Hi. I have seen these kind of instructions but there i lost it.
here is what i got.
Asterisk has a spa3102 to interface the PSTN line. It works smoothly and i
got in/outgoing calls. i do have the codec to g711alaw (since i am in
Europe). on the fxs port of the spa3102 i had the fax machine up to
Hi Matt
Thanks for your response. I have tried with two GXV3175 with same result.
Let me dig deep on this to find out the route cause
Sam
Matthew Jordan wrote:
On Thu, Jun 13, 2013 at 12:04 PM, resea...@businesstz.com wrote:
Hi there
I have asterisk 10.11.1 which seems to have problem
Hi,
some month ago we installed a VoiceRec-Module from Vestec
(https://www.vestec.com/) on Asterisk 11.x. It works so far and you will
find examples for your dialplan. It should be ok for your needs.
-Thorsten-
Am 13.06.2013 23:19, schrieb asterisk users:
Hello list,
'Just wondering if
Hello,
I'm trying to execute a stored procedure on a MSSQL Server from the dial plan,
but it's not working. I'm getting the following error: Unable to execute
query
Asterisk has been compiled with UnixODBC, and I've done the necessary
configurations in func_odbc, res_odbc and odbc.ini.
Note, that writing CDRs using ODBC to a MSSQL database does work. So I don't
know why this doesn't.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian
Sent: Friday, June 14, 2013 2:43 PM
To:
Hello Everyone,
I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP model.
We are looking to interconnect with the PSTN world, and our supplier
has given us
a few options. We can either do this over traditional PRIs, A-Links or
the SS7IP new.
I am really interested in SIGTRAN,
I think you need a SIGTRAN stack from Netfors or LeibICT.
Mitul
On Friday, June 14, 2013, Nick Khamis wrote:
Hello Everyone,
I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP
model.
We are looking to interconnect with the PSTN world, and our supplier
has given us
a
Hello Mitul,
Thank you so much for your response. During the testing phase
we would like to employ an open source solution, and wanted
to know what people have had success with, given the different
user part etc..
On a side note, anyone know of service providers offering SIGTRAN?
Kind Regards,
There is no open source solution for SIGTRAN yet.
If you come across one, do let everyone here know about it.
You can however request some time restricted demo from Netfors or LeibICT.
Mitul
On Friday, June 14, 2013, Nick Khamis wrote:
Hello Mitul,
Thank you so much for your response.
Howdy All,
They say opinions are like belly buttons, everybody has one. (that's
the clean version of the saying). So I'm asking for yours. I hope you
see it as a fun exercise.
I'm designing a phone system from the ground up. Will be about 1000-1300
seats mixed 80/20 VoIP/Analog. 58-acre
If they will do atm over oc-n, perhaps that would work better.
Ie, a perm virt circ for SS7 and as-needed vc's for ulaw.
Atm oc-n cards with linux sw support are widely available, according to goog.
libss7 and and ast *might* need a bit of patching to work with it, but it
shoudn't take too
On Thu, 2013-06-13 at 18:29 -0600, Joseph wrote:
When I play:
exten = s,n,Background(welcome)
and press extension 1 the system will not jump to this extension
immediately, there is a few sec. pause.
Mine looks like this:
exten = s,1,Answer()
exten = s,2,Set(TIMEOUT(digit)=1)
exten =
I'm trying to to to dial1 if caller id match:
but dial plan execute 220,n(dial1) regardless
exten = 220,n,GotoIf($[${CALLERID(number)} = 7804792668]?dial1)
exten =
220,n(dial1),Dial(${sales_support}${accounting}${family},25,m(penguin)w)
exten = 220,n,
I was under impression that if
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday, June 14, 2013 2:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] GotoIf($[${CALLERID(number)}
I'm trying to to to
Hello James, thank you so much for your response!
On 6/14/13, James Cloos cl...@jhcloos.com wrote:
If they will do atm over oc-n, perhaps that would work better.
Yes they will do atm over oc-n only not sure if they will ring or spur it...
Ie, a perm virt circ for SS7 and as-needed vc's for
I did install hylafax and iaxmodem. Everything installed correctly and
the services do run OK. The problem is the sip.conf and extensions.conf.
I do not know how to link the PSTN number 123456 to work as fax and then
send as email.
Hi. I have seen these kind of instructions but there i
On Fri, Jun 14, 2013 at 09:43:29AM -0600, Nunya Biznatch wrote:
System will use G.722 for VoIP Phones.
[...]
2-servers acting as gateways. Each handling 2 PRIs for outside
trunks.
So why use g722? Just use your local g711 law and thus avoid the
transcoding impact to/from the PSTN and calls
http://red-fone.com http://red-fone.com/products-new/fonebridge/ might be
a good place look and see if other ideas pop up. They have good products.
I am not affiliated with them, just a happy user on a couple of
deployments.
On Fri, Jun 14, 2013 at 11:43 AM, Nunya Biznatch
Another option instead of 2 servers dedicated as PRI gateways is to use
AudioCodes Mediant 1000 or 2000 gateways. Either of them will also
failover to a backup proxy if the primary proxy (server) is offline.
Probably much cheaper than the kick ass box you plan to build + PRI
card(s).
I'm not
On Thu, Jun 13, 2013 at 12:29 PM, vortex binary.vor...@gmail.com wrote:
Hello. i am running debian 6 with asterisk 11.4. The system has exim4 to
send to email the voicemails.
i would like to get rid of the analog fax machine and use asterisk to
send/receive faxes.
I do have a PSTN line with
Let 123456 be your fax number, iaxmodem0 the account name of your IAX
modem (configured in iax.conf), then in extensions.conf you would have
something like
[from-pstn...]
exten = 123456,1,Verbose(1,Incoming fax...)
same = n,Dial(IAX2/iaxmodem0,40)
same = n,Hangup()
and for outgoing
On 06/14/13 18:43, Noah Engelberth wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday, June 14, 2013 2:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Is anyone using Homer from sipcapture.org or anything like it for capture
sip traffic for debuging? If so what are your experiences.
Thanks
-Bryan Anderson
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