Displaying 20 results from an estimated 10000 matches similar to: "originate woes: extension never executes"
2015 Nov 01
5
no ringing tone with Dial option r
I'm not getting any ringing when I use option r with Dial:
Dial("DAHDI/1-1", "motif/8447/+1<called-num>@voice.google.com,,rTt") in
new stack
Otherwise all works. The call goes through, good audio.
sean
2014 Dec 21
2
11.5.0: blindxfer problems [Spam score:10%]
Have you enabled DTMF logging and seen the DTMF codes being recognised by
Asterisk? I had a bunch of soft phones that I had to change to using ?sip
info? for the DTMF signalling as the RFC signalling was not always being
recognised. This would cause transfers to appear as if the user had not
dialled any digits.
On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:
2014 Dec 22
2
11.5.0: blindxfer problems
On 12/21/2014 11:09 AM, sean darcy wrote:
> On 12/21/2014 04:42 AM, Patrick Beaumont wrote:
>> Have you enabled DTMF logging and seen the DTMF codes being recognised by
>> Asterisk? I had a bunch of soft phones that I had to change to using ?sip
>> info? for the DTMF signalling as the RFC signalling was not always being
>> recognised. This would cause transfers to appear
2014 Dec 20
2
11.5.0: blindxfer problems
On 12/19/2014 09:42 AM, Rusty Newton wrote:
> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote:
>> I've got a confbridge set up which works if dialed locally:
>>
>> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack
>> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1",
2014 Dec 21
0
11.5.0: blindxfer problems
On 12/21/2014 04:42 AM, Patrick Beaumont wrote:
> Have you enabled DTMF logging and seen the DTMF codes being recognised by
> Asterisk? I had a bunch of soft phones that I had to change to using ?sip
> info? for the DTMF signalling as the RFC signalling was not always being
> recognised. This would cause transfers to appear as if the user had not
> dialled any digits.
>
>
>
2014 Dec 17
2
11.5.0: blindxfer problems
I've got a confbridge set up which works if dialed locally:
-- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack
-- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack
-- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack
-- <DAHDI/1-1> Playing
2014 Dec 20
0
11.5.0: blindxfer problems
On 12/20/2014 03:22 PM, sean darcy wrote:
> On 12/19/2014 09:42 AM, Rusty Newton wrote:
>> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote:
>>> I've got a confbridge set up which works if dialed locally:
>>>
>>> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack
>>> --
2013 Mar 07
2
11.3: how to hang up on google voice
Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've hung up.
extensions.conf:
same => n,GoToIf($["${CALLERID(num)}"="office"]?email)
.................
same => n(email),System(/usr/local/bin/emailme........)
same => n,Answer() ; also tried without this
same =>
2014 Dec 19
0
11.5.0: blindxfer problems
On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote:
> I've got a confbridge set up which works if dialed locally:
>
> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack
> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack
> -- Executing [266 at internal:3]
2020 Jan 22
2
permission woes on systemd
I'm running asterisk 16 on Fedora 31. If I start asterisk as user
asterisk, all goes well. But if I start asterisk from systemd:
asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: res_sorcery_config.c:320
sorcery_config_internal_load: Unable to load config file 'pjsip.conf'
Jan 21 19:36:47 asterisk.riverside asterisk[1411]: [Jan 21 19:36:47]
ERROR[1411]: config_options.c:710
2020 Jan 22
0
permission woes on systemd
----- Original Message -----
> From: "sean darcy" <seandarcy2 at gmail.com>
> To: "Asterisk Users Mailing List, Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> Sent: Tuesday, January 21, 2020 9:22:28 PM
> Subject: [asterisk-users] permission woes on systemd
[..]
> So why would starting asterisk as user asterisk work, but fail using
2019 Apr 08
2
pjsip endoint woes
On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote:
> On 4/5/19 10:36 AM, sean darcy wrote:
> > I'm trying to set up pjsip to work with an obi202 and google voice. But
> > I can't configure the endpoint.
> >
> > pjsip:
> >
> > [obi202-auth](!)
> > type = auth
> > auth_type = userpass
> > password = <mypass>
> >
>
2013 May 16
2
11.4: motif can only handle one channel at a time?
I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.
motif only handles one 1 channel at a time??
sean
2004 Aug 17
6
dialplan woes
I am making some changes to the dial plan at the request of the company
president and have run into some problems. I have a couple of layers of
menu's and I am not sure how to handle them.
Here is how it should work (sorry for the crappy diagram)
main menu
--------Dial 1 for support
| Dial 2 for special
| Dial 3 sales
2013 Jun 10
3
no silk translation ?
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:
[Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c
[Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032
dial_exec_full: Had to drop call because I couldn't make
2019 Apr 05
2
pjsip endoint woes
I'm trying to set up pjsip to work with an obi202 and google voice. But
I can't configure the endpoint.
pjsip:
[obi202-auth](!)
type = auth
auth_type = userpass
password = <mypass>
[obi202-aor](!)
type = aor
max_contacts = 2
; ===== endpoints ========
[gv-voice](obi202-endpoint)
auth = gv-voice
aors = gv-voice
identify_by=auth_username
;identify_by=username ; I also tried
2005 Nov 28
3
Not an IMAP4 Server?
I've been testing alpha4, and every once in a while my client
(Thunderbird 1.5 RC1) tells me the dovecot isn't a valid IMAP4 server.
This is usually (maybe always) when I first bring up the client up. If
I just ask the client to get the mail again, it does so happily (without
any errors).
I must say I found dovecot as a pleasant surprise. We have been using
UW. Which didn't
2015 Mar 02
0
Upgrade to Fedora 21, now gv requires rtp ?
I just upgraded to fedora 21. I'm running asterisk 11.6.0. All works
with Fedora 20.
-- Executing [s at DialOut:15] Dial("DAHDI/1-1",
"motif/8447/+1212xxxyyyy at voice.google.com,,rTt") in new stack
[Mar 1 21:24:06] ERROR[2477][C-00000000]: rtp_engine.c:259
ast_rtp_instance_new: No RTP engine was found. Do you have one loaded?
[Mar 1 21:24:06]
2009 Sep 23
1
1.6.0.5: I need a really simple analog SendFax dialplan
Using Digium fax I've tried a simple dialplan:
'8447' => 1. Answer() [pbx_config]
2. Set(CALLERID(num)=xxxyyyzzzz) [pbx_config]
3. Dial(DAHDI/g0/1bbbcccdddd,,G(send)) [pbx_config]
[send] 4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config]
5. HangUp()
But I doesn't work. It executes
2009 Sep 18
1
digium fax: is this even close to working?
My set up is 1.6.0.15 with the digium fax modules. I want to capture a
fax from the internal analog fax machine (using an SPA2102), and then
resend it. I know the internal extension of the fax machine, and for now
I'm just testing it to one outside fax machine if I dial 8447.
In particular, I'm completely unfamiliar with the use of "G" in the Dial
app.
exten =>