23.04.2019 0:27, Joshua C. Colp wrote:
On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote:
Tried already.
"line" is good, but not perfect.
Every time I restart asterisk, it will generate new random string for ";line=".
So, every time I restart asterisk, registrar (Server1) will save one
more c
On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote:
> Tried already.
>
> "line" is good, but not perfect.
>
> Every time I restart asterisk, it will generate new random string for
> ";line=".
>
> So, every time I restart asterisk, registrar (Server1) will save one
> more contact in it's database
Hi,
Thank for your answer.
22.04.2019 23:47, Joshua C. Colp пишет:
On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:
Hi,
Got problems with incoming SIP calls.
Scenario:
Server1: 3cx or any other server
Server2: Asterisk 16.2.1 . PJPROJECT 2.8
Server2 registers on Server1 with SIP ID 1121.
R
On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:
> Hi,
>
> Got problems with incoming SIP calls.
>
> Scenario:
>
> Server1: 3cx or any other server
>
> Server2: Asterisk 16.2.1 . PJPROJECT 2.8
>
> Server2 registers on Server1 with SIP ID 1121.
>
> Registration is OK.
>
> Server2 outgoing calls
Hi,
Got problems with incoming SIP calls.
Scenario:
Server1: 3cx or any other server
Server2: Asterisk 16.2.1 . PJPROJECT 2.8
Server2 registers on Server1 with SIP ID 1121.
Registration is OK.
Server2 outgoing calls are OK.
INVITE, unauthorized, INVITE with password, OK, RINGING,...
Troub
Hello Thelma,
Friday, February 16, 2018, 2:16:02 AM, you wrote:
> Contact: "sip:pstn-"
> And it found in sip.conf only:
> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
> Is perhaps the name effected by the special character "-" (dash) that is
> why it only matches "pstn" and t
Le 16/02/2018 à 05:30, the...@sys-concept.com a écrit :
On 02/15/2018 04:49 PM, Joshua Colp wrote:
On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
Thanks again for the hint.
Here is the output from asterisk.
The call is coming on Audocodes gateway from: pstn-
But asteri
On 02/15/2018 04:49 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>
>
>
>>
>> Thanks again for the hint.
>> Here is the output from asterisk.
>>
>> The call is coming on Audocodes gateway from: pstn-
>>
>> But asterisk display:
>> Found peer 'pstn-9
Thelma
On 02/15/2018 07:16 PM, the...@sys-concept.com wrote:
>
> On 02/15/2018 04:49 PM, Joshua Colp wrote:
>> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>>
>>
>>
>>>
>>> Thanks again for the hint.
>>> Here is the output from asterisk.
>>>
>>> The call is coming on Audocod
On 02/15/2018 04:49 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>
>
>
>>
>> Thanks again for the hint.
>> Here is the output from asterisk.
>>
>> The call is coming on Audocodes gateway from: pstn-
>>
>> But asterisk display:
>> Found peer 'pstn-
On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>
> Thanks again for the hint.
> Here is the output from asterisk.
>
> The call is coming on Audocodes gateway from: pstn-
>
> But asterisk display:
> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
>
> Why not l
On 02/15/2018 04:08 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
>> On 02/15/2018 03:44 PM, Joshua Colp wrote:
>>> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
I'm using Audio-codes MP-114 unit and it has two public lines PSTN port
On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
> On 02/15/2018 03:44 PM, Joshua Colp wrote:
> > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
> >>
> >> IN audocodes setting I have:
> >>
On 02/15/2018 03:44 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>>
>> IN audocodes setting I have:
>> "EndPoint Phone Number"
>>
>> Channel: 3phone number: pstn-
>> Chann
On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>
> IN audocodes setting I have:
> "EndPoint Phone Number"
>
> Channel: 3phone number: pstn-
> Channel: 4phone number: pstn-9998
>
> When I am ca
I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
IN audocodes setting I have:
"EndPoint Phone Number"
Channel: 3phone number: pstn-
Channel: 4phone number: pstn-9998
When I am calling " pstn-" the port number "Channel:3" lights up but
asterisk is showing t
On Wednesday 26 Oct 2016, KyD wrote:
> Hi,
>
> My sip provider gave me 2 numbers for the incoming call via pstn.
>
> nro1 = 12341234
> nro2 = 45674567
>
> I have a dialplan for each.
> if i put this on my dialplan:
>
> exten => s,1,Dial(SIP/1001)
> exten => Hangup()
>
> Works!
>
> But if i pu
It seems like your SIP provider is not sending and DID information, or that
the information is not being sent in the same format you are using in your
dialplan.
You can check this by looking at the SIP debug information for the inbound
calls and/or by checking with your SIP provider (that they are
Hi,
My sip provider gave me 2 numbers for the incoming call via pstn.
nro1 = 12341234
nro2 = 45674567
I have a dialplan for each.
if i put this on my dialplan:
exten => s,1,Dial(SIP/1001)
exten => Hangup()
Works!
But if i put one of them:
exten => 12341234,1,Dial(SIP/1001)
exten => _1234
Hello Phil,
On Saturday, April 23, 2016, 11:11:29 PM, you wrote:
> Actually, this is now sorted. It turns out the latest recommended
> configs on the A&A wiki had peer vs. user confusion. On correcting
> this, all was well.
I'm glad you found it. It look me a while to track down that problem
whe
On Sat, 23 Apr 2016 22:45:32 +0100
Julian Beach wrote:
> Hello Phil,
>
> I have a couple of lines with A&A, and I have not been having any
> problems recently. When I have had similar problems in the past, it
> has been an issue with the SIP config. I originally had a number of
> contexts set up
Hello Phil,
On Saturday, April 23, 2016, 12:19:15 PM, you wrote:
> I have checked that the username and password in my config agree both
> ends, and have even tried changing them.
> The bulk of my calls come in on A&A, so I am obviously trying to find
> out what has gone wrong. No-one else is se
I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk).
VoIPtalk calls are unauthenticated and reach me fine, but Andrews &
Arnold calls are authenticated. The last call I successfully received
was on Tuesday afternoon. Initially, A&A were for some odd reason not
sending calls to my
On Sun, 20 Mar 2016, Trey Hilyard wrote:
On Mar 18, 2016 8:27 PM, "Steve Edwards" wrote:
>>
>> On Fri, 18 Mar 2016, Trey Hilyard wrote:
>>
>>> I thought this would be as easy as
>>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})
>
>
> How about something like:
>
> [parse-
On Mar 18, 2016 8:27 PM, "Steve Edwards" wrote:
>>
>> On Fri, 18 Mar 2016, Trey Hilyard wrote:
>>
>>> I thought this would be as easy as
>>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})
>
>
> How about something like:
>
> [parse-lrn]
> exten = _x.,1,
On Fri, 18 Mar 2016, Trey Hilyard wrote:
I thought this would be as easy as
exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})
Have you tried the '_!.' pattern?
--
Thanks in advance,
-
Steve Edwards
I am trying to set up my Asterisk server so that it will recognize an
incoming call to the Asterisk's own Location Routing Number (LRN),
validating the "rn" in the INVITE and then using the Called Number from the
INVITE as the extension in the dialplan.
The INVITE R-URI looks like:
INVITE
sip:+19
On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI
wrote:
> Le 18/03/2016 16:20, Trey Hilyard a écrit :
> > I am trying to set up my Asterisk server so that it will recognize an
> > incoming call to the Asterisk's own Location Routing Number (LRN),
> > validating the "rn" in the INVITE and the
Le 18/03/2016 16:20, Trey Hilyard a écrit :
I am trying to set up my Asterisk server so that it will recognize an
incoming call to the Asterisk's own Location Routing Number (LRN),
validating the "rn" in the INVITE and then using the Called Number from
the INVITE as the extension in the dialplan.
I thought this would be as easy as
exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})
But it appears that the pattern match doesn't work once I get to the "r" in
"rn". I am assuming that the pattern match doesn't like dealing with
characters without taking the entire URI.
I am
On Fri, 18 Mar 2016, Steve Edwards wrote:
Have you tried the '_!.' pattern?
The '_x.' pattern works fine.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https
On Fri, 18 Mar 2016, Trey Hilyard wrote:
I thought this would be as easy as
exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})
How about something like:
[parse-lrn]
exten = _x.,1, verbose(1,[${EXTEN}@${CONTEXT}])
same = n,
Larry Moore omninet.net.au> writes:
>
> sip.conf
>
> [general]
> faxdetect=t38
>
> [sipcall.ch]
> directmedia=no
>
> In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a
> T.38 re-invite this will trigger the switch to the Fax extension.
>
> If this proves successful you ca
El 20/01/16 a las 18:33, Alex Villacís Lasso escribió:
El 20/01/16 a las 16:25, Alex Villacís Lasso escribió:
Partial fix: Google Chrome accepts the call if videosupport is set to "no".
This is the SDP of the successful INVITE that Chrome accepts:
INVITE sip:8cj802p8@192.0.2.240;transport=ws
El 20/01/16 a las 16:25, Alex Villacís Lasso escribió:
I am having trouble getting Google Chrome to accept a WebRTC call coming from
Asterisk, even though Firefox can (now) accept the same call without issue.
My setup is as follows:
Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192
I am having trouble getting Google Chrome to accept a WebRTC call coming from
Asterisk, even though Firefox can (now) accept the same call without issue.
My setup is as follows:
Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around
https
On 8/21/15 6:45 PM, Technical Support wrote:
I got a new SNOM M65 which works fine for outgoing calls, but incoming
calls never ring at the handset. I captured the SIP traffic and see
that my M65 is replying with an "488 not acceptable here". From what I
read this is usually codec related but
-
From: Technical Support [supp...@telium.ca]
Received: sexta-feira, 21 ago 2015, 19:46
To: asterisk-users@lists.digium.com [asterisk-users@lists.digium.com]
Subject: [asterisk-users] Incoming calls get 488 error
I got a new SNOM M65 which works fine for outgoing calls, but incoming
calls never ring
I got a new SNOM M65 which works fine for outgoing calls, but incoming
calls never ring at the handset. I captured the SIP traffic and see
that my M65 is replying with an "488 not acceptable here". From what I
read this is usually codec related but both asterisk and the M65 are set
for ulaw a
Hello:
I'm newbie in asterisk, please help me.
My context is as follows:
192.168.4.2 --> Asterisk 11.13.1 complied from source
192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway
When I call from a GSM cell phone, my TG100 GSM gateway answers and
dials extension (configured as a hotline on
On Saturday 19 Jul 2014, Norman Molhant wrote:
> I tried many things on our FreePBX box and found out
> the problem seems somehow linked with the customer's
> extension (or phone number), not his inbound route
> (changing the latter has no effect on the problem).
>
> Creating a new extension with
:43 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] incoming calls fall into echo test mode
>
> Hello all,
>
> Weird trouble here:
> we have 60-some happy subscribers on a FreePBX box, each with its own phone
> number, with no problem at all, exce
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant
Sent: Saturday, July 19, 2014 10:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] incoming calls fall into echo test mode
Hello all,
Weird trouble here:
we have 60-some happy subscribers on a FreePBX box, each with
is being misrouted in the dialplan
From: asterisk-users-boun...@lists.digium.com
on behalf of Norman Molhant
Sent: Saturday, July 19, 2014 10:43 AM
To: Asterisk Users List
Subject: [asterisk-users] incoming calls fall into echo test mode
Hello all
Hello all,
Weird trouble here:
we have 60-some happy subscribers on a FreePBX box,
each with its own phone number, with no problem at all,
except for one (and only one) subscriber who has this
problem: his outgoing calls are ok, but when someone
dials his phone number (be it from our network or fr
On 3/02/2014 8:42 PM, Jakob-Matthias Böttger wrote:
Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger:
Am 03.02.2014 12:56, schrieb Larry Moore:
On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:
as He is describing it he should actually provide t.38. but i don't
know
why it is not working
Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger:
Am 03.02.2014 12:56, schrieb Larry Moore:
On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:
as He is describing it he should actually provide t.38. but i don't
know
why it is not working thus im now getting
Feb 3 12:32:55] WARNING[9942][C
Am 03.02.2014 12:56, schrieb Larry Moore:
On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:
as He is describing it he should actually provide t.38. but i don't know
why it is not working thus im now getting
Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp:
Failed to ini
On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:
as He is describing it he should actually provide t.38. but i don't know
why it is not working thus im now getting
Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp:
Failed to initialize UDPTL, declining image stream
[Feb
as He is describing it he should actually provide t.38. but i don't know
why it is not working thus im now getting
Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353
process_sdp: Failed to initialize UDPTL, declining image stream
[Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c
On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote:
Hi, changing
faxdetect=cng
and
t38pt_udptl=no
helped making it work.
Hmm, the fax will be received as an audio call rather than T.38, setting
t38pt_udptl=no has turned off T.38.
Do you know if your upstream provider supports T.38?
Larry
Hi, changing
faxdetect=cng
and
t38pt_udptl=no
helped making it work.
Thanks
Am 03.02.2014 11:57, schrieb Larry Moore:
On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote:
.
.
.
[sipcall.ch]
type=peer
insecure=invite
defaultuser=123456789
fromuser=123456789
fromdomain=voipdomain.com
secret=g
On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote:
.
.
.
[sipcall.ch]
type=peer
insecure=invite
defaultuser=123456789
fromuser=123456789
fromdomain=voipdomain.com
secret=gueswhat
host=voipdomain.com
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=123456789
add
directmedia=no
Hi, im using a Asterisk Server which is not behind NAT.
First i had problems with the fax detection. But this is now solved
after adding a wait(2) at the correct place. But i'm still unable to
receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too
short after the Fax session ha
B.H.
Hi!
On Wed, Jun 5, 2013 at 7:26 PM, jg wrote:
For a BRI device a single span has 2 channels, a PRI device up to 30. As
> far as channel variables go the actual channel does not seem to get
> reported, but this is not really necessary.
AFAIK, at least for AMI listeners, the real channel/s
Yes, my assumption was wrong and to make things worse, my CDR data
clearly show that "i" cannot denote incoming calls.
Maybe it's time that I learn the rules as well:
Analog channels do not seem to have a special identifier. The 1st call
for analog channel 13 would be s.th. like DAHDI/13-1.
> Sangoma's tech support is probably the better source of information.
>
> DAHDI: obviously DAHDI channel
> i: incoming call
The 'i' is for ISDN not incoming call since it will be this way for outgoing
calls as well.
> 3: span 3 (not the port)
> 211123456: CLID, probably subject to filtering (s
Sangoma's tech support is probably the better source of information.
DAHDI: obviously DAHDI channel
i: incoming call
3: span 3 (not the port)
211123456: CLID, probably subject to filtering (see
national/international prefix settings)
89c: internal counter (i.e. 2204 calls so far)
jg
--
__
Hi,
I use a Sangoma A104d-Card (with 4 x germany E1). I process some calls
via an AGI-Script. When parsing the AGI-Variables I can see one that
look like that:
[agi_channel] => DAHDI/i3/211123456-89c
What hat do the values mean in detail, please?
DAHDI : this is clear
i3 : does it mean, tha
On Tue, Nov 6, 2012 at 1:50 PM, Roy Abshire wrote:
> I have fax working but since most people and services don't know how to
> Fax to Extensions,
> I installed tesseract to convert the Fax to Text.
>
> I really only need the First Page converted and will tell Faxers to make
> sure they put To: Na
: Asterisk Users
Subject: [asterisk-users] Incoming Fax to Recipient using OCR
I have fax working but since most people and services don't know how to Fax
to Extensions, I installed tesseract to convert the Fax to Text.
I really only need the First Page converted and will tell Faxers to make
I have fax working but since most people and services don't know how to
Fax to Extensions,
I installed tesseract to convert the Fax to Text.
I really only need the First Page converted and will tell Faxers to make
sure they put To: Name on the cover page.
tesseract is converting the entire fa
I have fax working but since most people and services don't know how to
Fax to Extensions,
I installed tesseract to convert the Fax to Text.
I really only need the First Page converted and will tell Faxers to make
sure they put To: Name on the cover page.
tesseract is converting the entire fa
- Original Message -
> From: "Yaroslav Panych"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, April 17, 2012 6:56:17 PM
> Subject: Re: [asterisk-users] Incoming SIP call is rejected always.
>
> 2012/4/18 Mat
2012/4/18 Matthew Jordan :
> I imagine that this is the case, as ASTERISK-19601 noted that
> when this situation occurs, the NOTICE message indicates that
> there is a failure to match the extension, as opposed to a failure
> to match an allowed domain.
Yes, it was hell to detect real error cause
April 17, 2012 4:58:14 PM
> Subject: Re: [asterisk-users] Incoming SIP call is rejected always.
>
> 2012/4/17 Danny Nicholas :
> > Maybe it needs to be _4001020?
> >
>
> Not, it doesn't. Actually I have traced this incoming call step by
> step. Real r
2012/4/17 Danny Nicholas :
> Maybe it needs to be _4001020?
>
Not, it doesn't. Actually I have traced this incoming call step by
step. Real reason it refuses - wrong domain. But why it wrong - have
not any idea.
--
_
-- Bandwidth
: [asterisk-users] Incoming SIP call is rejected always.
Hi
Have an asterisk. Setup a couple of friends.
Sip.conf - http://pastebin.com/zUgiYbBi
Trying to make incoming call, and have such error(cli output)
http://pastebin.com/zFfgYcNR
NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from '
Hi
Have an asterisk. Setup a couple of friends.
Sip.conf - http://pastebin.com/zUgiYbBi
Trying to make incoming call, and have such error(cli output)
http://pastebin.com/zFfgYcNR
NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from
'RMT20' (192.168.8.1:5062) to extension '4001020' reje
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Hall
Sent: Friday, June 10, 2011 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Incoming Call Recording
Longtime lurker, first
Longtime lurker, first time poster. :)
A client of mine is in need of having Asterisk record every call that comes
in from a specific incoming route. I've added the following lines to the
sip_additional.conf file, but no recordings are showing up in the
/var/spool/asterisk/monitor/ folder.
reco
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Everybody,
I am experiencing some troubles with my Bria iPhone Edition (v. 1.2.8
build 5312, on iOS 4.2.1 iPhone 3G) and Asterisk 1.8.3.2 + TLS/SRTP on
LAN (without NAT).
With 2 computer clients (Blink, one on Mac, one on Windows/Linux),9i can
hav
On Mon, 3 Jan 2011, Thomas Perron wrote:
So, one Asterisk machine handling up to 100 DID numbers, correct?
The number of DIDs is not limited. You could handle a bazillion DIDs with
a simple dial plan like:
exten = _!.,1, verbose(1,[${ext...@${context}])
exte
On Mon, Jan 03, 2011 at 02:41:36AM +0400, Thomas Perron wrote:
>Cool. So, one Asterisk machine handling up to 100 DID numbers, correct?
As many as you like, modulo memory and CPU requirements.
>I assume that the DID mumbers dialed would be the exaxt match needed
>to start the respective context.
On Sun, 2 Jan 2011, Thomas Perron wrote:
Is it possible to have Calls incoming to different DIDs?
Yes*, depending on whether your provider 'provides' the DID in the call
setup.
*) Better subjects attract more readers. More detail yields better
answers.
--
Thanks in advance,
Cool. So, one Asterisk machine handling up to 100 DID numbers, correct?
Yes. I will have unique IVR flows/plans for each.
I assume that the DID mumbers dialed would be the exaxt match needed
to start the respective context. Correct?
On 1/3/11, Rick Hall wrote:
> Yes, I don't see why not. You j
Yes, I don't see why not. You just need to setup an IVR for each business
and then assign each individual DID to the appropriate IVR.
This may help:
http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu
Cheers!
Rick
--
Rick Hall
Senior Vice President
ReadyWire Multimedia Solutions
Affor
Is it possible to have
Calls incoming to different DIDs?
I want an AA that handles 100s of businesses.
[Incoming-pizza]
Exten => 4045551212,1,Goto(pizza,s,1)
[Incoming-hvac]
Exten => 8085551212,1,Goto(hvac,s,1)
[Incoming-gutter]
Exten => 6175551212,1,Goto(gutter,s,1)
--
Matt,
We are located on Costa Rica and so far there's just 1 TELCO running the
industrym with the CAFTA treatment the carrier had to open for
interconnection but they get to define the ground rules for the
interconnection.
They are arguing ISDN is and "end customer" circuit and you cannot use it
Just out of curiosity, what country are you in?
I agree with the others in this thread, this seems very bizzare that the
telco requires you to do SS7 for dialup connections. I would ask them for
specifics about the "legal" issues with what you are doing - it sounds to me
like they are just trying
ment” but no PRIs.
>
>
>
> A lot of this equipment was available by the pound a few years back.
>
>
>
> Cary
>
>
> --
>
> *From:* José Pablo Méndez Soto [mailto:aux...@gmail.com]
> *Sent:* Wednesday, November 24, 2010 8:34 PM
> *T
..@lists.digium.com] *On Behalf Of *José Pablo Méndez
> Soto
> *Sent:* Wednesday, November 24, 2010 7:31 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Incoming calls through SS7 for data
> modemtransmissions - possible??
>
>
>
> Hello,
>
> W
: asterisk-users@lists.digium.com
Subject: [asterisk-users] Incoming calls through SS7 for data
modemtransmissions - possible??
Hello,
We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem
Hello,
We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.
The Telco they were buying the trunks to discovered this configuration and
restricted th
On Thu, 18 Nov 2010, Flavio Miranda wrote:
Looking to dahdi show channles , I realized that all the trunks was in
the same context. So, I have changed this and everything works!
That's why I prefer to work from what Asterisk parsed the file as, not
what the poster thinks :)
--
Thanks in a
to Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
> Date: Thu, 18 Nov 2010 11:53:26 -0800
> From: asterisk@sedwards.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Incoming calls
>
> On Thu, 18 Nov 2010, Flavio Miranda wrote:
>
> >
On Thu, 18 Nov 2010, Flavio Miranda wrote:
> I'd like that each analog trunk of my TDM410p was received in different
> extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each
> trunk in a different context and in my extensions.conf, under [default]
> I put such contexts and an espec
Hi all,
I'd like that each analog trunk of my TDM410p was received in different
extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a
different context and in my extensions.conf, under [default] I put such
contexts and an especific estension to answer it. therefore,
Hi all,
After a lot of trouble with a TE110p working with mfcr2 , brazil variant,
everything looks great,but I can not go out of my calls.
When I try I receive the following log:
== Using SIP RTP CoS mark 5-- Executing [33220...@local:1]
Dial("SIP/4804-001a", "DAHDI/g11/33220567,,T"
Ok. Problem solved .
Thank you very much!!!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Wed, 15 Sep 2010 09:56:36 -0400
From: zisha...@gmail.com
To: kpflem...@digium.com; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] incoming
As Kevin said, you need to define an 's' extension where the calls will be
answered. Seems like you are using default configuration. Open file
'extensions.conf' in /etc/asterisk folder and look for context named
[default]. If it is not there, create one and add something under it, e.g.,
[default]
On 09/15/2010 07:20 AM, Flavio Miranda wrote:
> Recently I have instaled one Digium TDM410 on my Asterisk. After
> instaled , I can do outgoing calls but I cant receive calls. I receive
> the following messages:
>
> chan_dahdi.c: Got event 2 (Ring/Answered)...
> [Sep 14 11:24:44] NOTICE[2654
Hi all,
Recently I have instaled one Digium TDM410 on my Asterisk. After instaled ,
I can do outgoing calls but I cant receive calls. I receive the following
messages:
chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654]
chan_dahdi.c: Got event 18 (Ring Begin)...[Sep
On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood wrote:
> On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby
> wrote:
> > On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood wrote:
> >>
>
> My experience with Asterisk in the past has been with inbound analog
> lines so that would make sense :)
>
> See if you spot anyt
On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wrote:
> On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood wrote:
>>
>> I don't see any
>>
>> On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby
>> wrote:
>> >
>> > You don't have any extensions in your default context that match the
>> > extension that your sip pee
On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood wrote:
> I don't see any
>
> On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby
> wrote:
> >
> > You don't have any extensions in your default context that match the
> > extension that your sip peer is dialing in on. 's' is not a default
> > extension for SIP.
I don't see any
On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wrote:
>
> You don't have any extensions in your default context that match the
> extension that your sip peer is dialing in on. 's' is not a default
> extension for SIP...try using _X., and see what you get. Bump up the CLI
> (core s
On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood wrote:
> Hello.
>
> I have been beating my head over this problem for about 6 hours now.
>
> I have a SIP peer, who I register to (successfully), who should be
> directing all incoming calls at my [default] stanza in my
> extensions.conf:
>
> [ Context 'de
Hello.
I have been beating my head over this problem for about 6 hours now.
I have a SIP peer, who I register to (successfully), who should be
directing all incoming calls at my [default] stanza in my
extensions.conf:
[ Context 'default' created by 'pbx_config' ]
's' =>1. Wait(1)
phone in one room and
picking up in another room without disconnecting the line. Make a small test to
verify this and if this is the case you will need to discuss this with your
PSTN provider.
Harel
Date: Thu, 8 Jul 2010 12:01:40 -0500
From: Daniel - Asterisk
Subject: [asterisk-users] Incoming
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