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 no
hi:you have to install libpri,dahdi and asterisk for E1 cards.
Best regards,
James.zhuVega VOIP gateways, Sangoma Asterisk cards, SBC, NetBorder VOIP
Gateway, LYNC-TDM, Transcodingwebsite: www.hiastar.com
> From: longst...@gmail.com
> Date: Thu, 13 Jun 2013 10:31:28 +0200
> To: asterisk-users@
Yeah, probably wouldn't work too well in a business environment where
you actually NEED to answer calls. I go to a lot of trouble to make
sure people can't get in touch with me. :)
I keep my blacklist and whitelist in AstDB. However, I maintain it in
a bash script so that I can update the scrip
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.
I think because I have an extensions "1" and "11" in my system.
Is there a way to tell "Background" to execute the first match?
I see there a
On Thu, 2013-06-13 at 18:14 -0400, Eric Cooper wrote:
> Greg, would you mind posting your dialplan?
It may be a day or two before I can do that, as of course I will need to
sanitize it (remove passwords, commented lines, etc.)
--Greg
--
__
On Thu, Jun 13, 2013 at 02:32:22PM -0600, Greg Woods wrote:
> I do this, but without any white or black lists, and it works great. The
> greeting says "press one for , or two for ". That alone is
> enough to knock out virtually all the spammers (99% of them are
> robo-calls these days). Once 1 or 2
Hello list,
'Just wondering if anyone can point to a very light-weight and easy to
incorporate into Asterisk (v. 11.x) to handle a minimal set of responses,
like:
0 - 9
yes
no
(maybe * and # for some people)
The idea is that within an IVR menu, the caller could respond by speaking
to
Carlos Chavez wrote:
>
> I have been struggling with an audio issue for a week now and have
> not been able to solve it.
>
> We have an Asterisk server (now running 11.4 but started with 1.8)
> with several sip phones on an internal network and a SIP trunk for
> external calls. We recently put sev
On Thu, 2013-06-13 at 13:55 -0600, Joseph wrote:
> Good idea, I like your approach with press "number" to leave a message", this
> will definitely cut the robo-calls voice-mail.
I do this, but without any white or black lists, and it works great. The
greeting says "press one for , or two for ".
I fuzzily recall someone posting a script that shuffled off voicemails
to Google for conversion to text that could then be emailed. Anyone
have any luck with that? Anything new out there?
j
--
_
-- Bandwidth and Colocation
Maybe this can help:
http://ofps.oreilly.com/titles/9781449332426/asterisk-Fax.html
Best.
2013/6/13 vortex
> 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/
Thank you for input.
Good idea, I like your approach with press "number" to leave a message", this will definitely cut the robo-calls voice-mail.
Do you use database for white-list?
Can you post a section of your dial plan that deals with blocking?
This is a medical clinic so white-list, black-
Google the number and you can probably find other complaints and
possibly who it is. Not that it will matter, there's nothing you can
do but block it.
My approach to call filtering is:
Deny All
Allow Some
I have a whitelist of callers I always want to accept that may include
businesses outside
I have a subroutine to block spammer by CALLERID(number)
exten => 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1)
exten => 4,n,Set(goaway=${CALLERID(number):0:2})
exten => 4,n,GotoIf($["${goaway}" = "V4" ]?blacklisted,s,1)
exten => 4,n,GotoIf($["${goaway}" = "V3" ]?blacklisted,s,1)
but I just got ano
On Thu, Jun 13, 2013 at 12:04 PM, wrote:
> Hi there
>
> I have asterisk 10.11.1 which seems to have problem negotiating codec.
>
> Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p
> and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw,
> h263p. I have tri
Hi there
I have asterisk 10.11.1 which seems to have problem negotiating codec.
Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p
and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw,
h263p. I have tried similar combination of codecs and SIP phone but when
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have been struggling with an audio issue for a week now and have
not been able to solve it.
We have an Asterisk server (now running 11.4 but started with 1.8)
with several sip phones on an internal network and a SIP trunk for
externa
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 a SPA3102 adapter to interface it to
asterisk. The number of the PSTN line i
If you do have chan_dahdi.so already, make sure you have a valid
chan_dahdi.conf file and try "module load chan_dahdi" in the CLI.
--
Eric Cooper e c c @ c m u . e d u
--
_
-- Bandwidth and Colocation Provided by htt
Hello Eric,
Thank your for your reponse. We are discussing interconnects at a
different level. We are more interested in SS7 or ISUP-IP SS7IP type
interconnects. There are many people that offer DIDs channels etc.
over the internet. Including us.
N.
--
___
In article ,
James Bensley wrote:
> Hi All,
>
> I am looking for a way to troubleshoot issues with TDM (E1) trunks
> with a provider.
>
> Currently with SIP trunks I am using tcpdump to perform packet
> captures between our gateways and the SIP providers IPs, capturing
> traffic on all ports, to
They offer standard SIP DIDs.I don't have a sales contact (others deal with
that), but if Google should have some links.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, Jun
Hi, I've already post this to the forum three days ago, sorry if it's
sounds like a crosspost, but I've got no replies, so I'm trying other
channels :)
This is the link to the forum post if someone prefer to reply here:
http://forums.asterisk.org/viewtopic.php?f=1&t=86985
I'm using Asterisk 1.8.2
On 6/13/13, Eric Wieling wrote:
> Verizon (NE ILEC) has SIP handoff.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
> Sent: Thursday, June 13, 2013 8:11 AM
> To: Asterisk Users Mailing List
Verizon (NE ILEC) has SIP handoff.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, June 13, 2013 8:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [ast
Mickael MONSIEUR wrote:
>
> My version is Asterisk 1.6.2.9.
>
> Or have you seen NAT ? I have no NAT on my network . Have you seen my little
> diagram above ?
>
> Here it is:
>
> SIP friends (phones) <-> Asterisk <-> SIP gateway to PSTN converter
> 80.236.215.61109.69.217.6
Hi!
Depending which TDM board you are using there might already be tool to
get a pcap trace. E.g. if you have a Sangoma board, the wanpipemon
utility has a -pcap option. I don't know about other boards. Wireshark
already comes with basic support for ISDN protocols, so now work is
needed here.
On Thu, Jun 13, 2013 at 9:23 AM, James Bensley wrote:
> Hi All,
>
> I am looking for a way to troubleshoot issues with TDM (E1) trunks
> with a provider.
>
> Currently with SIP trunks I am using tcpdump to perform packet
> captures between our gateways and the SIP providers IPs, capturing
> traff
Hi All,
I am looking for a way to troubleshoot issues with TDM (E1) trunks
with a provider.
Currently with SIP trunks I am using tcpdump to perform packet
captures between our gateways and the SIP providers IPs, capturing
traffic on all ports, to include both the SIP messages and the RTP
stream.
Correction:
"I think VT1.5s mappings are more flexible?"
Sorry!
N.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On 6/12/13, Don Kelly wrote:
> Is there an OC-n to SIP solution that makes sense?
>
> --Don
Hello Don, what will be coming out of the network discussed above would be SIP.
Kind Regards,
Nick.
--
_
-- Bandwidth and Colocation P
Hello Brian,
Thank you so much
On 6/12/13, Brian LaVallee wrote:
> Hi Nick,
>
> Going from DS1 to OC-n is a multi-step process. Typically requiring a
> hardware device to handle each MUX step. But you can find hardware that
> handles multiple MUX steps together.
The connection is coming i
Do you see the channel driver chan_dahdi.so in /usr/lib/asterisk/modules?
If not, go back to the * src dir, issue a ./configure, then make && make
install and check what * got this time.
If you have played with menuselect you might have to check these
settings, too.
jg
--
_
Hello,
I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't
install DAHDI channel, if I issue command
connect*CLI> core show channeltypes
I would have response like:
connect*CLI> core show channeltypes
TypeDescription Devicestat
On Thursday 13 June 2013, Mickael MONSIEUR wrote:
> Hello Matthew,
>
> My version is Asterisk 1.6.2.9.
>
> Or have you seen NAT? I have no NAT on my network. Have you seen my little
> diagram above?
>
> Here it is:
>
> SIP friends (phones) <-> Asterisk <-> SIP gateway to PSTN converter
Hello Matthew,
My version is Asterisk 1.6.2.9.
Or have you seen NAT? I have no NAT on my network. Have you seen my little
diagram above?
Here it is:
SIP friends (phones) <-> Asterisk <-> SIP gateway to PSTN converter
80.236.215.61109.69.217.6 internal IP (
10.4.0.10/
36 matches
Mail list logo
