Hello,
Considering we're in the apex of daylight savings time confusions
worldwide, I was wondering if there's a way to make IFTIME()
timespecs take timezone information. We have offices around the
globe that are being handled by a common Asterisk instance, and it
seems otherwise impossible to
Hello,
I operate an Asterisk server (v11.13.1) on Debian stable, and it's
rock-solid. The other day, however, I accidentally upgraded the
kernel from the stable 3.16.0 to 4.9.0. Subsequently, audio stopped
working.
Below you can find my analysis while running the 4.9.0 kernel. 888
is a simply
Hi,
I have a line like
register => 1yyy1:x...@sipconnect.sipgate.de/incoming
in sip.conf, and a corresponding stanza (note especially the final
setvar):
[trunk-sipgate]
type=peer
qualify=yes
insecure=invite
language=de
dtmfmode=rfc2833
host=sipconnect.sipgate.de
also sprach martin f krafft <madd...@madduck.net> [2015-09-02 14:16 +0200]:
> However, when a call comes in through the sipgate trunk and gets
> routed to the in-trunk-sipgate context, the ${FOO} variable is not
> set and thus not available from the dialplan.
Thanks to [TK]-Fend
also sprach Steve Edwards asterisk@sedwards.com [2015-05-17 08:31 +0200]:
While preprocessing could be called 'templating,' this may be
confusing because Asterisk already as a configuration file feature
called 'templates.'
Fair point. Preprocessing it shall be.
And you find
Hello,
I am in the peculiar situation to have to set up a PBX for two
independent sites, but operated by the same entity. Yes, I could set
up two VPSs and install Asterisk to each, put common stuff (e.g.
conferencing setup) into Git and share between both using includes,
but for various reasons
also sprach Steve Edwards asterisk@sedwards.com [2015-05-16 23:22 +0200]:
I use a preprocessor
(http://software.hixie.ch/utilities/unix/preprocessor/) to tailor
dialplans and configuration files to each host based on the client
(or project) and the hostname.
Yeah sure, templating works,
Hey,
we're experiencing a weird problem with Asterisk 1.8.13.1
(1:1.8.13.1~dfsg1-3+deb7). Calls that leave and enter Asterisk via
a PBX (sipgate.de) work perfectly fine, almost 100% of the time.
However, calls that are routed to sipgate.de, which then routes the
call back to our Asterisk
By chance, I managed to fig into this a bit and found the exact
moment when audio stops. It is exactly the moment when the
counterparty picks up and RTP debug output says:
Got RTP packet from46.244.255.146:8058 (type 00, seq 000680, ts
340914880, len 000160)
Sent RTP packet to
also sprach Brandon B. bran...@brellsystems.com [2012.12.03.0132 +0100]:
[all-inbound-for-999]
; inbound extension through a conference room
exten = 999,1,MeetMeCount(999,COUNT-999);
exten = 999,2,GotoIf($[${COUNT-999}=1]?10);
exten = 999,3,Dial(SIP/99,999,G(6));
exten = 999,4,Hangup;
exten
also sprach Raj Mathur (राज माथुर) r...@linux-delhi.org [2012.11.16.1005
+0100]:
Warning: Not a fan of using whitespace as semantic markup, so no Django
this side. Fine with Perl or Java, though.
As long as we can agree on using a database (i.e. no MySQL) or the
filesystem (Git…), then the
also sprach Shaun Ruffell sruff...@digium.com [2012.11.08.1615 +0100]:
My systems are already managed automatically, thankfully no longer
with Puppet. ;)
Just out of curiosity why do you say this?
Sorry for the late reply, I don't want to go into this on the list,
but if you are curious:
also sprach Paul Belanger paul.belan...@polybeacon.com [2012.11.08.2304
+0100]:
Either way, it sounds like you need to store your data some place and
start building it out.
To recap: given that Asterisk RealTime doesn't really provide
anything more than real-time access to data (i.e. the data
also sprach Administrator TOOTAI ad...@tootai.net [2012.11.08.0954 +0100]:
Does anyone have a working example they would be willing to
share?
As said by James, you just have to transfer all parties in
a conference room and then you call this conference.
The scenario is usually that we are
also sprach Jeff LaCoursiere j...@sunfone.com [2012.11.07.2049 +0100]:
Just to chime in, if you REALLY want multi-tenant, it is super
easy and surprisingly efficient to use kernel level virtualization
to run multiple instances of asterisk (and even FreePBX). We use
LXC to do this. The host
also sprach Administrator TOOTAI ad...@tootai.net [2012.11.08.1018 +0100]:
For a 3 way conference, all those days phones are able to do this.
Yeah, except I want Asterisk to handle that, not my phone (which
might lose reception or run out of battery etc.).
--
martin | http://madduck.net/ |
Hello,
we are finally going to redesign our Asterisk-Setup, which has grown
quite complex. We have five sites with a total of 400 users, 15 SIP
registrations and 3 IAX registrations. We do not use any
VoIP-hardware, so it's all software-based. But we make heavy use of
features, including
Can Asterisk do virtual hosting? While I want/need the sites to be
hosted by the same instance (so that e.g. calls can be transferred
easily), I don't want to have to name my peers [site1-john], and
I want people to be able to SIP-dial j...@site1.example.org and
j...@site2.example.org and trust
also sprach Joshua Colp jc...@digium.com [2012.11.07.1831 +0100]:
Peer names have to be distinct, this is just a fundamental design
element of chan_sip. What a lot of people end up doing is instead of
treating peers as people they treat them as devices. The peer name
becomes the MAC address of
Dear list,
we would really like to be able to invite a third and fourth party
to our current one-on-one call. At the moment, we have to agree to
dial into MeetMe 10 minutes later, then make calls to the third
parties, and hope it all works out.
I have found a couple of examples on the Internet
also sprach Paul Belanger paul.belan...@polybeacon.com [2012.11.07.2340
+0100]:
What is your point of pain? Right now we do most of the
configuration, provisioning, and system management outside of
asterisk.
My systems are already managed automatically, thankfully no longer
with Puppet. ;)
I
also sprach Logan Bibby lo...@keobi.com [2012.11.08.0747 +0100]:
What about just setting up a database which stores your data
however you want then generate static files from that data or
creating views for realtime (where appropriate)?
Sure, I could do that. First, however, I would like to
also sprach Luki lugos...@gmail.com [2009.09.19.0745 +0200]:
sounds like the hiccup my E71 had once. I think the symptoms were
identical. Changing the transport type from Auto to UDP solved the
problem for me. The Auto setting worked, but only sometimes. Maybe
the E65 is similar...
I've tried
also sprach Torintino T torinti...@hotmail.com [2009.09.19.1356 +0200]:
Try to put qualify=yes.
I had qualify=2000, but even with the default, the problem prevails.
Thanks for taking the time to reply,
--
martin | http://madduck.net/ | http://two.sentenc.es/
den stil verbessern, das heißt
Hey folks,
I am trying to get an E65 to connect to asterisk, and I would really
appreciate a second set of eyes. The SIP dialog completes fine, but
the phone subsequently says Registration failed.
I am in a network that has what seems to be a SIP-capable NAT
gateway, but the asterisk is
also sprach Jeff Peeler jpee...@digium.com [2009.06.16.1757 +0200]:
Have you set the parkedcallreparking, parkedcalltransfers, and other
associated options?
Only parkext and parkpos and context. All others are left at their
defaults. But of course this seems to be what I am looking for.
also sprach Danny Nicholas da...@debsinc.com [2009.06.16.1656 +0200]:
The problem is the Asterisk Read function. It is set to accept as
many 0-9 and * as you want to throw at it, then stop on # or
timeout. Unless you disable the # stops, you can't use # in
features. I would strongly caution
Hey folks,
I can park a call with #70 after enabling that feature in
features.conf. However, once I retrieve the call from the parking
lot, #70 cannot be used to park it again. Worse yet, none of the
keys defined in the featuremap work anymore, include blindxfer or
automon.
Any ideas what may be
Hi folks,
I was using the following featuremap:
blindxfer = *1
disconnect = *9
atxfer = *2
parkcall = *7
automixmon = *0
and everything worked.
Then the need arouse to use some features like automixmon during
a conference, but MeetMet has the * key bound to the
(admin) menu. Thus, in
also sprach Doug Lytle supp...@drdos.info [2009.06.16.1142 +0200]:
I can park a call with #70 after enabling that feature in
features.conf. However, once I retrieve the call from the parking
lot, #70 cannot be used to park it again. Worse yet, none of the
You fail to mention the version of
also sprach Doug Lytle supp...@drdos.info [2009.06.16.1314 +0200]:
Please remember, the patch is for 1.4
Right, and I found the corresponding lines in 1.6. But there are
more questions now:
- snprintf(returnexten, sizeof(returnexten), %s||t, peername);
+ snprintf(returnexten,
Hi folks,
When I try to park a call, my SIP phone puts the other party on hold
and MOH starts to play a tune. I then dial 700 and wait for the
parking slot announcement.
As soon as the other party gets put into the parkinglot, the MOH
tune starts again from the beginning. Is there a way to
Hi there,
I am experiencing a strange problem and am looking for advice to
where to start looking. Or any clues, really.
I have Asterisk running on our router, and it is configured to
forward calls to a provider out there (who is also using Asterisk).
On the inside of the Asterisk are several
rejection for user martin f. krafft
sip:[EMAIL PROTECTED];tag=fipzt
and SIP debugging then prints:
OPTIONS sip:sip05.insphone.ch SIP/2.0
Via: SIP/2.0/UDP 84.75.148.xxx:5060;branch=z9hG4bK71785803;rport
From: asterisk sip:[EMAIL PROTECTED];tag=as05fc20f4
I am not calling as username asterisk
also sprach Brent Davidson [EMAIL PROTECTED] [2008.03.28.2149 +0100]:
With canreinvite=no you are forcing asterisk to remain in the call path.
As long as Asterisk is in the call path, it is supposed to be transcoding
the calls, so it doesn't care what the compatible codecs are between then
Hi list,
I am faced by a situation where I am trying to make a softphone and
a Siemens C450IP talk to each other. Both are hooked up directly to
the same asterisk, in the same IP net.
- a softphone runs on 192.168.14.3
- the C450IP is at 192.168.14.30
- asterisk runs on the machine known
Hi,
I am on amd64 Linux and not really too happy with twinkle, linphone
and ekiga. Unfortunately, X-Lite and Zoiper, even though they
provide Linux versions (w00t!) have only x86 versions for download.
Do you guys know of amd64 versions of those, or can you recommend
other softphones that will
also sprach Tim Nelson [EMAIL PROTECTED] [2008.03.28.1637 +0100]:
I may be missing something here... but won't a 32bit binary run
just fine on a 64bit platform? Would you even see a performance
increase or advantage to a 64bit soft phone versus a 32bit
version?
Not if all the libraries have
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