imported g729 licenses from digium so that
asterisks can use g729?
http://store.digium.com/productview.php?category_id=5&product_code=8G729
CODEC
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21
Hi;
I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my
endpoints.
It seems that when I enable G729 on my peers in sip.conf and make a call I
am getting the following errors:
Called crp_uk/806575011971553141421
Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a code
I assume if all the SIP trunks are to the same host/port, Asterisk
cannot distinguish which trunk is active when an incoming call is
made- it will dump all incoming calls to the context specified in the
last trunk entry of sip.conf
Thanks
John
2009/12/11 Martin :
> On Fri, Dec 11, 2009 at 10:23
I'm looking for a reliable ATA FXO/FXS adapter.
Linksys 3102 - a lot of echo problem + two of them died within a year (not
reliable)
Sangoma USBFXO - problem installing drive in Gentoo.
I've tried two Chines units: AG-188N and YGW30B
none are of them have real FXO port that will register with A
Hi there,
I am trying to configure chan_dahdi.conf for TDM404E. Should I
separate channels for dialing out and recieveing calls on this card or
should I leave it random so that outgoing and incoming call get first
available channels.
;FXO Modules
group = 2
echocancel = yes
signalling = fxs_ks
con
> I'm actually there, but I was wondering if the tables there are up to
> date and if any changes took place. I see all kinds of comments about
> changes.
You could go ahead and install and then look at the table structure
using your dbms.
- Noah
___
-
>> The echo between our extensions (using Polycom 550 handsets) disappears
>> once I removed the Digium echo module.
> Are you routing internal calls from SIP -> DAHDI -> SIP? The digium
> echo module will not have any effect on pure SIP <-> SIP calls. Do
> you have acoustic echo cancellation a
On Fri, Dec 11, 2009 at 10:23 AM, John Taylor wrote:
> Thanks - have done that and am now trying a one out. However, I'd
> still like to know whether 1 asterisk server can support multiple
> trunks/registry entries. Does it cause problems?
yes, Asterisk does support multiple registry entries...
if
On Fri, Dec 11, 2009 at 10:30 AM, Leif Madsen
wrote:
> Warren Selby wrote:
> > Can I install my free fax for asterisk license on more than one
> > machine? I.e using my digiun account to download the free FFA module,
> > am I restricted to just the first machine I put it on, or can I put
> > the
Warren Selby wrote:
> Can I install my free fax for asterisk license on more than one
> machine? I.e using my digiun account to download the free FFA module,
> am I restricted to just the first machine I put it on, or can I put
> the free FFA on multiple servers?
I would believe you get a s
Thanks - have done that and am now trying a one out. However, I'd
still like to know whether 1 asterisk server can support multiple
trunks/registry entries. Does it cause problems?
Thanks
John
2009/12/3 Tim Nelson :
> - "John Taylor" wrote:
>> I want to use an asterisk box to provide a voip
Hi,
Using this OctoBRI card in a bristuff-0.4.0-RC4-xr7.tar.gz-enabled machine,
I discovered zttool was not able to detect its NT mode.
I opened a thread on this as I'm suspecting this older type of OctoBRI card
might need a specific driver.
Regards
___
> The echo between our extensions (using Polycom 550 handsets) disappears
> once I removed the Digium echo module.
Are you routing internal calls from SIP -> DAHDI -> SIP? The digium
echo module will not have any effect on pure SIP <-> SIP calls. Do
you have acoustic echo cancellation active on
Where in the code does something like:
register => user[:secret[:authuse...@host[:port][/extension]
from sip.conf 1) get parsed 2) actually register.
I tried looking in channels/chan_sip.c and don't see where that happens.
Can someone point me the right file and or function.
T
Can I install my free fax for asterisk license on more than one
machine? I.e using my digiun account to download the free FFA module,
am I restricted to just the first machine I put it on, or can I put
the free FFA on multiple servers?
Thanks,
--Warren Selby
_
Hi,
Using Xorcom's bristuff-0.4.0-RC4-xr7.tar.gz (ie asterisk 1.4.25) with an
old Junghanns OctoBRI (ie not the 2.0 version), zttool shows every port in
TE mode, though half of them are in NT. Alarm column shows OK for every
port, such as :
OK octoBRI PCI ISDN Card 1 Span 1 [TE] Lay
Steve Underwood wrote:
> Something is wrong if Asterisk is sending:
>
> a=T38FaxFillBitRemoval
> a=T38FaxTranscodingMMR
> a=T38FaxTranscodingJBIG
>
> Spandsp supports T38FaxFillBitRemoval, but neither spandsp or Commetrex
> support the other two options. The Commetrex guys have said so in the F
> What are the limits with asterisk server running on one decent (4GB, 4 CPU
> etc.) machine.
There are a LOT of factors involved. You will likely have to do your
own testing with just the specific features you want.
> How many MeetMe conferences it can support? What is the limit of number of
Hi,
I've been having a strange problem recently where real-time asterisk will
unregister a IAX friend at random times when the registration should not have
expired.
I have a Zoiper soft phone client (on windows) connecting to asterisk over a
LAN (no firewalls). The default reregister time of 6
Hi,
We had a last-minute cancellation from Vivox for today's conference.
It happens that someone suggested a guest idea, Howler Technologies
CTO Jay Fenton, who agreed to join the call from the road. Anything
you want to know about transcoding to and from g729 is out topic for
the first hour. My p
Thanks.
I didnt stop that.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: 11 December 2009 11:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Ca
On 11 Dec 2009, at 11:19, Dan Journo wrote:
> Is there any way to write a debug log to disk so that I can check it
> as soon as a call is lost?
> It happens randomly once or twice a day to different callers.
/var/log/asterisk/full?
Most 'standard' setups produce it. Failing that google will re
The info you need is here
http://www.voip-info.org/wiki/view/Asterisk+config+logger.conf
Ish
Dan Journo wrote:
>
> Hello,
>
>
>
> We have a problem that calls seem to be dropping for no reason.
>
>
>
> Is there any way to write a debug log to disk so that I can check it
> as soon as a call
Hi all,
As member of Activa Team I invite all interested in an Asterisk Tapi
Service Provider to integrate with TAPI applications to try activaTSP
and send us feedback in our forum / bugtracker.
Oliver, Activa TSP was developed by ICR and given to the Community to
facilitate the creation o
Hello,
We have a problem that calls seem to be dropping for no reason.
Is there any way to write a debug log to disk so that I can check it as soon as
a call is lost?
It happens randomly once or twice a day to different callers.
Many thanks
Dan
___
-
2009/12/11 marek cervenka
> hello,
>
> there is new version of the best open source TAPI driver for Asterisk -
> Activa 1.6.1
>
> * NEW: Asterisk 1.6 compatibility (partially sponsored by IPEX a.s.
> http://www.ipex.cz)
> * NEW: FEATURE_CODES standardization for AgentACD integration login,
> logo
hello,
there is new version of the best open source TAPI driver for Asterisk -
Activa 1.6.1
* NEW: Asterisk 1.6 compatibility (partially sponsored by IPEX a.s.
http://www.ipex.cz)
* NEW: FEATURE_CODES standardization for AgentACD integration login, logout,
ready, notReady.
* NEW: ActivaTSP x64
Emre Kurnaz wrote:
> Hi everybody,
>
> First of all i am sorry my English :)
>
> i want to configure my asterisk server as a sip server that stores sip users
> in the mysql database connecting directly over odbc driver. My odbc
> configuration works as below
>
> [r...@ao042 asterisk]# isql -v ast
Hello,
How can I go to the Leg B channel in Asterisk Dialplan _after_ call ends?
I can use Dial G option to go to Leb B channel when call is answered, but
how to go here when call ends?
Is here any option/function in Dial Plan?
Or should I use ast_bridged_channel(chan) to get bridged channel an
I am not sure, but I think you will get nothing with those commands if realtime
cathing is not set.
--Original Message--
From: Emre Kurnaz
Sender: asterisk-users-boun...@lists.digium.com
To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
S
Hi everybody,
First of all i am sorry my English :)
i want to configure my asterisk server as a sip server that stores sip users in
the mysql database connecting directly over odbc driver. My odbc configuration
works as below
[r...@ao042 asterisk]# isql -v asterisk
+---
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