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 enf
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 Ech
also sprach martin f krafft [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]-Fender, we isolated the pro
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
fr
also sprach Steve Edwards [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 preprocessing/templating complex?
also sprach Steve Edwards [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, but it introduces a lay
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 (a
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 46.24
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 instance
also sprach Brandon B. [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 => 999,6
also sprach Raj Mathur (राज माथुर) [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 question of which la
also sprach Paul Belanger [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 in the
database are not any
also sprach Shaun Ruffell [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:
http://madduck.
also sprach Administrator TOOTAI [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/ | http://two.sentenc.
also sprach Jeff LaCoursiere [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" runs an inst
also sprach Administrator TOOTAI [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 in a discus
also sprach Logan Bibby [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 keep scouting
f
also sprach Paul Belanger [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 am only talking about con
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 Joshua Colp [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 the device t
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 tha
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 voicemail
also sprach Torintino T [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 den gedanken verbesser
also sprach Luki [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 this before, b
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 configu
also sprach Danny Nicholas [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 against that
also sprach Jeff Peeler [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.
However, if I set e.g.
also sprach Doug Lytle [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, sizeof(returnexten), "
also sprach Doug Lytle [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 Asterisk t
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
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,
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 preven
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 voi
auth rejection for user "martin f. krafft"
;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" ;tag=as05fc20f4
I am not calling as username asterisk, but
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 th
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 h
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 ru
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 a
38 matches
Mail list logo