Un-topposted
Eric Wieling writes:
> Using qualify=10 ?
qualifyfreq=10 is fine, but Asterisk will not AFAIK do anything to a
call just because the peer goes unreachable qualify-wise. You are still
stuck with running a script that listens to qualify-unreachables and
does the appropriate thing to
Christopher Harrington writes:
> Since nobody seems to have come up with an Asterisk-specific solution, it
> sounds like the real approach here is something more generic.
> You can set up Nagios to fire off an event if it detects endpoints or
> infrastructure are suddenly dead. In particular, Nag
Mitch Claborn writes:
> Shouldn't asterisk somehow know when the agent disappears?
You are a bit out of luck since SIP session timers, the obvious
solution, cannot be set lower than 90 seconds.
rtptimeout set to e.g. 10 seconds may work, but you need to then set
rtpholdtimeout higher and hope t
isr...@gmail.com writes:
> Just my pitch in to post
> From a blackberry you can only top post there is no way of bottom
> posting
> So if I would have to wait to get to a computer to bottom post I would
> just never answer
Just delete the original post then. Not including context is perfectly
f
Gergo Csibra writes:
> Complaining about top posting on a list where's no moderation,
> no sanction if somebody top posting is pointless.
There is a sanction. People like me will score top posters lower and
soon not see their posts at all.
It is often a quick way to see if it is worth respondin
Carlos Chavez writes:
> I have a new install and the customer is complaining that they
> hear noise on all calls, no matter if it is internal or external, desk
> phones or softphones. The noise is only present when the user is
> speaking, not the remote side. The remote side does not hear t
Jeff LaCoursiere writes:
> Nifty! Love this Raspberry Pi. I keep thinking of new things I want to
> do with it. If I could only clone myself. I have a "video doorbell"
> project at the top of the list, if I don't find a USB FXS device :)
The Raspberry Pi has some problems with USB cameras and ot
Jeff LaCoursiere writes:
> The basic question was "has anyone made a USB FXS device work with
> asterisk". Now that I have additionally defended my architecture
> decisions, can anyone actually answer the question?
The Open USB FXS project is exactly what you want. It seems to be
discontinued.
Olivier writes:
> That's the point : to me, casual @pickupmark mechanism don't work with
> calls that entered into a queue : the extension rings but you can't pick
> the call up with a directed pickup.
> (For general pickup, that's another strory).
>
> (and I would be very pleased to be wrong)
T
Mitch Claborn writes:
> In our sales queue, we have wrapup time set to 15 seconds. When the
> phones are really busy, the operators would like the ability to bypass
> that 15 second wait and grab the next call in the queue. Is that
> possible? How to accomplish?
Slightly hacky solution which
"Adam K. Dean" writes:
> Hi,
>
> I was wondering if anyone has any experience in streaming a MeetMe conference
> so that others might listen in to it?
>
> It would be nice if the audio format could be AAC, but at first any format
> will do.
>
> I did come across this: http://www.voip-info.org/w
Markus writes:
> thanks! I wasn't clear enough in my original mail. What I meant is:
> the volume of the stream that a user is listening to is adjusted, but
> the volume of the conference itself is not changed! That means, a
> conference is going on, and everyone is listening to the same music at
Steve Edwards writes:
> Won't 200 simultaneous calls result in a lot of 'head thrashing' that
> would be avoided by staging the recordings to some form of
> non-mechanical storage and then copying the the recording at the
> completion of the call?
The extX family of file systems probably will no
Leandro Dardini writes:
> A single sata disk will be an unacceptable single point of failure. Get
> three disks and get in raid5 configuration. You'll gain in safety and
> speed.
RAID-5 is slower than single disks when it comes to write IOPS (a commit
is not done until the slowest disk has answe
"Raj Mathur (राज माथुर)" writes:
> Precisely. In fact, if a packet from 192.168.2.n is received on /any/
> interface, the response will always go out from the 192.168.2.X
> interface. (Barring some weird routing/iptables configuration, of
> course.)
This is only the case for TCP, because TC
"Kevin P. Fleming" writes:
> I've just looked into this a bit, and I don't see how using connect()
> would actually solve the problem. If we receive a UDP datagram from a
> SIP endpoint, we could use socket() and connect() to create a socket
> specifically for sending to (and receiving from) that
"Kevin P. Fleming" writes:
> I must be missing something. If a phone sends a UDP packet to
> 192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1
> interface on the Asterisk server?
The easiest way is that the Asterisk server itself is the router. Phones
on 10.0.2.0/24 have 10.0.2.1
"Kevin P. Fleming" writes:
> That's quite interesting; can you describe a scenario where this occurs?
Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24
and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to
move phones between the networks without changing t
Vladimir Mikhelson writes:
> But interestingly enough, yesterday morning I had zero (0) bytes in the
> swap file and still experienced missing DTMF detection on an outgoing
> call.
Executables do not get written to swap, their pages just get discarded
under pressure, and reloaded directly from t
"Owais Ahmad" writes:
> Hello guys,
>
> I need to be able to throw cdrs on more than one servers at a time. Please
> let me know how this can be done.
cdr_adaptive_odbc handles multiple servers. Just define several with
[foo] and [bar] and it Just Works.
/Benny
--
__
man, 07 05 2012 kl. 12:03 +0200, skrev Bart Coninckx:
> What about phones like the Unidata WPU-7800
> ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have
> experience with those? Would these also suffer from connection losses?
I don't know that particular phone, but dedicated wifi
Bart Coninckx writes:
> has anyone any experience in using Wifi smartphones as SIP clients?
Yes...
> Does this work properly?
It works nicely for home use for power users who can accept the odd lost
call and know how to restart the app or the phone when something goes
wrong. Unfortunately I ha
Jonas Kellens writes:
> I know you can do this by pressing "DND" on your IP-phone, but can this
> state also be set in the dialplan ?
You cannot actually achieve this by pressing "DND" on your IP-phone. All
that will accomplish is that the phone answers all calls with "busy",
but Asterisk will k
Markus writes:
> Does such a thing exist?
How does a2billing do it? It should be pretty easy in an AGI. If you can
afford a linear lookup per call, just grep through the array of prefixes
to find the ones matching a particular call, then pick the cheapest from
the results.
If you need something
"Jason W. Parks" writes:
> Thanks for the info. As we move forward, we'll be testing and making a
> phone selections. No doubt we'll run into this. Are you saying if the
> phone is stated to be a 10/100 phone, it still may not work at 10?
I must admit it isn't something I have looked at particul
"Jason W. Parks" writes:
> I can move my voice infrastructure to an IP-based one running 10Mbps,
> utilize existing wiring infrastructure, with the only cost outlay
> being low cost PoE managed switches (48 ports for about a grand), and
> it ends up a lot cheaper than upgrading the data network t
Carlos Alvarez writes:
> Perhaps you meant one that is not wi-fi? I would agree with that, I
> have a pile of totally useless wi-fi phones, they are all garbage.
Ascom has some fantastic Wi-fi phones. They are expensive, but they are
the only Wi-fi phones I have tried which actually work as phon
Olivier writes:
> 1. But, on your own 1.8.7 system, do you have something related to
> CURL when typing core show functions (or core show applications) ?
> I'm asking because func_CURL is missing from
> https://wiki.asterisk.org/wiki/display/AST/Dialplan+Functions
> (asterisk 1.8 version) which i
Olivier writes:
> I've seen that function CURL is missing from 1.8 but back in with 10
> (see wiki.asterisk.org).
I see the CURL function in Asterisk 1.8.7.1, found in the res_curl
module. In Fedora it is available in a separate package called
asterisk-curl.
If you do not get res_curl, it is li
"Luke Hamburg" writes:
> Carlos-
> Sorry if this is too much of a digression but this piqued my interest as
> I've been pretty happy with Polycom in my limited experience (haven't used
> the SPAs much, just Yealink & Polycom, and an occasional Snom here and
> there). If the config files were no
David Backeberg writes:
> Thanks for clearing that up. I was getting all excited that I could
> flash the PAP2T; I've always used regular voice tones over SIP with
> the PAP2Ts.
SPA-2102 supports T.38. If you ignore the WAN-port, it is practically a
PAP2T. The only time you cannot ignore the WAN
Tom Browning writes:
> My question is this: Is Asterisk simply relaying the client's DTMF
> signalling untouched or do I need to look at Asterisk more
> closely and turn some knobs.
I would recommend that you grab some wireshark traces before and after
the DTMF traverses Asterisk. It should be
ge...@riseup.net writes:
> Any idea how to solve this?
You can control src address selection with with ip route command.
E.g. if you know that you want to reach 192.168.0.0/24 with a source
address 192.168.0.50, you can do:
ip route change 192.168.0.0/24 src 192.168.0.50 scope link dev eth0
(Yo
Paul Hayes writes:
> If you time the *8 just right so it is being handled during the end of
> the Dial then I got:
>
> [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data
> is NULL
> [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data
> is NULL
Does this happe
Benoit Panizzon writes:
> Is there a way to get asterisk not to invent a CALLERID(name) if there is
> none?
>
> Id did try to set ${CALLERID(name)=""} but that resulted in From: ""
> and the displaying of this empty string on the subscribers phone.
I believe you have hit issue 17451,
https://
A J Stiles writes:
> Really, building packages from source *IS* *NOT* *HARD*, and it doesn't even
> take long anymore (on any target system with the grunt to run Asterisk).
> The only thing to beware of is, if "configure" complains that you need a
> package that you already have, then you ne
"Kevin P. Fleming" writes:
> OT: Take a look at 'systemd'; this is exactly what's happening there,
> and Fedora is likely to incorporate it into Fedora 16, and it will
> make its way into other distros after that.
It was incorporated into Fedora 14, and it is the default in Fedora
15...
/Benny
Paul Belanger writes:
> Sounds like asterisk was not told to generate a coredump, add the
> following, then you can generate a backtrace[1]:
>
> asterisk.conf
> [options]
> dumpcore = yes
The challenge with Asterisk and core dumps is that the Asterisk user
often does not have permissions to writ
Jeff LaCoursiere writes:
> Hasn't anyone managed to solve this with something better than a
> caching DNS server, which seems to only last a short while? What
> exactly is going on that is failing?
If your recursive DNS server returns errors quickly rather than actually
trying to look up the na
A J Stiles writes:
> (For my part, I'm actually surprised that nobody came up with a proper
> protocol for encapsulating the stream of zeros and ones that make up a fax
> transmission but rely on the precise timing inherent with a circuit-switched
> network, into something more suitable for se
maill...@lightspeed.ca writes:
> We've had several customers report since upgrading them to our new
> Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer
> works. No significant changes have been made to their SIP
> configuration, nor to their ATA configuration.
My testing of 1.
Mathieu Chouquet-Stringer writes:
> I've googled and pretty much tried all forms of the syntax but I've yet
> to make it work. For instance I tried not including stdexten and
> calling Gosub with (stdexten,${EXTEN},1(SIP/phone1)) but it didn't work
> either...
stdexten in the default extensions
"Kevin P. Fleming" writes:
> Why do you need a Local channel to do this? If extension 234 exists in
> some context, the Dial() statement in that extension can dial
> SIP/234-foo and SIP/234-bar itself.
Good point.
It can be a bit of fun keeping track of the phones when they are
added to or remo
"--[ UxBoD ]--" writes:
> Hi,
> With Asterisk 1.8 is it now possible to register the same SIP account at
> multiple endpoints and for both to ring when the associated extension is
> dialed ?
No. Our solution is to give each phone an account and make a
Local/234@somecontext which dials SIP/234-fo
Piotr Górski writes:
> So how to bill customers? Number portability makes it pretty impossible...
In the US, you pay the same to call a cell phone as you pay to call any
other phone. The callee pays for the airtime. This is a sensible
arrangement, as it allows for number portability and price co
Jonas Kellens writes:
> Hello list,
>
> I'm having some troubles with DTMF tones. When pressing numbers on a Snom
> phone, the DTMF-signal takes too long.
Which phone model? If 870, you may want to look at this thread:
http://forum.snom.com/index.php?showtopic=4084
You may want to experiment w
ken...@gnat.com (Richard Kenner) writes:
> Here's a possible design:
>
> - There's optionally a file in the config
> directory called "master_key". It contains just a string.
>
> - A CLI command "core encrypt " is added to Asterisk. It takes the
> provided string, encrypts it using the strin
Tilghman Lesher writes:
> Correct; and Asterisk needs to be started as root, even if it will drop
> privileges after startup. Do this, and there should be no problems.
Starting as root + dropping privileges is fine. Running configure as
root is not so fine; that basically makes building RPMS im
Sorry for resurrecting an old thread...
Tilghman Lesher writes:
> Out of curiosity, what platform are you running on? On most platforms
> that are able to run Asterisk, with the possible exception of Solaris,
> increasing the maximum file descriptor for use with select(2) is
> possible.
I am not
David Backeberg writes:
> So you're saying if you turn off t38 in sip.conf, you receive faxes
> successfully?
>
> Problem solved. Don't use T.38 in your particular environment.
That is not particularly useful advice. Fax over VoIP without T.38 is
inherently unreliable except in very controlled e
bruce bruce writes:
> Other than the price difference (2.5" is more expensive and can't find
> many of the 1TB or so) is there any preference, advantage, or
> disadvatage of chosing 2.5" HDD or 3.5" when it comes to the server
> operations or Asterisk operation?
There is no difference. Pick
Michelle Dupuis writes:
> Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com
> (also on the voip wiki)
>
> You get the cluster/heartbeat & replication without needing to add openSER
> or full HAlinux. A simpler approach - easier to config and manage
How do you handle replicat
cov...@ccs.covici.com writes:
> But it surpresses in both directions! I still want to hear the other
> end. For a test is there a way to turn off that feature to see if that
> is the cause?
Ah, so it isn't Asterisk doing silence suppression, it's Asterisk being
unable to handle that other devic
cov...@ccs.covici.com writes:
> Hi. I am having a very strange problem --aren't they all -- with the
> release candidate. I have softphone which talks to asterisk from behind
> nat -- the asterisk is on a public ip -- and when I hit mute on the
> softphone, all rtp traffic ceases! Now, a versio
Chris Owen writes:
> So I guess my question is what is the real purpose of the qualify
> setting in a non-NAT situation and can one safely set the
> qualification as something higher. I'd think something like 15 seconds
> would be more than enough for BLFs and the like.
The purpose is simply to
"Bryant Zimmerman" writes:
> Is there a way to force the connection to drop and reconnect after let's
> say 50 attempts.
Most firewalls have tools for removing specific connections from the
connection table. Alternatively a switch to SIP/TCP might help, but I've
never tried SIP/TCP with Asterisk
"Bryant Zimmerman" writes:
> As I look to move our systems to version 1.8 I am looking at making a
> change from mySQL to PostgreSQL.
>
> I love mySQL but am getting very concerned about i'ts new owners.
> Should I be able to move all my realtime stuff to PostgreSQL is it fully
> supported with a
Hose writes:
> The most straightforward way would be to just define explicit patterns.
> Obviously that works, but doesn't seem scalable in terms of maintenance.
I don't think that maintaining the list in the dial plan is all that
bad, actually. Dump it in its own context and file...
If that is
Christian Weeks writes:
> Hello
> I purchased an AEX800 card to replace the ageing cheap channel bank/T1
> card solution a few months ago, assuming that it would be a more robust
> solution for my small scale phone system. However, it appears to be
> anything but that.
>
> Originally implemented
Please don't top-post.
Geraint Lee writes:
> to get accurate cdr's i just use a "border" server to send every call
> through that logs cdr... doesn't matter how many times it gets transferred
> internally the "border" server still gets accurate records of the whole
> call.
That is what we do to
Carlos Chavez writes:
> I have searched for some time but I have not found an asnwer on how to
> fix the CDR when a call is transferred. The problem is that if someone
> dials a cell phone and then transfers the call to another extensión the
> CDR for the cell call stops and there is no wa
"Alan Lord (News)" writes:
> Fedora is *not* a server operating system and not one I would choose to
> run asterisk on.
Fedora is an excellent server operating system. I manage more than a
thousand installs if you count virtual ones.
> I would recommend using either CentOS or a Debian/Ubuntu S
Please do not top post.
Sherwood McGowan writes:
> I'm going to go ahead and say that while I'm not one of the
> developers, I think it's safe to say that you cannot record to a file
> and play it back at the same time. Probably something like file
> locking (for the record, locks it from acces
SIP writes:
> Spammers sign up to the Asterisk mailing list and send spam once in a
> while. My spam filter rejects it, and bounces the emails back to the
> Asterisk list, which then drops me from the list because it got a single
> bounce.
Don't ever bounce spam! You WILL get blacklisted for
Randy R writes:
> I'd think twice about trying this, taking into account the recent
> spate of attacks to so many of us coming from Amazon EC2 and
> particularly their answer to complaints, which was something like
> "Deal with it."
Indeed, my personal threshold for dealing with EC2 traffic has
Travis Langhals writes:
> [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on
> SIP/5211-0078
Is SIP/5211 a Linksys or a Grandstream or something else?
Do you have relaxdtmf=no?
Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10?
/Benny
--
_
Ira writes:
> running on Centos 5 with "yum update" showing it's all up to date.
> I think it's 5.2 or 5.4, I just don't know how to get it.
cat /etc/system-release. Or redhat-release, if system-release doesn't
work on CentOS yet.
It should be 5.5 if you're all up to date.
/Benny
--
__
I would appreciate it if you didn't top-post.
das sandesh writes:
> Hi Benny...
>
> DTMF tones are heard at the SIP phones side and not the other
> party...'server side' means from the Asterisk side.from the
> wireshark captures it appeards that the dtmf digits were sent from the
> aster
das sandesh writes:
> In the wireshark capture attached we could see the random dtmf
> digits have been sent from the server side.can anyone share your
> thoughts in regards to this...
Which end hears the DTMF, the SIP phones or the phones on the PSTN?
When you say "sent from the server
Quy Pham Sy writes:
> they've just named as xxx.wav so I guess there is no problems with copying
> or linking solutions.
You're simply lucky that the header is short enough to not sound too
bad.
/Benny
--
_
-- Bandwidth and
Frank Church writes:
> Is there a database of MAC address prefixes used the common VoIP
> devices. I see the Linksys Sipura devices state with 00:0E.
>
> Does the same apply to other Linksys VoIP equipment?
>
> Is there some way VoIP equipment allow themselves to be identified by
> requesting dat
Zeeshan Zakaria writes:
> making use of the fact that both Cat5 networks and BRI ports
> don't use all the 8 pins, so why not use extra wires in the cable for
> something useful instead of wasting them.
For Ethernet, this is only true for 10Mbps and 100Mbps. Gigabit and up
uses all four pairs.
"William Stillwell (Lists)" writes:
> I have several remote phones that experience a slight call delay when
> answering phones, ie, they will answer, speak a few words, and then the
> remote caller will hear them, and the first half is cutoff?
This is actually a somewhat common problem in SIP
Håkon Nessjøen writes:
> But for a few years ago, I did some testing with Local/ channels, and they
> seemed somewhat unstable in large quantity.
>
> Are they more safe now? Is it safe to use local channels with the /n
> modifier as queue members? (i need the n modifier to be able to count
> cont
Is it possible to disable silence suppression by adding silenceSupp:off
to the SDP Asterisk transmits even when Asterisk is using internal
timing? As far as I can tell Asterisk stops sending silenceSupp:off when
internal timing is on, which does make sense, but I would like to avoice
silenceSupp fo
Steve Edwards writes:
> Wouldn't a "set time" function be more usefull?
I really like that idea. Enough that I could try to lobby internally for
funding, if you know someone who is willing to do the work...
/Benny
___
-- Bandwidth and Colocation Pr
Gavin Spurgeon writes:
> iSip (£2.39)
> http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8
I have been very impressed by the audio quality from iSip, at least from
the "other end" so to speak. It shares the basic flaw of not being able
to run in the background wi
hbk writes:
> Where to look for forgotten DTMF detection settings?
Try relaxdtmf=no. sip show settings to check that it worked.
/Benny
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSC
Leif Neland writes:
> Because I might have more phones than mouths :-) If I'm busy with one
> conversation, I don't want to hear another phone ring. I might have a
> desktop and a portable phone.
This use case is indeed very difficult to implement in Asterisk dial
plans today. Especially if you
SIP writes:
> It may work in Austria, and may even be valid in Austria. But if that's
> the case, it's because Austrian dialing is a complete hack -- NOT
> because that's the way it's intended OR designed.
Err no? It's perfectly sane, and it was intended and designed that way.
You are providing
Philipp Kempgen writes:
> Just to be sure: Is there a dialplan function in Asterisk that
> parses custom "name-addr"-style SIP headers for me?
No.
> so I guess SIP_PARSE_HEADER() would need an index argument, just
> like SIP_HEADER().
Yes, Asterisk array handling leaves a bit to be desired as
Leif Neland writes:
> I think a modification should be done around here to return busy if just
> one channel was busy (only enabled if an option on dial is set)
> in asterisk-1.6.0.15/apps/app_dial.c, line 610
That is doable, but it can result in a bad experience for the caller.
The Dial() is li
"C. Chad Wallace" writes:
> So, does anyone know of a way to detect whether a call from a SIP phone
> is the first step of an attended transfer or an original call?
This is impossible. At that point the phone has done this:
1) Put the original caller on hold
2) Made a new outgoing call
At so
Warren Selby writes:
> I believe I spoke with Aastra and Snom at the Astricon tradeshow and
> they said they support it on their newer models as well.
For Snom the enhancement request is SCPP-227, but I don't believe it has
been implemented. I can't find it in any release notes at least. The
gen
JT writes:
> I'm struggling with an intermittent crosstalk issue resulting in a
> caller's audio being broadcasted to other calls (only one way as they are
> unable to hear the others listening in).
This may be a long shot... I have experienced this when two SIP phones
had the same IP address (a
"Cary Fitch" writes:
> Is there a plain 64K codec that would simply pass through the SIP server and
> be handed off to a PRI or phone co. trunk on a T1 on the other side of the
> SIP server? Digital 64K telco sounds very good as a phone conversation.
You can't get a guaranteed bit-for-bit ident
Tilghman Lesher writes:
> Many consumer-grade switches effectively turn into hubs when more than 1023
> MAC addresses are seen on a network. This may be done intentionally by
> somebody attempting to eavesdrop on all network connections sent through
> the switch. A reboot of the switch might (t
Steve Edwards writes:
> atftpd can do PCRE substitutions to transform a requested file name into
> something else. I've not used this facility, but I'm guessing you could
> transform:
>
> SIPDefault.cnf -> cisco/SIPDefault.cnf
> sip.cfg -> polycom/sip.cfg
> spa841.cfg -> sipur
Olivier writes:
> Most (if not all) IP phones support provisioning through DHCP/TFTP.
> The trouble is some phones seem to require to store their config files in
> TFTP root directory.
A lot of IP phones support HTTP instead of TFTP. This helps, because it
is fairly easy to write a script which
"C. Chad Wallace" writes:
> It would only be trying one agent at a time for each waiting queue
> member...
Would it? Almost all our queues are on a ringall strategy.
> I don't know how expensive it is to open and close a Local channel and
> do a DB lookup, but I wouldn't expect it to be a real
Benny Amorsen writes:
> Would it perhaps work to simply Wait(30) if the call is rejected by the
> phone? If the Queue assumes that the phone is busy for those 30 seconds,
> I have accomplished my goal. It's worth a shot.
This works! Actually I tried out Wait(1000), but that work
"C. Chad Wallace" writes:
> OK, I decided to write it up in AEL. It's incomplete and untested, but
> it probably gets the idea across a little better.
>
> context agentcalls {
> _2XX => {
> Set(AGENT=${EXTEN}); // Assuming agent ID is extension.
>
> if (${EPOCH}>${DB(AgentPaused/
Elliot Otchet writes:
> That shouldn't be too hard to accomplish. If you've got the addons
> (and mysql) installed you could store them in a MySQL table
> (timestamp, device) and have a cron job set to run at X frequency that
> un-pauses the queue members via AMI. Don't want to go to MySQL? Use
>
Lenz Emilitri writes:
> You could configure them as agents and have them log off automatically
> after a while they're not responding.
Agents have to log in and wait for calls though, don't they? There used
to be AgentCallbackLogin, but that has been replaced by dialplan code
and chan_local.
Ot
Elliot Otchet writes:
> Have you tried autopause=yes in your queue configuration? You can then
> unpause the member by either the dialplan (e.g. having the cell phone
> user "log back in") or using an AMI based program to change the
> "paused" state.
>
> You can read more about the latter here:
>
We have the possibly rather unique setup where we have cell phones
posing as SIP devices. The SIP registration for those unfortunately
doesn't go away just because the phone is off, since the registration is
done by our cell-phone<=>SIP gateway, and that gateway has no way of
knowing whether the ph
Gordon Henderson writes:
> I use Draytek Vigor 2820's these days. Mostly (when not having something
> more "corporate" or dealing with geeks who want a Linux based one) Built
> in hardware assist VPN too. They do have a SIP ALG, but it's turned off by
> default (the earlier ones had it turned
Vieri writes:
> Hi,
>
> I'm wondering if someone can share their thoughts on how to implement a
> system that periodically checks active channels which have been up for more
> than X minutes and plays/injects a sound file. The idea is to simply warn
> users that they've been on the phone for q
Örn Arnarson writes:
> I'm seeing the same behavior in 1.6.1.6.
>
> Any info on this?
It would be helpful if you copied the exact error message involving the
username field.
/Benny
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Jared Smith writes:
> In a nutshell, you can pass the test without having any experience on
> Polycom IP phones and Digium cards, as long as you know how to use
> Asterisk itself.
You certainly can, but I think it's worth it to invest ~30 minutes
beforehand so you know where you put IP addresses
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