Displaying 20 results from an estimated 2000 matches similar to: "Crash related to "asterisk -rx" ?"
2007 Oct 03
2
Where to download Junghanns ISDNguard software?
Hi list,
I recently purchased an ISDNguard from Junghanns. It came with no
software and there is no sign on their website or in any of their
documentation where to download it. I have looked in
http://www.junghanns.net/downloads/ and there is no sign of it there
either. The only thing remotly close ther is
isdnguard-asterisk-1.2.13.patch. Their documentation refers to
/usr/sbin/ISDNguard. Where
2018 Apr 04
2
Iridium integration / gateway
Thanks for reply, but this is irrelevant, I'm looking for an *Iridium*
gateway.
Regards,
--
Jean-Denis Girard
SysNux Syst?mes Linux en Polyn?sie fran?aise
https://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.797.527
Le 03/04/2018 ? 16:05, albert zhang a ?crit?:
> http://www.dinstar.cn/en/index.php/GSM/
>
> 2018-04-04 10:01 GMT+08:00 Jean-Denis
2016 Mar 07
2
Asterisk now available with bundled pjproject!
On Mon, Mar 7, 2016 at 2:53 PM, Jean-Denis Girard <jd.girard at sysnux.pf>
wrote:
> Hi,
>
> Le 07/03/2016 09:28, George Joseph a ?crit :
> > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released.
>
> I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got:
>
> [pjproject] Unpacking /tmp/pjproject-2.4.5.tar.bz2
> [pjproject]
2016 Feb 18
2
Grandstream Early Dial
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Hi list,
I've been using Grandstream phones for more than 10 years, but only
yesterday tried to use Early Dial... and I failed. What is needed on the
Asterisk side to reply 484 to INVITE? Phones are talking to chan_pjsip
on Asterisk-13.7.1.
Thanks,
- --
Jean-Denis Girard
SysNux Syst?mes Linux en Polyn?sie fran?aise
2015 May 21
0
PJSIP CCSS
2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <jd.girard at sysnux.pf>:
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> Le 21/05/2015 00:16, Joshua Colp a ?crit :
> > If CCSS is needed then the only option is to use chan_sip. The
> > chan_pjsip module does not implement CCSS in any way.
>
> Is CCSS support planned for PJSIP? chan_sip is in
2015 May 21
4
PJSIP CCSS
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Le 21/05/2015 00:16, Joshua Colp a ?crit :
> If CCSS is needed then the only option is to use chan_sip. The
> chan_pjsip module does not implement CCSS in any way.
Is CCSS support planned for PJSIP? chan_sip is in "extended" state in
asterisk-13, so chan_pjsip should be preferred for new installations, ri
ght?
Thanks,
- --
2009 Feb 04
1
AOC-E pass through
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Hi,
I'd like to know what is the current situation with regard to AOC-E,
when Asterisk is inserted between the telco and an existing PBX, using
E1 / EuroISDN. Can Asterisk pass the AOC-E information received from the
telco to the PBX, so that billing system still works? The system would
be for a hotel, so breaking billing system is not possible.
2014 Jan 20
1
DUNDI or ENUM or ?
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Hi list,
I'm looking for the best / recommended solution for automatic discovery
of phone numbers for a multiple Asterisk system. This would be for an
administration, with many branches (~30), but a common infrastructure
(DNS, LDAP). Most branches would have Asterisk servers for various
reasons (location, administrative). All contacts would be in
2015 May 21
2
PJSIP CCSS
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Hi list,
It looks like Call Completion Supplementary Services is not available
for PJSIP channels, am I right? Is there another solution?
Thanks,
- --
Jean-Denis Girard
SysNux Syst?mes Linux en Polyn?sie fran?aise
http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27
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2016 Feb 19
2
Grandstream Early Dial
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Hi Bryant,
Thanks for your reply.
It didn't work immediately, I had to create a second context, or else it
was looping between the second and first line. This seems to work:
[earlydial] ; Test Early Dial
exten => _.,1,Set(l_Extension=${EXTEN})
exten => _.,n,Goto(earlydial2,${l_Extension},1)
[earlydial2]
exten => _.,n,Goto(noMatch,1)
2016 Feb 19
2
Grandstream Early Dial
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Le 18/02/2016 11:03, Richard Mudgett a ?crit :
> I've been using Grandstream phones for more than 10 years, but onl
y
> yesterday tried to use Early Dial... and I failed. What is needed
on the
> Asterisk side to reply 484 to INVITE? Phones are talking to chan_p
jsip
> on Asterisk-13.7.1.
>
>
> Look into the
2015 Jul 27
2
PJSIP T.38 issues
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Hi list,
2 weks ago I asked questions about PJSIP and T.38 but got no replies. I
upgraded Asterisk to git as of yesterday (309dd2a), and I'm still having
the same issues.
In the trace below, I'm sending a fax from Hylafax server through
iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw)
connected to the PSTN via ISDN; the
2018 Apr 04
4
Iridium integration / gateway
Hi list,
I have a request to integrate Iridium in a Asterisk system. A quick
search didn't return much: I expected to find products similar to GSM
gateways, but this does not seem to exist. so I'd be very interested
about possible solutions. Has it be done already, how?
Thanks,
--
Jean-Denis Girard
SysNux Syst?mes Linux en Polyn?sie fran?aise
2013 Nov 18
1
CEL for attented transfer
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Hi list,
I'm trying to use CEL to display channel information in real time. It
works fine for simple calls, blind transfers, or SIP attended transfers
(initiated from the SIP phone). My problem is for Asterisk attended
transfers (atxfer as configured in features.conf).
The scenario is:
. phone 107 calls phone 100,
. 100 dials the atxfer code,
2015 Jul 29
2
PJSIP T.38 issues
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Thanks for your reply Larry.
Le 27/07/2015 01:22, Larry Moore a ?crit :
> I think the "488 Not acceptable here" is occurring because the channel
> connecting through is not T.38 capable, that will be the IAX channel
> from iaxmomdem.
This is what T38gateway is supposed to do. And I'm very happy to report
that after one more
2015 May 20
2
CHANNEL(aor) CHANNEL(contact) return nothing
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Hi list,
I'm trying to use CHANNEL(aor) and CHANNEL(contact) on PJSIP channel, on
asterisk-13.3.2, but they don't return anything. Is this a bug, or did I
miss something?
Here is my test dialplan:
exten => *98,1,Answer
same => n,NoOp(Channel=<${CHANNEL(name)}>,type=
<${CHANNEL(channeltype)}>)
same =>
2005 Jul 05
2
Previously: Queue + optional URL
Does anybody know if there is an app that will cause similar to occur on users
PC?
I have a scenario where users will have snom phones on their desks. Ideally when
their phone receives a call I need to popup a web browser with a specific url.
Any ideas appreciated.
Neil
on 5/7/05 10:52 PM, Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com> wrote:
2015 May 20
1
CHANNEL(aor) CHANNEL(contact) return nothing
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Le 20/05/2015 00:50, Joshua Colp a ?crit :
> It looks like this is an incoming leg, in which case that information
> isn't available. There is no association of an AOR and Contact on
> incoming legs (it MAY be possible to deduce but it certainly wouldn't
> work in all cases). Since you specify one explicitly on outgoing, that
2008 Oct 02
1
Asterisk Queue question
When the asterisk a queue reset their counters?
I 'm talking about this kind of info in asterisk console.
>show queue 600
600 has 0 calls (max unlimited) in 'ringall' strategy (4s
holdtime), W:0, C:14, A:8, SL:0.0% within 0s
I just say that because I have a queue with strategy "Fewest Calls"
working for a couple of mouths, and a new agent has been added this
2013 Feb 11
2
[OT] Mediatrix Euro ISDN hangup problem
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Hi list,
I'm getting a strange problem with a Mediatrix 3631 Gateway connected to
the PSTN via an E1 PRI link configured for Euro ISDN signaling. The
Mediatrix sends incoming calls from the PSTN to an Asterisk server via
SIP: this works fine. But when the caller hangs up, the Mediatrix
doesn't send "Bye" to Asterisk, so the call is