> Now, referring to the error above, I see (in voip-info.org) that
> t38passthrough is an R/O variable and not an R/W, but in any case, I got
> 0 as a result, so it should have been OK, and it's not, as ReceiveFAX
> still sends a T.38 reINVITE. If I can't modify it, what should I do?
For the tes
> I have no idea where you got the idea that such a thing is possible...
> it's not. sip.conf settings for SIP endpoints are not channel variables,
> and cannot be modified from the dialplan unless the CHANNEL() dialplan
> function has been specifically extended to support them.
I was actually HOP
Hello,
We're trying to receive G.711 (aLaw) faxes on the asterisk and convert
them to tif. With T.38, we have several issues, so we are trying to use
G.711, since the gateway is located in the same LAN, so there's no
bandwidth/packet-lose issue.
We also use on the same Asterisk Real-Time proce
> Cyprus VoIP wrote:
>
>> This is the reINVITE SDP received from the SIP Proxy:
>> ---
>> Content-Type: application/sdp
>> Content-Length: 353
>>
>> v=0
>> o=root 30427 30428 IN IP4 194.98.xxx.xxx
>> s=session
>> c=IN
al Message
Subject: Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9
From: Cyprus VoIP
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Friday, 04 December, 2009 18:21:59
>> It's probably because you are using 1.6.1.9; that release (and older)
>> had a &
>> We're trying to receive faxes on the Asterisk server, but for the time
>> being T.38 negotiation fails.
>>
>> The SDP that the Asterisk reINVITE sends contains these lines:
>> --
>> m=image 4968 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:9600
>> a=T38FaxFillBitRemoval
Hello,
We're trying to receive faxes on the Asterisk server, but for the time
being T.38 negotiation fails.
The SDP that the Asterisk reINVITE sends contains these lines:
--
m=image 4968 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval
a=T38FaxTranscodi
Users Mailing List - Non-Commercial Discussion
Date: Thursday, 10 December, 2009 05:26:07
> On Dec 9, 2009, at 10:15 PM, Cyprus VoIP wrote:
>
>> Hello,
>>
>> We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would
>> like to use the sip,extens
Hello,
We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would
like to use the sip,extensions and voicemail in realtime mode.
Where can we find the database tables structure for these versions?
Thanks,
Andreas
___
-- Bandwidth and Colo
> It's probably because you are using 1.6.1.9; that release (and older)
> had a 'feature' that allowed automatic switching back to audio from T.38
> if one of the endpoints sent an audio packet. It turns out that wasn't a
> good idea, and it's been removed... but in later versions. You'll have
> to
> Cyprus VoIP wrote:
>
>> So, I enabled the full logger, and the strange thing I see is this message:
>> "Got T.38 Re-invite without audio. Keeping RTP active during T.38 session"
>>
>> It seems that this might be the reason Asterisk initiates a reINVITE
> Set 'canreinvite=no' on all applicable peers?
>
I tried with yes and no. No difference. I'm almost certain it's related
to the "Keeping RTP active during T.38 session" issue.
___
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> Cyprus VoIP wrote:
>
>> Thank you for your answer. The 'internal extension' is indeed a T.38
>> capable device that works perfectly when connected directly to the
>> Proxy/ITSP.
>>
>> As you said, the key to debugging/resolving this issue is the
>> We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the
>> local extension and remote Proxy, but it still forces the call to go
>> back to a voice call.
>
> Define 'internal extension'. Is this a T.38-capable device? If not,
> Asterisk doesn't support TDM-to-T.38 FAX relay (yet
Hello,
We are trying to send faxes by T.38 protocol to a remote SIP proxy from
a local extension. The local extension sends the INVITE, Asterisk sends
the call to the Proxy the call is connected with a regular audio codec.
After a few seconds the remote proxy sends an INVITE with UDPTL and the
>> The problem is that the online module update is not working for me
>> (Cannot connect to online repository (mirror.freepbx.org). Online
>> modules are not available.) and I couldn't find online a working
>> solution :-(
>
> DNS/Gateway ok on server?
>
Yes. The problem is with the FreePBX modul
>> I tried to install Asterisk + Asterisk addons + FreePBX (latest
>> versions
>> of all), but in the FreePBX screen, I don't have the option to set
>> ring
>> groups and IVRs
>>
>> Can anyone tell me what I'm doing wrong?
>
> You are not posting on the FreePBX forums? ;)
>
I figured "Asteris
Hello,
I tried to install Asterisk + Asterisk addons + FreePBX (latest versions
of all), but in the FreePBX screen, I don't have the option to set ring
groups and IVRs
.
Can anyone tell me what I'm doing wrong?
Thanks,
Andreas
___
-- Bandwidth and C
bject: Re: [asterisk-users] Music On Hold
From: Cyprus VoIP
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Saturday, 03 October, 2009 09:28:20
>
> > What does your musiconhold.conf look like?
> >
>
>
> [general]
&
> What does your musiconhold.conf look like?
>
[general]
[default]
mode=files
directory=/var/lib/asterisk/moh
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now:
>
> What is the output of "moh files show" CLI command ?
>
pbx*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity
File: /var/lib/asterisk/moh/macroform-robot_dity
Fi
an III
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Wednesday, 30 September, 2009 15:27:28
> On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote:
>
>> > You see the wav files but do you see the files encoded for the codecs
>> you are using?
>>
Hello,
Has anyone encountered that when Asterisk sends RTCP messages, it stops
sending RTP packets until it gets an answer?
Can that be fixed?
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - Octo
> I'm afraid I can't be much help as I am both a newbie and it works just
> fine for me on 1.6.1.6. Of course, mine was a fresh installation.
Thanks for your help, John. Mine is also a fresh installation, but now
at least I know it's not a version issue.
> Is there anything in the logs to gi
ne knows how to debug/fix it, your help would be HIGHLY
appreciated. We're really stuck.
Thank you all in advance.
Original Message
Subject: Music On Hold
From: Cyprus VoIP
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Tuesday, 29 September, 2009 14:31:
Hello,
We need help in debugging Music On Hold on our Asterisk 1.6.1.6
From the SIP debug, I see that an extension sends an INVITE of the call
to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but
I don't see in the console any reference to the call being placed on hold.
Whe
Hello,
I see that there's 1.6.0.x and 1.6.1.y versions of Asterisk.
Is there a clear table that describes the features and/or differences
between them?
Are both stable enough?
Is T.38 Fax supported on both? If yes, which spandsp is supported? I saw
on voip-info.org that version 6 is not suppo
Hi,
Is there anyone there that installed successfully the CRI package and
manages to play the calls listed in the "call monitor" page?
Regards.
Original Message
Subject: Re: [asterisk-users] Crystal Recording Interface
From: Cyprus VoIP
To: Asterisk Users Mailing
ct, everything will be as expected. This might not be a fun hill to
> climb.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP
> Sent: Friday, August 28, 2009 10:37 AM
> To: Ast
ssion
Date: Sunday, 30 August, 2009 19:27:46
> Cyprus VoIP wrote:
>> I think that the missing component is mysqlclient, but when i "yum
>> update mysql", it does nothing.
>>
>>
>
> You need to make sure that mysql-devel is ins
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Sunday, 30 August, 2009 18:58:48
yum search mysql client
yum install 'TheClientYumHasReturnedForYourSystem'
Olivier
Cyprus VoIP a écrit :
I think that the missing component is mysqlclient, but when i "
Asterisk Users Mailing List - Non-Commercial Discussion
Date: Sunday, 30 August, 2009 18:28:35
> You have to fix the dependency issues, which means install the stuff
> you are missing that cdrmysql depends on so u can recompile it.
>
> Sent from my iPod
>
> On Aug 30, 2009
hanks.
Original Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Tilghman Lesher
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Sunday, 30 August, 2009 17:17:59
> On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:
>> Hello all,
>>
Hello all,
I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but
without success.
Is there a proper online manual that describes all the steps to follow
and debugging/monitoring information?
When I type in the CLI "module show", cdr_addon_mysql.so is not listed,
although in m
sk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP
> Sent: Friday, August 28, 2009 9:53 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Crystal Recording Interface
>
> He
Hello all,
I download from Tikal's site the Crystal Recording Interface and
installed it on my Asterisk server, but there's no reference in the
installation instructions there regarding the necessary settings on the
Asterisk itself.
Is anyone using it? Any detailed explanation on the impleme
m: Philipp Kempgen
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Tuesday, 28 July, 2009 14:10:55
> Cyprus VoIP schrieb:
>> I ran into this problem: When I change the CALLERID(num and name) to
>> anonymous, they are also changed in the RPID line and not only in the
ist - Non-Commercial Discussion
Date: Monday, 27 July, 2009 17:16:45
> Cyprus VoIP schrieb:
>> I would like to use Asterisk to add/modify SIP headers in the INVITE
>> message, to include Privacy information, if the INVITE includes a *67
>> prefix (or another predefined prefi
Hello all,
I would like to use Asterisk to add/modify SIP headers in the INVITE
message, to include Privacy information, if the INVITE includes a *67
prefix (or another predefined prefix).
That's an example of the INVITE I get:
/INVITE sip:*6700112233...@192.168.1.100 SIP/2.0
From: "123456789
Hello,
I would like to add SIP headers to the REGISTER messages Asterisk (1.6)
sends to an external proxy.
Also, I want to be able to reorder the lines.
Is it possible?
If yes, how?
Thanks.
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