Re: [asterisk-users] Asterisk 1.6.2.13 - have asterisk reply from same IP address

On 10-09-23 07:01 PM, Mike wrote: > Hi, > > I have a server with multiple IP address, Asterisk binding with all of > them. I'd like Asterisk to reply to a SIP peer from the same IP address > as the peer used to register to Asterisk (as opposed to using the main > IP address all the time regardless

Re: [asterisk-users] rtp problem with 1.8.0-rdc1

On 10-09-23 05:40 PM, cov...@ccs.covici.com wrote: > 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 ce

Re: [asterisk-users] http://www.asterisk.org/downloads naming schema

On 10-09-22 11:45 AM, Klaus Darilion wrote: > Hi! > > Since some time the download of the newest Asterisk does not contains > the version number anymore, but is just called "asterisk-1.4-current.tar.gz" > > This gives me a tarball where I do not know the version without looking > into the tarball.

Re: [asterisk-users] Bug with Realtime?

On 10-09-16 09:43 AM, Dan Journo wrote: >> That's not a bug. Only when the phone registers or performs some sort of >> action >> (such as placing a call, etc...) does Asterisk query the database. If your >> phones have a short re-registration time this becomes less of a problem. > > How do you exp

Re: [asterisk-users] Bug with Realtime?

On 10-09-15 03:41 PM, Dan Journo wrote: > I think ive found a bug but need someone to double check. > > Whenever I issue a "reload" in Asterisk, any realtime extensions stop > receiving calls. > > I have to reboot the sip phones in order to get them to re-register. > > Can anyone see if they have a

Re: [asterisk-users] Asterisk 1.6.2.12 Download

On 10-09-15 12:13 PM, Paul Belanger wrote: > On Wed, Sep 15, 2010 at 11:54 AM, Ryan Wagoner wrote: >> Anybody else notice that the 1.6.2.12 download has a version and >> changelog for 1.6.2.12-rc1? >> > I can confirm, asterisk-dev notified. Odd, not sure how this happened, but I'll be rebuilding

Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

On 10-09-15 05:25 AM, Jonas Kellens wrote: > I think I've found it : > > Asterisk always reboots on this part : > > [Sep 15 11:16:32] -- Goto (azura,pbx,1) > [Sep 15 11:16:32] -- Executing [...@azura:1] > NoOp("SIP/INTERTELin-", "3252480333 = pbx formule") in new stack > [Sep 15 11:16:32] -

Re: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4?

On 10-08-24 09:59 AM, Gareth Blades wrote: > Zeeshan Zakaria wrote: > > If you are planning to move then perhaps look at 1.6 to give you a > longer lifecycle with that release. > I wouldnt go with 1.8 as I dont believe it is out of beta yet or if it > is, it is only just and I wouldnt risk it in a

Re: [asterisk-users] asterisk & distributed device state => res_jabber Versus res_ais

On 10-08-10 04:11 AM, Olle E. Johansson wrote: > > 26 jul 2010 kl. 18.13 skrev Leif Madsen: > >> On Asterisk 1.6.2, your only option for distributing device state is with >> res_ais. I've used it in a labbing system and it works well -- the caveat is >> that y

Re: [asterisk-users] asterisk on Vmware

and having an extension (my phone) dialed into the Milliwatt() application to listen for when audio quality starts to die, and note that as the max load / max calls / max memory for the system. * Good luck! :) Leif Madsen. -- __

Re: [asterisk-users] What do you use for Invoicing?

callrating.com/ The pay service is here: http://www.dthvoipbilling.com/ Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thur

Re: [asterisk-users] 1.8.0 beta2: courtesy tone being played to callee

On 7/31/2010 11:56 AM, cov...@ccs.covici.com wrote: > Leif Madsen wrote: > >> On 7/29/2010 8:30 PM, cov...@ccs.covici.com wrote: >>> Hi. I am using *1 in features to initiate a mix monitor recording. >>> However, when I hit *1, the callee hears the courtesy tone wh

Re: [asterisk-users] 1.8.0 beta2: courtesy tone being played to callee

on the other channel. I'm just speculating at this point though. Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Please test: STUN patch for Asterisk behind NAT

On 7/30/2010 2:00 PM, Philipp von Klitzing wrote: > Hi there! > > David has put up a patch to fix the STUN issues that has plagued Asterisk > 1.6 ever since that feature was introduced. Now we need testers to verify > the patch, so please grab the patch (or check out the SVN branch) and add > your

Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

ing? Is there a limit on the number of queues? (I'm sure there is a page on the website that answers most of these questions, heh :)) Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

Re: [asterisk-users] DUNDi questions

emove the information assigned to regexten for the peer upon (de)registration. Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Th

Re: [asterisk-users] Grab voicemail WAV file when done

On 10-07-27 10:02 PM, Michelle Dupuis wrote: >>> From: asterisk-users-boun...@lists.digium.com >>> [asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen >>> [leif.mad...@asteriskdocs.org] >>> Sent: Tuesday, July 27, 2010 9:22 PM >>> To: A

Re: [asterisk-users] Recording interface (pause/PLAY/RERECORD)

; n,Playback(${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}) same => n,Goto(handle_recording) exten => accept,1,Verbose(2,Recording accepted!) same => n,System(mv ${GLOBAL(CUSTOM_RECORDINGS)}/temporaryRecording-${RandomNumber}.wav ${GLOBAL(CUSTOM_RECORDINGS)}/${RecordedF

Re: [asterisk-users] Grab voicemail WAV file when done

On 10-07-27 08:38 PM, Michelle Dupuis wrote: > I need to grab the voicemail WAV file once the voicemail command is done. Is > there a hook to be notified that voicemail is done, and get the name of the > recorded file? Look at the 'externnotify' option to voicemail

Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?

On 10-07-27 06:08 PM, bruce bruce wrote: > :-) I knew someone would bring up FreePBX. I have FreePBX installed and > it's not good for Queues at all. It's using the reporting tool from > Areski and Areski has recently released an upgrade to it which again is > not what I want. > > There are few oth

Re: [asterisk-users] IAX bandwidth optimisation

On 10-07-27 07:56 AM, G star wrote: > How can I change that voice packet rate ? I think you want to read doc/rtp-packetization.txt in your Asterisk source. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digit

Re: [asterisk-users] "Register Attacks" End of ENUM ?

REGISTER attacks during the >> last days. >> > [...] > > Do like most of us are acting: use fail2ban. That's pretty much the solution to that problem right there: fail2ban. Leif Madsen. -- _ -- Bandwidth and Colocati

Re: [asterisk-users] chan_skinny still maintained?

week or so, you're probably going to be further ahead with chan_sip. Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] asterisk & distributed device state => res_jabber Versus res_ais

l -- the caveat is that your machines need to be on a low latency network (i.e. LAN). With Asterisk 1.8 (currently 1.8.0-beta1) you can use XMPP to distribute your device states over the WAN. I've made it work with the Tigase XMPP server. More info

Re: [asterisk-users] PBX Lua with Asterisk ODBC

exten => start,1,Set(MyResult=${ODBC_GET_MY_DATA(banana_lollipop)}) How you structure that in pbx_lua I'm not sure, but you "create" the functions with func_odbc.conf, which is probably the piece you're missing. Leif Madsen. -- ___

Re: [asterisk-users] Management interface

On 10-07-26 08:15 AM, Tony LaMear wrote: > I need graph the utilization of my t1s. Does anyone know of a plug-in, > code, or web interface I can use to help do this. I am currently using > Asterisk 1.4 I've been looking at the OpenNMS project recently. http://www.opennms.or

Re: [asterisk-users] URgent - capturing 'answered'

some special indication in SQL that I'm unfamiliar with. Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

sk Version field a little further down on the Report Issue page and select 1.8.0-beta1 from there. Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introduc

Re: [asterisk-users] Video IVR Asterisk ?

On 10-07-16 02:38 PM, Anita Hall wrote: > Is it possible to receive video calls using Asterisk and then process > them as an IVR ? One of our clients wants to set-up a video IVR system > in the US and we are evaluation possible options. > > Also, what is the bandwidth of receiving a video call in U

Re: [asterisk-users] Minimum modules required to run VoIP and CDR

Warren Selby wrote: > On Wed, Jun 30, 2010 at 8:50 AM, Frank Church > wrote: > > The DNS setup itself is fine. The sip module just seems to take too > much time to load. My modules.conf uses autoload=yes and it seems that > many unwanted modules are loade

Re: [asterisk-users] What‘s the best operating syst em suggest for Asterisk 1.6.2.9

I'm not entirely sure I see where he implied it was. His answer refers to the question, "I want to know what is the best OS for installing Asterisk...?" I like both CentOS and Ubuntu. The next edition of the O'Reilly Asterisk book will cover installing Asterisk on both OS's. Leif. Tiago Geada

Re: [asterisk-users] Using SetVar with System() is it possible?

Steve Edwards wrote: > On Sat, 19 Jun 2010, bruce bruce wrote: > >> Is it possible to harvest the output of system into a SetVar(variable)? >> >> exten => s,n,SetVar(var=system(asterisk -rx "sip show channels" | grep >> -c "(ulaw)") >> >> ??? any problem with the syntax? > > ) Your parentheses d

Re: [asterisk-users] Asterisk 1.6 + Jabber crashes

Michael wrote: > I am attempting to setup Asterisk to work with Gtalk. > > I am using the following versions: > Slackware Linux 12.0 > Asterisk 1.6.2.9 > GNU TLS 2.8.6 > Iksemel (svn v25) > OpenSSL 0.9.8o > > It all compiles however about 10 seconds after starting Asterisk it crashes. > > If the

Re: [asterisk-users] Ring + Music on Hold in the same call

Danny Nicholas wrote: > Not sure how this would work, but you could create a special MOH file > that was 10 seconds of ringing followed by the normal MOH – I know this > CAN be done, just takes a bit of trial and error. That's what I would suggest as well. You could use Monitor() initially to ca

Re: [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel

sean darcy wrote: > Richard Kenner wrote: >>> Is this bug alive in 1.6.2.9-rc1? I'm getting segfaults from chan_dahdi. >>> If it does effect 1.6.2.8-rc1, I'll just wait for rc2 to see if this >>> is my problem, instead of filing. >> I reported another instance of this today and it was fixed in

Re: [asterisk-users] Persuing the gtalk issue - not only jack-related

reasonably confident that the problem is in only one technology. [1] https://issues.asterisk.org/view.php?id=13812 Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Asterisk 1.6.2.8 Now Available

--[ UxBoD ]-- wrote: > - Original Message - >> The Asterisk Development Team has announced the release of Asterisk >> 1.6.2.8. This release is available for immediate download at >> http://downloads.asterisk.org/pub/telephony/asterisk/ > > Which release will http://issues.asterisk.org/view

Re: [asterisk-users] Cause and cure for "Exceptionally long voice queue length queuing to Local"?

David Cunningham wrote: > Hello, > > We're seeing lots of warnings like the following, running Asterisk > 1.6.1.12. Does anyone know the cause or cure? > > One explanation I've come across is that the peer is congested and > sending RTCP messages asking us to slow the RTP down. Is there any way >

Re: [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?

Greg Woods wrote: > I am still running 1.4 because of this bug: > > https://issues.asterisk.org/view.php?id=15129 > > I haven't tried any 1.6 versions recently; looks like some patches have > been checked in since I last tried it, although the bug is not closed. > So I may have to try it again wh

Re: [asterisk-users] Which issue is keeping you from updrading to1.6.2 ?

Danny Nicholas wrote: > If I'm going to bother with 1.6.2, I'll wait a few months for 1.8. But in > the spirit of your question: > (1) dialplan conversion > (2) loss of functions like Gosub Can you be more specific about what 1) and 2) mean? Leif. -- ___

Re: [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?

Olivier wrote: > As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an > issue with BLF-pickup which kept me from going further. Which bug number have you reported your issue in? Leif. -- _ -- Bandwidth and

Re: [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?

David Backeberg wrote: > meetme CLI arguments changed between 1.6.0 and 1.6.2 > Don't know where the delta was, and I haven't looked. > I prefer the new syntax, and especially prefer the 'concise' option, > but it might break features people have built in the past. > > Specifically, > 1.6.0 'meetm

Re: [asterisk-users] Lookup ${EXTEN} in database, update context/route if found... AGI?

Doug Lytle wrote: > Tim Nelson wrote: >> Greetings all- >> >> box on the same network. Instead of paying twice for the call to go out to >> the PSTN on one channel and back in on another channel, I'd like the ability >> to lookup the destination number in a MySQL database >> > > I use the my

Re: [asterisk-users] CDR to MS-SQL via ODBC issue

Tilghman Lesher wrote: > Okay, second idea is that you should very carefully examine your CDR table > layout and ensure that the columns that you have match EXACTLY what the > module expects you to have. If Asterisk expects you to have a column that you > don't (or the column type is wrong), that

Re: [asterisk-users] Asterisk 1.6.2.7 Now Available

Richard Kenner wrote: >> Should be the latest available on the Digium downloads site. It says >> version 1.6.2.0 but I've been using Skype for Asterisk on my 1.6.2 >> branch for quite some time (I just updated it last week). > > Hmm. So was I until it abruptly stopped working. It started again w

Re: [asterisk-users] Asterisk 1.6.2.7 Now Available

Richard Kenner wrote: >> The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. > > What version of Skype for Asterisk works with this release? Should be the latest available on the Digium downloads site. It says version 1.6.2.0 but I've been using Skype for Asterisk on my

Re: [asterisk-users] Asterisk 1.6.2.7 Now Available

sean darcy wrote: > If I'm reading the ChangeLog correctly 1.6.2.7 = 1.6.2.7-rc3. Right? Correct -- all releases are a direct copy of the last release candidate (in nearly all cases anyways). Leif. -- _ -- Bandwidth and Coloc

Re: [asterisk-users] sending T.38 fax negotiation problem

To make it clear, the change was merged to the 1.6.2 branch recently, and will not be in 1.6.2.7 as those releases candidates were made a couple of weeks ago. The changes will be available in the next set of release candidates, slated to be 1.6.2.8-rc1 sometime this week. Leif. Miguel Amez wro

Re: [asterisk-users] Issue with (pattern) matching extension

Andrew Latham wrote: > Are you guys talking about the Asterisk Cookbook Because that > could be released in the next 20 years at this point... The Asterisk Cookbook probably won't ever be released unless someone else wants to step up and start it. We (as in the authors of Asterisk: TFoT) had

Re: [asterisk-users] Issue with (pattern) matching extension

Danny Nicholas wrote: > Good snippet, Leif. It's easier to read 100 threads on this forum than the > 100 pages of the infamous "Asterisk Book" PDF. Infamous? Ouch :) Leif. -- _ -- Bandwidth and Colocation Provided by http://ww

Re: [asterisk-users] Issue with (pattern) matching extension

Jim Dickenson wrote: > I banged my head with a like problem a few days ago. > >>> exten => _fn-.,1,NoOp(ISN: ${DIALSTATUS}) > > n does not mean the letter n in a pattern it has a special meaning! Right! Be very careful about what you're matching! When it comes to matching things like 'N', 'X',

Re: [asterisk-users] More efficient dial plan for a list of selective inbound numbers

bruce bruce wrote: > I have a list of CLIDs prefixes that I want to use in a context. > > Basically, I want to do this but the list of prefix numbers is much > longer. List of prefixes (556,557,557,989.) > > [custom-inbound] > exten => _556,1,answer > exten => _556,n,playback(beep) > > exte

Re: [asterisk-users] How to record a call in a single file when transfered...

Carlos Chavez wrote: > Another question if I may, with variable inheritance is it possible to > do something like: Set(__CDR(userfield)=${INITIALCID})? That way I can > follow the call no matter where it is transferred to by having the > original outside callerid in the userfield. I'm not s

Re: [asterisk-users] Manipulating audio in asterisk

Slawek Sloma wrote: > Is there an option in asterisk to manipulate the audio in a call? > I would like to, for example change the voice of one caller but > without manipulating the audio that comes from another caller. > I have read about something called JACK but i don't know if i can use it > fo

Re: [asterisk-users] How to record a call in a single file when transfered...

Carlos Chavez wrote: > On Tue, 2010-04-20 at 15:04 -0400, Leif Madsen wrote: >> Carlos Chavez wrote: >>> I have a customer that needs to record all calls coming in and out. >>> The problem I am having is when a call comes in to the operator and it >>> is tr

Re: [asterisk-users] 1.6.2 No "soft hangup"?

others on the issue tracker (http://issues.asterisk.org). The default formatting though is 'module verb argument'. Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] How to record a call in a single file when transfered...

Carlos Chavez wrote: > I have a customer that needs to record all calls coming in and out. > The problem I am having is when a call comes in to the operator and it > is transferred to another extension. The first mixmonitor begins > recording when the operator picks up but the recording stop

Re: [asterisk-users] Extensions Reload | Asterisk Freezes ? 1.4

Positively Optimistic wrote: > We have what I consider to be a large dialplan (-= 1501 extensions (2559 > priorities) in 99 contexts. =-) > > If we have more than 10 or so channels up (all SIP, no TDM) and issue > the "extensions reload" command.. quite often, asterisk will completely > freez

Re: [asterisk-users] Evaluating Asterisk

e peak days for them). System heavily utilizes a database for call routing, CDRs, etc... and has been running steady for at least the last year. I've replaced a couple of smaller queue systems as well at other locations with Asterisk, and in all cases they are wishing they had don

Re: [asterisk-users] Asterisk and MWI with Exchange 2010

Jay Vocaire wrote: > I have been working on getting Asterisk and Exchange 2010 UM working > together, and so far I am pretty happy. The one thing not working right now > is MWI. > > I searched a bit and found this: https://issues.asterisk.org/view.php?id=13028 > > Now, please pardon me for bei

Re: [asterisk-users] Exceptionally long voice queue length errors...

James Lamanna wrote: > I'm seeing a lot of "Exceptionally long voice queue length" errors in > my logs, and then I seem to have a problem > where Asterisk will drop the registration for a significant number of > phones (they go UNREACHABLE), but then they > come back approximately a minute later. >

Re: [asterisk-users] Multicast Paging

Jonathan C. Bailey wrote: > I know this may be a bit off topic... > > > I'm trying to play a pre-recorded message to a group of Aastra phones using > multicast paging. I can page phone to phone without issue, but sending from > one of my servers to the phones results in garbled audio. Anyone el

Re: [asterisk-users] Is there any Diguim distributor in Lahore

Digium hasn't sold the X100P for something like 2 years now. Leif. Faheem wrote: > Hey, is there any Diguim distributor in Lahore,Pakistan? I need to buy > X100P. -- _ -- Bandwidth and Colocation Provided by http://www.api-di

Re: [asterisk-users] Context vs. Custom Context

Alejandro Cabrera Obed wrote: > Yes, Custom Context is a module from FreePBX in order to define calling > routes. I'd suggest using the FreePBX forums as I imagine the majority of people responding on this list are vanilla Asterisk users. Leif. -- _

Re: [asterisk-users] Context vs. Custom Context

Alejandro Cabrera Obed wrote: > Dear all, if I use the CustomContext module in Asterisk in order to > create new customized contexts for my extensions to managed > outbound/inbound calls, do these custom contexts replace the original > context defined in sip.conf, like "context=from-internal ???

Re: [asterisk-users] how to configure caller id

cool dude wrote: > hi leif, > thx for replying. can u plz ellabroate how to use 'o' optioan in Dial so > that callerid should work. There isn't much to elaborate on. Just enable the 'o' option in Dial like any other option, and that should pass through the callerID based on the description you

Re: [asterisk-users] too much sockets open by asterisk

CHEN XUEQIN wrote: > I have a similar problem when using AGI for call control. Also > udp port leak for some incomplete call. I wonder if the problem > is related to issue 16774. Only way to know would be to reproduce on a development machine, and then try testing the patch on 16774 to see if the

Re: [asterisk-users] Call Drops while doing assisted transfer from remote location

das sandesh wrote: > We are using Yealink T28 phones. Asterisk version: 1.4.21.2, dahdi: 2.2.0.2 This sounds like a few bugs which were opened (and recently closed) related to call transfers. I'm not sure when those bugs were introduced, but upgrading to a newer version of Asterisk may resolve

Re: [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app

Motorola Droid can run iPhone/iPod touch apps? Cool! :) Leif. Zeeshan Zakaria wrote: > Thanks Matt. This should be useful. I'll give it a try on my Motorola > Droid/Milestone. > >> On 2010-03-18 5:31 PM, "Matt Riddell" > > wrote: >> >> I've released another free ap

Re: [asterisk-users] Define an array of sip number in sip.conf

Zeeshan Zakaria wrote: > You'll have to type them all in manually. Or do what I did several > times, write a script in php which will generate the sip.conf with that > many extensions. Even better look into using realtime architecture, > where you can quickly generate as many extensions as you l

Re: [asterisk-users] how to configure caller id

cool dude wrote: > now i want when i call from my mobile to pstn line my mobile no should > be displayed in softphone Use the 'o' option in Dial(). Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] too much sockets open by asterisk

Ilya Pichugin wrote: > Hi All! > > I've set up Asterisk asterisk-1.6.2.2 > > > My question is why asterisk opens so many sockets and does not close > that? Sounds like an open bug in mantis. https://issues.asterisk.org/view.php?id=16774 Searching for "udp sockets" in mantis produced that issue

Re: [asterisk-users] Asterisk 1.4.24 DUNDi CLI commands not found

Klaus Darilion wrote: > These commands are also available for 1.4. Looks like the DUNDI module > is not loaded. Watch debug logging during module load for errors. > > Try "ldd /usr/lib/astersik/modules/res_dundi.so" and watch for > unresolved dependencies. Just to be clear here, I think you mea

Re: [asterisk-users] dialplan reload: not working with large dialplans

Andreas Brodmann wrote: > Lief, > > I'd be glad to receive your feedback. > > I don't think it's a limit of lines by itself. I haven't found any useful > debug information so far, but I think the dialplan parser stumbles > over something. > > The problem is reproduceable on different hareware, i

Re: [asterisk-users] Attended transfer broken in 1.6.0.25

;s > why I try to understand something about the patch release. Should I > perhaps file a bug as well but now against this specific branch? There is no need to file a patch. You can check the ChangeLogs from the archive files (even if you don't install from source), or you ca

Re: [asterisk-users] cli_originate malfunction after upgrade from 1.6.2.0 to 1.6.2.1-5

set of release candidates at http://www.asterisk.org/node/49915 Thanks! Leif Madsen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Remote Agents

Matt wrote: > Already found it -- but I was under the impression this was deprecated > and removed in 1.6? Try looking in the doc/ subdirectory of your Asterisk 1.6.2 source. You're looking for the building_queues.txt file. Leif. --

Re: [asterisk-users] dialplan reload: not working with large dialplans

Andreas Brodmann wrote: > the dialplan currently holds 1792 lines. It's a plain old .conf file. That's interesting, because I have a dialplan over 2400 lines and it seems to load fine... However, I'm using this on an ABE machine which is based on 1.4. Perhaps I'll try loading this on a 1.6.2 b

Re: [asterisk-users] string length in dialplan

kely a more efficient way of doing that, but I haven't gone through and looked at the functions to see if there might be a way of avoiding the loop :) Also, totally untested, I just wrote it in this email ;) Leif Madsen. http://leifmadsen.wordpress.com -- _

Re: [asterisk-users] Important security alert: update your?dialplans now!

Tilghman Lesher wrote: > On Monday 15 February 2010 18:01:11 Vinícius Fontes wrote: He probably means AgentCallbackLogin >>> While it has been deprecated, that hasn't been removed, either. If >>> an >>> enterprising person would like to try to fix it, I don't have an >>> objection. >> Wasn't

Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

Jason Parker wrote: > Brian wrote: >> Each time the server is rebooted Asterisk duly >> deletes the manually created /var/run/asterisk directory - quite why it >> does this I just don't know - perhaps it is a bug? > > Your assumption is incorrect. Some Linux distributions will empty /var/run/ >

Re: [asterisk-users] asterisk sudden restart - 1.4.18.1

das sandesh wrote: > Hi, > > Asterisk got stopped this morning after 20 minutes and phones went to > 'No Service' and then got started automatically after 20 min, as I could > see in the full log that asterisk got started at so and so time: > [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk E

Re: [asterisk-users] how to strip + from the caller-ID

he value to a temporary variable, then stripping from that when you reassign it to CALLERID() ? Also, you're using name, and I think you meant CALLERID(num)? exten => s/_+X.,1,Set(TMP_CID_NUM=${CALLERID(num)}) exten => s/_+X.,n,Set(CALLERID(n

Re: [asterisk-users] Asterisk 1.6.1.12 Now Available

Warren Selby wrote: > Is the new Fax For Asterisk being released in conjunction with this > release? If it's not already available, then it will be available very early next week. Leif Madsen. ___ -- Bandwidth and Colocation Provi

Re: [asterisk-users] 3 ed party sip client for Nokia sy

you have? I have the E71 and it has a built in SIP client already. Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium

Re: [asterisk-users] DEVICE_STATE

Philipp Kempgen wrote: > Magnus Benngård schrieb: > Set > call-limit=10 > (or any other value > 0) Actually, I believe call-limit is deprecated, and you can instead use callcounter=yes Leif Madsen. ___ -- Bandwidth and Colocation P

Re: [asterisk-users] Free Fax for Asterisk

e FFA license on all of them :) Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk registers with private IP

Joao Gomes Pereira wrote: > Nice!!! > Thanks a lot. > Its not the case (because Im using a fixed IP), but if the IP where dynamic? Setup a dynamic hostname that updates automatically using something like dyndns.com and then use externhost=myhost.dyndns.com Leif.

Re: [asterisk-users] Patch for app_dial.c: exit when just one ext is busy.

Leif Neland wrote: > Leif Madsen wrote: >> Leif Neland wrote: >> >>> I made a patch to app_dial.c to make Dial(ext1&ext2&ext3,tumeout,B) >>> return busy when just one extension is busy. >>> >> In order to have your patch consider

Re: [asterisk-users] "Dropping incompatible voice frame" error

what the issue (or issue number) was. Updating to the latest version from subversion in the 1.4 (or whatever other branch you're using) may help. Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Patch for app_dial.c: exit when just one ext is busy.

ng the license agreement. Otherwise, the developers (at least at Digium) won't look at the code or be able to offer any feedback. Thanks! Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing lis

Re: [asterisk-users] Parsing custom SIP headers

Philipp Kempgen wrote: > Just to be sure: Is there a dialplan function in Asterisk that > parses custom "name-addr"-style SIP headers for me? Try this: https://issues.asterisk.org/view.php?id=16268 Leif Madsen. ___ -- Bandwidt

Re: [asterisk-users] Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10 Now Available

Dave Cotton wrote: > On 19/11/09 15:37, Asterisk Development Team wrote: >> The Asterisk Development Team is pleased to announce the release of Asterisk >> 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate >> download at http://downloads.asterisk.org/pub/telephony/asterisk/

Re: [asterisk-users] Pbx-cards

f months to several months of service. Some of which allow you to use it like a pre-paid phone card, and is around a penny per minute. [* Yes, you can use a certain PCI modem which contains the appropriate chipset (not any PCI modem), but the time and money to get such a modem is not worth the t

Re: [asterisk-users] Changing labels on Phones

Julian Lyndon-Smith wrote: > We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a > "hotdesk" type system where anyone can log on to an extension - however > what I would love to do is relabel the phone with the current "owner" > when this logon happens. I know that I can cha

Re: [asterisk-users] 1.6.0.18-rc3: SendFAX causes restart

sean darcy wrote: > On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX > asterisk restarts: > > Before I file a bug, is there anything I'm missing? Does this happen on earlier versions of the 1.6.0 series prior to this release candidate? I'm curious if this is a regression,

Re: [asterisk-users] Sip incoming call issue with Asterisk 1.6

Eric van der Vlist wrote: > After a migration to asterisk 1.6, I don't receive sip incoming calls > anymore. > > As fas as I understand the SIP debug traces, my server receives the > request and reject it: > > ++ > <--- S

Re: [asterisk-users] Queue application in Asterisk 1.6

Bandino Jurumai wrote: > Can anyone tell me how to specify subroutine call with arguments in the > Asterisk 1.6 Queue application? > Documentation does not mention what is the syntax for specifying the > subroutine with arguments. If that functionality exists... try using ^ for the separator: Q

Re: [asterisk-users] Brandable SIP SoftPhone (Windows) ?

Gavin Spurgeon wrote: > Thanks to some answers on this list and another I now have a MultiTenant > system/setup working the way that I want it to, So now my next job is > to find a SIP SoftPhone that I can brand to my own company images and > so on. > > Again an OSS would be preferred, Even though

Re: [asterisk-users] Inquiry:Where to download Asterisk 1.4.13 for Debian server?

hadi motamedi wrote: > Dear All > Can you please do me favor and let me have the link to download the > Asterisk 1.4.13 for my Debian server ? Please let me know how to install > it . > Thank you in advance Well, 1.4.13 is quite old now, but you can find the older released versions of Asterisk

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