Hi,
Is this http://www.voip-info.org/wiki/view/Asterisk+vzaphfc page data
still up to date ?
In other words, is it possible to use One port BRI cards with Dahdi ?
Regards
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source ?
How can I check this ?
2. How can I get rid of these alarms ?
3. Shall I report this ?
4. Waht would you suggest ?
Regards
2012/6/21, Olivier :
> Hi,
>
> After an upgrade, I discovered yesterday strange things I would like
> to share here.
>
> Basically, I'me com
Hi,
After an upgrade, I discovered yesterday strange things I would like
to share here.
Basically, I'me comparing platforms:
The first one is a 2.6.26 (Debian Lenny) platform, with Asterisk
1.6.1.18, Libpri 1.4.10.2, Dahdi revision 8853 (must be between 2.3
and 2.5, I think).
The second one is a
I stricly followed instructions steps 1 and 2 and I'm very to report it works !
Thanks for your detailed answer.
May I post here suggestions that may help others to use this script ?
2012/6/6, Lefteris Zafiris :
> On 06/06/2012 10:47 AM, Olivier wrote:
>> Hi,
>>
>>
Hi,
I recently discovered http://zaf.github.com/asterisk-mstts/ .
In the page above, it is mentioned you have to subscribe to Microsoft
Translator API on Azure Marketplace.
In Azure Marketplace, I found something called "Microsoft Translator".
This API is free within a 2 000 000 characters per on
Hi,
At the moment, I'm mostly using a "Day/Night toggle" button to let
users deal with week-ends, holidays and opening hours.
As Asterisk 1.8 introduces Calendar capabilities, I'm wondering if
better alternatives now exist.
Is it possible, safe, reliable and easy to refer from Asterisk to a
publi
Hi,
I'm facing a strange situation.
Though it's not directly related to Asterisk, I do think it is
interesting to this mailing list.
The setup is a single line which is split between an ADSL
modem/routeur and a fax machine (Asterisk was removed from the
equation).
Any time the fax machine rings
2012/5/14, Matthew Jordan :
>
> - Original Message -
>> From: "Olivier"
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>
>> Sent: Monday, May 14, 2012 4:39:40 AM
>> Subject: [asterisk-users] How to script cdr_
Hello,
I'm using asterisk 1.8.
I can enable cdr_mysql module using "make menuselect".
How can I do that within a shell script, something equibvalent to this:
./configure
echo "MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM" >
menuselect.makeopts.defaults
make USER_MAKEOPTS=menuselec
2012/5/11 Gordon Henderson
> On Fri, 11 May 2012, Olivier wrote:
>
> 2012/5/10, A J Stiles :
>>
>>> On Thursday 10 May 2012, Bart Coninckx wrote:
>>>
>>>> I'm looking for a smaller,
>>>> appliance-type like PC, preferably solid st
2012/5/10, A J Stiles :
> On Thursday 10 May 2012, Bart Coninckx wrote:
>> I'm looking for a smaller,
>> appliance-type like PC, preferably solid state and fanless PC.
>> Since it's only going to run Asterisk for a couple of extensions I don't
>> think CPU and RAM need to be maxed out.
>>
>> Does a
Hi
No firewall on the server
Other idea ?? Hihi
Olivier
Le jeudi 26 avril 2012, Duncan Turnbull a écrit :
> Usually its a firewall issue, or at least it has been for me
>
> Its saying it can't form sip packets, and that will be because something
> isn't letting it,
>
Anyknow know this problems ?
I read on the net that it's a possible network problems, but i don't think
because it's a VMWare server and in the same server i have other
asterisk without this problems.
best regards
Olivier
Le 25 avril 2012 09:35, Olivier CALVANO a écrit :
Perfect that's work ;=)
very thanks
Le 25 avril 2012 10:19, Olivier CALVANO a écrit :
> Ok thanks i test.
>
> I put "match_auth_username=yes" on the two server ?
>
> And for insecure, into the realtime database ? or into sip.conf of the
> second server ?
>
Ok thanks i test.
I put "match_auth_username=yes" on the two server ?
And for insecure, into the realtime database ? or into sip.conf of the
second server ?
best regards
olivier
Le 25 avril 2012 09:34, Leandro Dardini a écrit :
>
>
> 2012/4/25 Olivier CALVANO
>&g
of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not
permitted
anyone know what is this error ?
thanks
olivier
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call with Linksys SPA942 B, i use the context "I-User02" and
the call are sent
to SIP account "USER02" but Server A "Trader" reject the call
immediatly with this error:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have , digest has
[Apr
Hi
No idea ?
thanks
Olivier
Le 24 avril 2012 16:06, Olivier CALVANO a écrit :
> Hi
>
> i have a strange problems on my asterisk server:
>
> I have two asterisk server.
>
> On the first, i use realtime with a MySQL Database,
> i have two user:
> USER01
&g
er:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have , digest has
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device "Olivier"
;tag=as0cd775ab
The exten:
On I-User01: exten => _0XX.,1,Dial(SIP/USER01
Hi Sammy,
Yes my telco have a lot of IP, i receive a call from ~20 ip ..
I can't put a subnet ?
best regards
Le 23 avril 2012 07:57, SamyGo a écrit :
> Hi,
>
>> No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'
>
>
> This line is telling you everything. The peer you've declared isn't be
ist
He search the extention "280" in "default" but not in "Internal"
Anyone know why ?
for information, the 172.16.1.14 is a old asterisk server and i have
put it into calltokenoptional
thanks for your help
Olivier
--
__
Hi,
In DAHDI 2.6.1 changelog includes this :
README, drivers/dahdi/Kbuild: Build OSLEC EC if in the tree Build
the OSLEC echo canceller (drivers/staging/echo and
dahdi_echocan_oslec) if the code of oslec is present in the tree.
Also closing another issue regarding do
Apr 21 08:01:16] NOTICE[11906]: chan_sip.c:18527
handle_request_invite: Call from '' to extension '331NUMNOFOUND'
rejected because extension not found.
a idea of the problems ?
My supplier use a lot of server, i thi
, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote:
>> Hi
>>
>> can i don't sent into the SIP invite the "Session Timer" ? on asterisk 1.6
>
> Have you tried 'session-timers=refuse' ?
>
> --
> Barry
>
> --
> _
Hi
can i don't sent into the SIP invite the "Session Timer" ? on asterisk 1.6
Best regards
Olivier
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Hi
greats thanks that work very good
Olivier
Le 16 avril 2012 12:47, Stuart Elvish - IP Exchange Systems
a écrit :
> Hi,
>
> If you are using IAX and a later version (I know it works in 1.8.x) you
> can use IAXVAR.
>
> The following URL has a post which has a good
Hi,
Which free or non-free (as beer) Sugarcrm plugin would you recommend
to add click to dial feature with asterisk ?
I can see a quite long list of such plugins but not all of them seem
up-to-date (judging by comparing with latest Sugarcrm version number).
Regards.
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_
LLERID(num){0:4}})
> to this
> exten => _x,2,Verbose(passed ID ${$CALLERID(num)})
> exten => _x,3,AGI(MyScript.agi,${$CALLERID(num){0:4}})
>
> and post your CLI output. We need to see if the OP's suggestion is getting
> to Asterisk #2.
>
> -Original Message---
i am search on google ;=) but no result for this moment hihi
Le 15 avril 2012 21:14, Olivier CALVANO a écrit :
> Very thanks for your help, but no, it's not good
>
>
> Le 15 avril 2012 20:54, Danny Nicholas a écrit :
>> I believe they were trying to say
>> e
ium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
> CALVANO
> Sent: Sunday, April 15, 2012 1:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Set variables from one asterisk ta a second.
>
> Hi
>
>
Hi
Thanks for your help but i don't know this variable: $CALLID[1-4]
it's correct:
exten => _x,2,AGI(MyScript.agi,${$CALLID[1-4]})
?
best regards
olivier
Le 15 avril 2012 12:55, Administrator TOOTAI a écrit :
> Le 15/04/2012 10:44, Olivier CALVANO a
t it into a AGI script.
It's possible ?
thanks for your help
Olivier
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Le 14 avril 2012 11:30, Ben WIlliams a
écrit :
> This is a really simple problem that I just can't get to work. There
> are two Asterisk servers with the following sip user and peer. When a
> call is attempted, Asterisk
Which instance are you talking about, here ?
> is not sending authenticat
2012/4/11, Satish Barot :
> I would implement it in a different way.
> As you seem to be a seasoned player just a hint here.
> How about adding local channels as queue members and executing agi in local
> channel context before actual dial()?
Just a thought: does this comply with dynamic agents fe
Hi,
May I ask this off-topic question ?
I've got an Asterisk 1.8/Freepbx 2.10 install on a Squeeze server.
No matter which language is selected with the top right corner
scrolling list, the GUI remains in english.
In my system, there are plenty of amp.po localized files, here and there.
Any hint
Hi,
On a 1.8.10 system, I've got (with cdr show status) :
Call Detail Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: No
* Registered Backends
---
csv
cdr-c
Hi,
I would be curious to learn about experiences with Polycom-Kirk 6000
and DECT/GAP handsets (mostly Gigaset handsets of all kinds) in a
multi-cells environment.
More precisely, what about roaming and handover ?
Regards
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Hi,
Does Cisco 79XX with SIP firmware support asterisk's BLF ?
Has someone been successful with this ?
Regards
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Hi
i am search a solution for "change" the number called.
Sample:
I have a Linksys SPA942 connected in SIP with my server.
When this phone call a number: 043112
automatiquely change in 3343112
because my carrier want a number in international format.
It's possible ?
Hi,
For training sessions, I'm evaluating the possibility to use a single
physical server to host 5 virtual servers, each with its own Dahdi
PCIe card, instead of using 5 physical machines, hoping a single
physical server would easier to transport, more quiet and cheaper to
provision and maintain.
Thanks but i read:
"; The maximum number of concurrent calls you want to allow"
Not limit the duration of a call ;=)
Le 2 avril 2012 16:55, Bakko a écrit :
> Hi,
>
> look at maxcalls parameter on the asterisk.conf file.
>
> regards
>
> El 02/04/2012 16:46, Oli
Hi
it's possible into Asterisk 1.6.x to limit a call at 120 mn ?
after 120mn, hangup and the customer call a new time
thanks
olivier
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2012/3/22, John Knight :
> I've tried this in the past and while FreePBX and its base modules work
> fine in an http proxy environment, some applications like fop2 fail to
> connect properly as they obviously rely on direct connections via ajax
> using the browser as a client.
>
> That said, I've
Hi,
Tough Freepbx is not the main focus of this list, may I ask if Freepbx
and its End Point Manager module can work in an environment with an
HTTP proxy ?
In my testing, everything works OK but one thing: I can't upload End
Point product list :
in End Point Configuration tab, when I click over
2012/3/21, Stefan Schmidt :
> Am 20.03.12 10:15, schrieb Olivier:
>> Hi,
>>
>> I would like to test the following COLP use case :
>>
>> Alice and Bob are both using a SIP phone registered on a Asterisk 10
>> server.
>> Alice dials Bob's extension.
Hi,
I would like to test the following COLP use case :
Alice and Bob are both using a SIP phone registered on a Asterisk 10 server.
Alice dials Bob's extension.
While Bob's phone is ringing, Asterisk updates Alice phone screen with
Bob's name, so that at a glance, Alice can check she dialed the
c
2012/3/13, resea...@businesstz.com :
> I am struggling to get the mac-addresses of IP phones that are connected
> to asterisk as the phone are in different VLAN with * and they were
> manually configured. I want to centralize their configuration using
> res_phoneprov or tftp
>
> I have tried nmap a
Hi,
I've got the following setup:
PSTN/ISDN < E1-> Asterisk < E1-> Alcatel 4400 PBX
<> TDM phones
When a TDM phone is dialing out to a national number, it seems that
the PBX is using enbloc dialing.
When a TDM phone is dialing out to an international number (variable
length
2012/2/16, Administrator TOOTAI :
> Le 16/02/2012 16:55, Olivier a écrit :
>> Hi,
>>
>> I've read here and there how Asterisk could send SMS but I didn't find
>> much about how to receive SMS and forward them to an email box.
>>
>> 1. First of all
Hi,
I've read here and there how Asterisk could send SMS but I didn't find
much about how to receive SMS and forward them to an email box.
1. First of all, I don't think my telco would let me receive any SMS
my landline.
2. Maybe I could find providers selling this service for a monthly fee;
3.
the answer would be a definite 'it depends' ;-)
>
> -d
>
>
> On Feb 15, 2012, at 3:03 PM, Olivier wrote:
>
>> Hi,
>>
>> When someone says "T.38 is not reliable on a (normally loaded and
>> managed) LAN", would you rather agree or disag
2012/2/15, Tim Nelson :
> - Original Message -
>> Hi,
>>
>> When someone says "T.38 is not reliable on a (normally loaded and
>> managed) LAN", would you rather agree or disagree ?
>> In this case, fax calls are coming in through an analog gateway,
>> passing trough Asterisk and then going
Hi,
When someone says "T.38 is not reliable on a (normally loaded and
managed) LAN", would you rather agree or disagree ?
In this case, fax calls are coming in through an analog gateway,
passing trough Asterisk and then going out to ISDN through a digital
gateway.
Comments ?
Regards
--
Hello,
For backup, I'm looking after cheap ISDN phones I could use in BRI PtP
or PtmP lines to forward incoming calls (typing a DTMF sequence).
Which model and brand would you recommend for this ?
Regards
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@CALL${cnam}~${cnum}|nc -w 1
> ${IP_of_screenpop_user} 10629)
>
> That code hasn't been tested -- it's just an example.
>
It's interesting : I'll give it close look !
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:aster
Hi,
For an RFP, I need to implement screen popup where caller names are
searched in outlook folders.
I would both consider free or paid solutions.
Regards
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2012/2/10, Jason W. Parks :
> I'm in a similar situation. However, most of my buildings were re-wired
> around 1994 to provide Cat5 or 5E to the desktop for data, and 2-pair
> Cat3 for voice, all in a star topology. I can move my voice
> infrastructure to an IP-based one running 10Mbps, utilize exi
one.
I agree that if a caller requires anonymity, then, except for very
special cases (emergency services, for instance), ANI should also be
hidden by the telco.
>
>
>
> Greetings from Wuppertal
> Max Grobecker
>
>
> Am 09.02.2012 11:12, schrieb Olivier:
>> 2012/2/8
2012/2/8, Kevin P. Fleming :
> On 02/08/2012 12:40 PM, Olivier wrote:
>> 2012/2/8, Kevin P. Fleming:
>>> On 02/08/2012 10:06 AM, Carlos Alvarez wrote:
>>>>
>>>> On Wed, Feb 8, 2012 at 2:35 AM, Olivier>>> <mailto:oza_4...@yahoo.fr>> wrote
2012/2/8, Carlos Alvarez :
> If the customer is so cheap that they won't properly build out the network,
> why would they have gigabit switches to the desktop which have a limited
> set of applications that actually benefit from it?
>
> Then there's PoE, which is expensive to start and very expensi
2012/2/8, Kevin P. Fleming :
> On 02/08/2012 10:06 AM, Carlos Alvarez wrote:
>>
>> On Wed, Feb 8, 2012 at 2:35 AM, Olivier > <mailto:oza_4...@yahoo.fr>> wrote:
>>
>> I always thought that ANI (Automatic Number Identification) could not
>> direc
2012/2/8, Vieri :
> I'm trying to understand why vendors keep making 100Mbps integrated 1-port
> switches in their hardware SIP phones. Even the recently-announced D40 and
> D50 Digium phones are limited to 100Mbps. Only the more expensive models
> (like the D70) can run at 1000Mbps.
> However, you
Hi,
This morning I called an analog line from a cell phone.
For the first call, I made a standard call and could both read my CID
and ANI in my Asterisk console.
For the second call, I choosed to hide my ID and, to my surprise, I
read that my CID and ANI were both empty.
I always thought that AN
2012/1/31 John Knight
> Personally, I don't think what Digium is doing is necessarily a perfect
> approach (hey, what is? we're all human), but they've vastly improved the
> quality of Asterisk from a support perspective.
>
>
I also agree that IMHO, Asterisk quality has vastly improved.
Though n
2012/1/29 Mike Diehl
> Hi all,
>
> I'm working with the Digium fax for Asterisk product, which is working
> pretty
> reliably for me.
>
> However, the sendfax application isn't sending status events to AMI. The
> receivefax application does.
>
> For example, with this call file:
>
>
Great !
I'll test it ASAP and report back here (tomorrow, if possible).
2012/1/23, Alec Davis :
>
>> How can I test this solution on a 1.8.8.1 system ?
>> If I'm not mistaken, diff
>> https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1.
>
> I've just checked out 1.8.8.1 and download my
2012/1/20, Alec Davis :
> This maybe not what you want.
>
> Our solution was monitor a queue with a BLF, instead of a queue member
>
> This review https://reviewboard.asterisk.org/r/1619/ allows a BLF lamp to
> flash when a queue is ringing, then the queue can be picked up by the BLF
> button.
>
>
Which tool would you then pick read or analyse those ISDN, Q.SIG, SS7
pcap files ?
I took quick look at Wireshark capabilities but I'm still not sure
about its relevance.
2012/1/20, Tony Mountifield :
> In article <4f18a763.90...@digium.com>,
> Kevin P. Fleming wrote:
>> On 01/19/2012 05:25 PM, T
; There you have also a changelog documenting the differences. Steve Underwood
> commit here the latest changes in spandsp source.
>
> http://fisheye.freeswitch.org/changelog/freeswitch.git/libs/spandsp
>
> Kristijan
>
> 2012/1/11 Olivier :
>> Hi,
>>
>> Mayb
Are both A and B extensions of the same Asterisk system or is A an
inbound caller ?
2012/1/16, Gunnar Schaller :
> Hello Eric,
>
>> See http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE pay
>> special attention to the sendrpid note.
>
> That does not work. CONNECTEDLINE is for answere
Hi,
Freepbx includes a fax_process.pl which convert TIF files into PDF
files before sending by email.
I'm used to use sSMTP with Asterisk.
I'm certain ssmtp is correctly configured in my (Debian Squeeze) setup
as I'm correctly receiving voicemails in email box.
Is it possible to tell fax_process
I've never done it myself yet but I think I would look after COLP
function (1.8 and above).
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2012/1/16, bakko :
> http://www.voip-info.org/wiki/view/Asterisk+local+channels
I don't know why but I was thinking of some sort Dial app magic and
didn't look after Local channels options.
Thanks for correcting me.
>
> Regards
>
> - Original Message -
> From
Hi,
Where to find meaning of /n in Local/6613@from-queue/n ?
Regards
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libjpeg.so.62 => /usr/lib/libjpeg.so.62 (0xb750e000)
libz.so.1 => /usr/lib/libz.so.1 (0xb74fa000)
/lib/ld-linux.so.2 (0xb77a2000)
So, with those 2 commands, I couldn't directly check the link between
asterisk and spandsp, and check am I'm really using spandsp0.0.6pre1
Hi,
I've recently upgraded a system from 1.8 to asterisk 10 and also
updated spandsp while doing so.
I wondered what is the safest and easiest way to check from command
line which libraries a running Asterisk system is currently using
(just like "dahdi show version", for instance).
Though I'm cur
2012/1/12, Kevin P. Fleming :
> On 01/12/2012 06:39 AM, Olivier wrote:
>> Hi,
>>
>> I'm having some questions related to echo cancellation configuration
>> on a Digium board enabled systems (B410P, TE420, TE420B, ) for
>> cases when a hardware ech cancel
Hi,
I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech canceller is present or not.
I read in TEXXX manual that when setting echocancel=yes in
chan_dahdi.conf on a VPMOCT64-equiped syste
latest
> pre18.
T.38/T.30 gatewaying can tricky enough to configure so moving this
library version choice out of the equation is a great step forward for
me.
I'll go with pre18, as also suggested elsewhere.
Thanks for sharing this !
>
>
> *José Pablo Méndez
> *****
>
I didn't know spandsp could do 14.4k : that's great !
So, now I'm realizing how strange my previous question could resonate.
Thanks for clarifying this.
2012/1/11, Steve Underwood :
> On 01/11/2012 11:16 PM, Olivier wrote:
>> 2012/1/11, Steve Underwood:
>>> On 0
2012/1/11, A J Stiles :
> On Wednesday 11 January 2012, Olivier wrote:
>> At the time I first wrote my question, libcurl4-openssl-dev was
>> missing from my system so func_CURL was not available, which lead me
>> check with wiki.asterisk.org.
>
> It's *always* a
Yes, thanks to Jim's help, I added a missing library, recompiled and
then I could also see CURL function presence in my system.
Forgetting this library made me discover the missing entry in
wiki.asterisk.org 1.8 section.
I opended a ticket for it.
Cheers
2012/1/11, Benny Amorsen :
>
2012/1/11, Steve Underwood :
> On 01/11/2012 03:01 PM, Olivier wrote:
>> 2012/1/5, Kevin P. Fleming:
>>> On 01/04/2012 12:25 AM, Matt Darnell wrote:
>>>> Aloha,
>>>>
>>>> We are looking to roll a solution that will have the following network
2012/1/11, Jim DeVito :
> I think the wiki may have just missed func_curl. I have a couple 1.8.x
> machines with working func_curl. Have you tried to compile it anyway?
At the time I first wrote my question, libcurl4-openssl-dev was
missing from my system so func_CURL was not available, which lead
7;m asking because func_CURL is missing from
https://wiki.asterisk.org/wiki/display/AST/Dialplan+Functions
(asterisk 1.8 version) which is misleading.
2. How would you rate CURL function performance ? Would you recommend
it (for CID Lookup, for instance) ?
2012/1/11, Benny Amorsen :
> Ol
Hi,
Which plugin can I add to my iceweasel browser (debian squeeze) to
play gsm sound files ?
Cheers
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Hi,
I've seen that function CURL is missing from 1.8 but back in with 10
(see wiki.asterisk.org).
With asterisk 1.8 and above, for a custom CID Name lookup application,
which is the most efficient way to send an HTTP GET from the dialplan
and parse its response (code and content) ?
Regards
--
_
2012/1/5, Kevin P. Fleming :
> On 01/04/2012 12:25 AM, Matt Darnell wrote:
>> Aloha,
>>
>> We are looking to roll a solution that will have the following network
>> layout:
>>
>> ISDN-PRI<--> Asterisk<--> T.38<--> ATA<--> Fax
>>
>> Does version 1.8 with the Digium fax driver have this capabilit
Hi,
Maybe I missed it while checking it, but which spandsp version is
recommended to play with Asterisk 10 and T.38/T.30 gatewaying ?
I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here
(http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a
changelog documenting
2012/1/10, Olivier :
> Hi,
>
> 1. This patch didn't correct the issue but I'm far from certain that I
> correctly applied the patch.
I was right to suspect I was wrong : now, after correctly applying
the DAHLIN-275 patch, it's working OK (with the EchoCan module
p
HLIN-275 ticket.
2012/1/9, Shaun Ruffell :
> On Mon, Jan 09, 2012 at 01:47:48PM -0600, Shaun Ruffell wrote:
>> On Mon, Jan 09, 2012 at 07:52:02PM +0100, Olivier wrote:
>> > Hi,
>> >
>> > On a brand new system, I met an issue I've never met before.
>>
Hi,
On a brand new system, I met an issue I've never met before.
My setup is :
debian 6.0.3
asterisk 1.8.8.1
dahdi 2.6.0
libpri 1.4.12
freepbx 2.9.0.4
TE420FB (with hardware EC)
This is the very first time I'm using Freepbx and the whole
configuration was first generated by a "make samples" comm
Hello,
Which QSIG (ECMA or ISO) variant and profiles does asterisk support ?
(I could not find this info inside chan_dahdi.conf)
Regards
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New to Asterisk?
2012/1/3, Marco Mooijekind :
> Dear all,
>
> I have the following challenge using Asterisk 1.8, using a Digium B410P
> card on BRI (The Netherlands, KPN ISDN) .
>
> DAHDI is running, dahdi_tools indicates OK on my span and light on back of
> card is green.
>
> However, in Asterisk i get the followi
MDL/TEI managemement message, but configured for mode other
than PTMP!
As you may guess, to my knowledge, I don't have no "span 0" so this
seems to show a bug.
I'll report here further progress on this.
Cheers
2011/12/9, Shaun Ruffell :
> On Fri, Dec 09, 2011 at 04:52:53
patience and guidance!
>
>
> On Dec 29, 2011, at 8:40 AM, Olivier wrote:
>
>> OK !
>> But AEL2's ifTime keyword do not use it, does it ?
>>
>> 2011/12/28, Mindaugas Jasiulis :
>>> Hi,
>>>
>>> This function sets TESTTIME global vari
OK !
But AEL2's ifTime keyword do not use it, does it ?
2011/12/28, Mindaugas Jasiulis :
> Hi,
>
> This function sets TESTTIME global variable and if TESTTIME variable is set,
> then GoToIfTime use time from this variable.
>
>
> On 2011.12.28, at 17:28, Olivier wrote:
Hi,
Thanks for replying.
I'm afraid this :
[foobar]
exten => 123,1,Verbose(0,Into context ${CONTEXT})
exten => 123,n,Verbose(0,Time is ${STRFTIME()})
exten => 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
exten => 123,n,Verbose(0,Time is ${STRFTIME()})
exten => 123,n,HangUp()
... give
2011/12/27, Eric Wieling :
> We are running 1.8.8.0.
>
Then the issue you're having differs from the one I had (which
appeared and disappeared instantly when I upgraded to 1.8.7 and 1.8.8
respectively).
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-- Bandwidth and Colo
Could you try with 1.8.8.0 ?
I think this one includes a fix for that error.
2011/12/26, Joseph :
> On 12/23/11 10:40, Eric Wieling wrote:
>>I'm getting various codec related warnings after upgrading to 1.8. Did I
>> miss something in the UPGRADE file? Does Asterisk no longer transcode
>> 8-)?
>
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