Re: [asterisk-users] Iridium integration / gateway

Hi, This looks like it may work for you. https://www.iridium.com/products/beam-potsdock-extreme-docking-station/ Harry On April 3, 2018 9:35:53 PM MST, Bertrand Lupart wrote: >Hello, > > >> Thanks for reply, but this is irrelevant, I'm looking for an >*Iridium*

Re: [asterisk-users] Panasonic PBX connect to Asterisk

Hi, You need to find out more about the configuration of this specific TDA600, as it could be either POTS or E1, once you know that, you can determine what options are best. -Harry On 09/13/2016 10:51 PM, Ikka Tirtawidjaja wrote: Dear Harry, Thx for the explanation. My team manage

Re: [asterisk-users] Panasonic PBX connect to Asterisk

bank on the second interface, and handed 4 POTS lines to the Nortel Key System. The key is to give your self the most flexibility to change later, and preserve your existing investment. -Harry Thanks in advance, Regards, Ikka -

Re: [asterisk-users] 1000 analogue lines with asterisk

, around here it averages $100/drop in office environments, if your only doing one run to each room, it gets even higher, this is where the real cost of using VoIP handsets comes into play. -Harry On 02/17/2016 07:38 PM, Jeff LaCoursiere wrote: That would be the expensive route. The inexpensi

Re: [asterisk-users] 1000 analogue lines with asterisk

, taking up about 2U of space per 2 channel banks. You could service this with six eight port T1 cards, or with eleven/twelve quad T1 cards. I would distribute across two, three, or even four servers for redundancy/resiliency and load balancing. -Harry On 02/17/2016 12:16 AM, Goke Aruna wrote

Re: [asterisk-users] [OT] switches

On 02/24/2015 09:30 PM, Thufir wrote: On Fri, 20 Feb 2015 13:05:56 -0700, Harry McGregor wrote: For a very basic setup it would work, but I would suggest POE at a minimum, and vlan support if possible. Gigabit uplinks, 10/100 for the poe ports http://www.amazon.com/NETGEAR-ProSAFE-M4100-D10

Re: [asterisk-users] [OT] switches

ords=netgear+poe and Gigabit all ports http://www.amazon.com/Netgear-ProSAFE-GS110TPv2-Gigabit-GS110TP-200NAS/dp/B00LW9A328/ref=sr_1_5?ie=UTF8&qid=1424462577&sr=8-5&keywords=netgear+poe -Harry On 02/20/2015 12:58 PM, thufir wrote: Pardon, this might be off-topic. I'm read

Re: [asterisk-users] probleme with web-meetme.3.1.0

Hi matt I use Asterisk release 1.6.1. So if it's my version which is source of problem could you suggest an application like web-meetme to manage asterisk conferences ? Regards Harry 2009/9/3 Matt Riddell > On 4/09/09 3:24 AM, harry R wrote: > > Hi everybody > > > >

[asterisk-users] probleme with web-meetme.3.1.0

inc on line 35 * note that I want to use sqldb.conf for users authentication and not ldap. Regards Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Regi

Re: [asterisk-users] application order when you make a call

> > Yeah I know that flag, it's 'r' option but If i remember, that's usefull > only if your terminal don't generate ring tone. > I don't think that answer() is necessary with that option. > > Harry > > I just test ringing() application it'

Re: [asterisk-users] application order when you make a call

you'd need to Answer() the call first. > > Gordon > Yeah I know that flag, it's 'r' option but If i remember, that's usefull only if your terminal don't generate ring tone. I don't think that answer() is necessary with that option. Harry

Re: [asterisk-users] application order when you make a call

> If you're using Asterisk to "bridge" an incoming call to a device (eg. a > SIP phone), then you just need > > Dial(... > > No need to Answer (as that then starts to cost the caller money if calling > via the PSTN or some PSTN to VoIP bridge to get the call into your Asterik > box in the first p

[asterisk-users] application order when you make a call

Hi I have a question about application order when a do a call to a terminal Do I need to use application in this order ? Ringing() Answer() Dial() or in this order Ringing() Dial() Answer() Suddenly I have some doubt. Personally, I usually do it like the first one. Regards. Harry

[asterisk-users] create applicationmap and use it in dialplan

al(SIP/111,,Tt) but How can I exactly use features or application map which I created in features.conf, in my dialplan ? Regards. Harry. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoeni

Re: [asterisk-users] application missed in asterisk 1.6.1 - SetCallerID()

> Read the UPGRADE.txt > > Solution is to use functions instead: > > Set(CALLERID(name)); > Set(CALLERID(num)); > Set(CHANNEL(language)); > etc Thanks again for solution Harry. ___ -- Bandwidth and Colocation Provided by ht

[asterisk-users] application missed in asterisk 1.6.1 - SetCallerID()

Hi A few day ago, I notice that some applications missed in asterisk 1.6.1 release even if *.so file which normally create them were compiled during Asterisk install. SetCallerID(), SetCIDNum(), SetCIDName(), SetLanguage() ... and maybe so more. anyone already notice that to ? If it's not normal

Re: [asterisk-users] followme app

> Hope this helps. Again, bear in mind that we are new to this so if > > someone suggests a better way, they are probably right :-) - John > Thank you John for this example. I'll try to implement it and give you a backup if I have any questions or

[asterisk-users] followme app

Hi Someone may give me an example of followme() application using in a dialplan (including what to configure in followme.conf) ? I use asterisk 1.6.1 so if your example can match to that release it's will be wonderfull. Thank in advance.

Re: [asterisk-users] asterisk 1.6.1.1 + Asterisk GUI v2.0

reinstall GUI 2 again and we will see :) And sorry I dont try to control a TDM400P yet. So no help for you :( Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http:

[asterisk-users] asterisk 1.6.1.1 + Asterisk GUI v2.0

Hi Has anyone already use asterisk 1.6.1.1 with asterisk GUI last release ? I'm trying it but I have this problem : Just after I logged, I have system status main page but no other links where I can click to go to other pages (remember left panel!) Regards

Re: [asterisk-users] mysql sip realtime

>Have you set the qualify column in the sip table? yes and default set to yes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-user

Re: [asterisk-users] mysql sip realtime

yes auto clear : 120 remind : I use asterisk 1.6, mysql 5 Harry. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing lis

Re: [asterisk-users] mysql sip realtime

> > Well, it shouldn't. The Contact header should identify how to contact the > peer, and that is currently saved into the database on register (unless you > have updates turned off). Which column are concerned by ? my update is on because some column are dynamically updated each time a terminal

Re: [asterisk-users] mysql sip realtime

> Unfortunately not, I built ours myself. > ok thank for all these advices and sol ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk

Re: [asterisk-users] mysql sip realtime

> > > I try CLI command sip prune realtime and my peer infos was > > perfectly updated when I do sip show but have you any idea > > of how I can do that automatically ? > How are you updating your sip table? Are you doing it manually or have > you built an interface for it? If you have built an i

Re: [asterisk-users] mysql sip realtime

> > All generic parameters are still taken from sip.conf and you must set > rtcachefriends=yes > > If you change anything in your mysql sip table you do not need to reload > the modue, what you need to do is > sip prune realtime > from the CLI > > As stated previously, you should never have to rel

Re: [asterisk-users] mysql sip realtime

> > When I reload chan_sip.so, it seems that connected terminals are no > longer > > detected by Asterisk because when I tape CLI command "sip show peers", > > there is no results displayed. Any reflexions about that ? > > They won't be found in the CLI command until Asterisk receives another > pac

[asterisk-users] mysql sip realtime

? When I reload chan_sip.so, it seems that connected terminals are no longer detected by Asterisk because when I tape CLI command "sip show peers", there is no results displayed. Any reflexions about that ? thanks in advance for your

Re: [asterisk-users] mysql error (err 2002)

Hi I just solve my problem. It seems just because Astérisk didn't find (or maybe it's mysql fault) socket file path. so I just look in my my.cnf and copy socket file path in res_mysql.conf like this dbsock = Hope it will be helpfull if someone else (will) have this problem. regar

Re: [asterisk-users] mysql error (err 2002)

>mysql -uasterisk -pasterisk asteriskdb When I do that in a linux terminal it works. But I always have this err 2002. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now

[asterisk-users] mysql error (err 2002)

table for sip named sip_terminal. So any tips to resolve my error message will be welcome. regards. Harry. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now

[asterisk-users] res_ldap.conf

anyone already used the realtime driver for LDAP in order to interact Astérisk with Active Directory ? regards Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register

[asterisk-users] Asterisk + realtime applications

directory ? thanks in advance for your answers. Harry. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To

Re: [asterisk-users] Asterisk + CDRTool

2009/8/14 Pascal Bruno > Did you get CDRTool to work with Asterisk or Areski's CDR Stats? Hi finally use Areski but until now I dont try all features, just CDR Report button. But I had a quick look on other button and it seems work. ___ -- Bandwidth a

Re: [asterisk-users] Asterisk + CDRTool

Hi I just solve my problem today. Just a package on redhat that I need install. H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asteris

Re: [asterisk-users] Areski CDR + Mysql + asterisk 1.6

2009/8/13 Ishfaq Malik > Have you configured your /etc/asterisk/cdr_mysql.conf file? Yeah I configure it. Now everytime I do a call, a CDR line is added in my table "cdr". ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Ast

Re: [asterisk-users] Asterisk + CDRTool

I linked to Areski's CDR Stats (which I've used a few times): > > http://www.areski.net/asterisk-stat-v2/about.php > > Asterisk-Stat is a visualisation layer for Asterisk CDR statistics which > are pulled from a database. It provides graphs as well as allowing you > to get more information on indi

[asterisk-users] Areski CDR + Mysql + asterisk 1.6

"ipbx"); define ("PORT", "3306"); define ("USER", "asterisk"); define ("PASS", "asterisk"); define ("DBNAME", "asteriskdb"); define ("DB_TYPE", "mysql"); define ("DB_TABLENAME", &q

Re: [asterisk-users] Asterisk + CDRTool

>CDRTool operates on CDRs generated by RADIUS servers into the standard > >'radacct' schema, along with some custom attributes added by OpenSER. > >CDRTool is designed for use with OpenSER, not Asterisk. > src http://www.voip-info.org/wiki/view/Asterisk+GUI CDRTool

Re: [asterisk-users] Fwd: User Authentication in sip.conf

> > Dear all, > I want to setup the incoming calls, that don't use authentication in > sip.conf file. > My configurations as follows, > > [2000] > type=peer > host=dynamic > insecure=port,invite; (both) > context=Testing > > But when I call '2000', I noticed the following message

[asterisk-users] Asterisk + CDRTool

Hello Anyone who have already use/configure Asterisk with CDRTool ? Or maybe can suggest another CDR GUI ? regards. Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona

Re: [asterisk-users] regcontext regexten

amic ; phone will register w/ Asterisk > secret=mysecret > regcontext=some-context > regexten=6123 > > Thank Jared. So I have one more and last question about regcontext. Where do asterisk create context "some-context" ? I see context by taping "dialplan show s

[asterisk-users] regcontext regexten

Hi Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? thx regards Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register

Re: [asterisk-users] dialplan tips

> exten => 101,1,Ringing > > exten => 101,n,Answer() > > exten => 101,n,Dial(SIP/quentin,10) > > exten => 101,n,Goto(101-${DIALSTATUS},1) > > exten => 101-NOANSWER,1,VoiceMail(1...@default,u) > > exten => 101-NOANSWER,n,Playback(vm-goodbye) > > exten => 101-NOANSWER,n,Hangup() > > exten => 101

Re: [asterisk-users] sip.conf parameter and sip msg between server <-> client

? Is it an issue to limit incoming/outgoing call for a device ? - 'callcounter' is it an issue to limit incoming/outgoing call for a device ? - I read that 'call-limit' or 'busylevel' (in asterisk 1.6) is an issue to limit incoming/outgoing call on a device. Is it

Re: [asterisk-users] dialplan tips

exten => 101,n,Goto(101-${DIALSTATUS},1) exten => 101-NOANSWER,1,VoiceMail(1...@default,u) exten => 101-NOANSWER,n,Playback(vm-goodbye) exten => 101-NOANSWER,n,Hangup() exten => 101-BUSY,1,Playback(busy) exten => 101-BUSY,n,Wait(3) exten => 101-BUSY,n,VoiceMail(1...@default,b) ext

[asterisk-users] sip.conf parameter and sip msg between server <-> client

Hello I have few questions : - what's the difference between a subscribe request et a register request ? - in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please someone could explain how doest it work because I think i'm a little bit confuse. - if I configure a sip terminal in

Re: [asterisk-users] asterisk users

database 'astdb'. I notice that every soft or physical phone registred on the asterisk server are referenced there. Harry 2009/7/24 Jonathan Moore > On Fri, Jul 24, 2009 at 9:22 AM, harry R wrote: > > > How can I say to asterisk server : "don't accept other regi

[asterisk-users] asterisk users

Hi I have a new question. Here the situation : I use softphone on 2 computers (soft1 and soft2) located on the same subnetwork. When I register on asterisk server using soft1 with one user (e.g JOHN) which I declared in sip.conf I can register again with this same user using soft2. Is it normal ?

Re: [asterisk-users] dialplan tips

- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *harry R > *Sent:* Friday, July 24, 2009 7:44 AM > *To:* asterisk-users@lists.digium.com > *Subject:* [asterisk-users] dialplan tips > > > > Hi

[asterisk-users] dialplan tips

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general paramet

Re: [asterisk-users] PRI problem

I had the exact same problem and errors some time ago (search the archives for "PRI dropping #2") using Asterisk 1.4.18, Zaptel and a Digium TE121. I tried all kind of things, had telco technicians come out and whatnot. The solution was two-folded - 1) I reinstalled my server, 2) I updated to Aster

Re: [asterisk-users] PRI dropping #2

hours. 2009/3/26 Harry Vangberg : > It's 2 feet from the Nokia network terminal from the telco. > > 2009/3/26 Jared Smith : >> On Thu, 2009-03-26 at 20:24 +0100, Harry Vangberg wrote: >>> This sounds like what is happening, and is in order with what one of >>&g

Re: [asterisk-users] PRI dropping #2

It's 2 feet from the Nokia network terminal from the telco. 2009/3/26 Jared Smith : > On Thu, 2009-03-26 at 20:24 +0100, Harry Vangberg wrote: >> This sounds like what is happening, and is in order with what one of >> the technicians said - I was about 20 dB below their amplit

Re: [asterisk-users] PRI dropping #2

They didn't show up in the list archive. I'm terribly sorry. 2009/3/26 Steve Howes : > The first and second times were sufficient. > > On 26 Mar 2009, at 19:24, Harry Vangberg wrote: > >> Okay. Trying third time, sorry for that, might just be my mail client, &g

Re: [asterisk-users] PRI dropping #2

amplitude. The equipment shall react within 12 ms by issuing AIS." This sounds like what is happening, and is in order with what one of the technicians said - I was about 20 dB below their amplitude, theirs being 2048. Does this make *any* sense? 2009/3/26 Harry Vangberg : > Hey, >

[asterisk-users] Fwd: PRI dropping #2

re than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS." This sounds like what is happening, and is in order with what one of the technicians said - I was about 20 dB below their amplitude, theirs being 2048. Does this make *any* sense? 2009/3/26 Harry V

Re: [asterisk-users] PRI dropping #2

ms by issuing AIS." This sounds like what is happening, and is in order with what one of the technicians said - I was about 20 dB below their amplitude, theirs being 2048. Does this make *any* sense? 2009/3/26 Harry Vangberg : > Hey, > > I wrote yesterday about PRI dropping, which t

[asterisk-users] PRI dropping #2

Hey, I wrote yesterday about PRI dropping, which turned out to just be a regular reset of unused B-channels. This time there's a real issue. As noted earlier I have an ISDN-30 connection, a Digium TE-121 with VPMADT032 echo cancellation. These are my configurations files: == /etc/zaptel.conf load

Re: [asterisk-users] PRI dropping

e /etc/asterisk/zapata.conf? > > > > On Tue, Mar 24, 2009 at 3:21 PM, Harry Vangberg wrote: >> >> Hello, >> >> I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo >> cancellation. Every 30-60 minutes I experience PRI dropping. >> >

Re: [asterisk-users] PRI dropping

And nevermind. I just noticed that I didn't have warnings this time, and it's perfectly normal. 2009/3/24 Harry Vangberg : > Hello, > > I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo > cancellation. Every 30-60 minutes I experience PRI dropping. &g

[asterisk-users] PRI dropping

Hello, I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo cancellation. Every 30-60 minutes I experience PRI dropping. @@@ /etc/zaptel.conf: loadzone=dk defaultzone=dk span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 @@@ @@@ /etc/asterisk/zapata.conf [channels] switch

[asterisk-users] BACKGROUNDSTATUS not available?

Hello, According to this page: http://bugs.digium.com/svnstats/asterisk/trunk/2006-09.html#298 a BACKGROUNDSTATUS variable was added in 2006 with revision 43814. As far as I can understand it should return different values wether the file played with a Background() command was played through or int

Re: [asterisk-users] Outside call not coming through

Screwed up really bad. This is the correct config and sip debug: ## sip.conf [general] context=incoming register => 36946811:[EMAIL PROTECTED]/1234 port=5060 bindaddr=0.0.0.0 srvlookup=yes ## extensions.conf [incoming] exten => _X.,Background(hello-world) ## sip debug (updated) <--- SIP read fro

[asterisk-users] Outside call not coming through

ceived=62.107.1.48;branch=z9hG4bK-d8754z-c59c0dcc191940e4-1---d8754z-;rport=5060 Max-Forwards: 16 Contact: To: From: "Harry";transport=UDP;tag=688c7f1d Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OP

[asterisk-users] Playing mp3-files – will it b e OK?

Hello 99% of all my users are calling from GSM phones, and my system basically just plays some sound files back. The PBX is connected to an ISDN-30 connection. Are there any modules for playing MP3 files, so I can use them with commands like Play() and Background()? And will it have any effect o

[asterisk-users] Redirecting channels?

Hello I am going to have a setup like this: One Asterisk-box with two TE121-cards, and an ISDN-30 line. On the other hand, I also have another box with VoiceGuide and Dialogic. As a temporary migration-solution i would like to redirect some of the ISDN30 channels from the Asterisk to the Dialogic

[asterisk-users] Will this be sufficient for 20+ concurrent calls?

This is my first time setting up Asterisk in production and we are buying the Digium TE121-card for use with an ISDN-30 connection. We are considering buying a Fujitsu-Siemens Primergy TX200 S4 - http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html - for handling the

[Asterisk-Users] IBM eServers?

Hi, Has anyone used a Digium PRI card in an IBM eServer x346? I know that Digium's website lists the x345 as having problems, but I am restricted to buying only IBM eServers for this possible project. I would like to use the TE411P

RE: [Asterisk-Users] Re: Presence + Eyebeam + Asterisk 1.2

I'm sure look at rfc3265 (SUBSCRIBE/NOTIFY) which is not support by asterisk. How can you monitor the states of the buddies ? Harry --- Ben Buxton <[EMAIL PROTECTED]> a écrit : > > Are you sure? I've got it working with Eyebeam, > showing me just who is > availab

RE: [Asterisk-Users] Presence + Eyebeam + Asterisk 1.2

Don't waste your time asterisk does not support presence --- Mark van Kerkwyk <[EMAIL PROTECTED]> a écrit : > Hi, anyone managed to get a Presence Agent > configuration with Asterisk 1.2 > and X-Ten Eyebeam working. I believe this should be > paritally supported > now in 1.2 ? > > regards > >

[Asterisk-Users] Asterisk dial plan

Hello, When asterisk receive a registration with a private address is it possible to forward the sip request for this agent to a sip proxy ? Regards Harry ___ Appel audio GRATUIT

RE: RE : [Asterisk-Users] Asterisk doesn't start

Try to post your problem to asterisk-dev I guess they could solve or explain this problem better than asterisk'users . Harry --- Olivier Taylor <[EMAIL PROTECTED]> a écrit : > Yes, beta2 works perfectly, but 1.2 released version > gives me this error. > > Olivier >

RE: [Asterisk-Users] Asterisk doesn't start

Hello, You built asterisk on freebsd ? Harry --- Olivier Taylor <[EMAIL PROTECTED]> a écrit : > > Hello > > Whan starting astersik(1.2) (asterisk -vvc), I > get this message : > > [res_config_mysql.so] => (MySQL RealTime > Configuration Driver) &

RE: [Asterisk-Users] harry's project

Hello, I need SER for IM/presence and sip routing. Harry --- "Jonathan k. Creasy" <[EMAIL PROTECTED]> a écrit : > http://www.automated.it/guidetoasterisk.htm > > I don't think you even require SER in that case. > > That will be $100. > > -Jonat

[Asterisk-Users] harry's project

some people who help me to configure that . Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur

RE: RE : RE : [Asterisk-Users] What does it mean?

l, tu n'as qu'a payé pour ce que tu demandes. IL me semble même me souvenir avoir lu un développeur te faire la remarque "les utilisateurs de nos projets vous ne profitez que de notre travail !". Pour répondre à ton problème configure logger.conf . Harry --- Olivier Tayl

RE: RE : RE : [Asterisk-Users] What does it mean?

Merci pour ces précisions. Harry --- Olivier Taylor <[EMAIL PROTECTED]> a écrit : > SIBYLLIN, INE. adj. Qui appartient aux sibylles. Il > n'est guère usité au > sens propre que dans ces locutions : Les oracles, > les livres, les vers > sibyllins, Les oracles, les livre

RE: RE : [Asterisk-Users] What does it mean?

Je ne connais pas la signification de "sybillines". Harry --- Olivier Taylor <[EMAIL PROTECTED]> a écrit : > Tes réponses sont aussi sybillines que tes questions > :) > > Olivier > > -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[E

RE: [Asterisk-Users] What does it mean?

Hello, Read the Makefile in apps. Harry --- Olivier Taylor <[EMAIL PROTECTED]> a écrit : > Hello, > > I have compiled asterisk cvs under freebsd, no > problems. > > When starting asterisk, I get : > > [res_config_mysql.so] => (MySQL RealTime > Configur

[Asterisk-Users] GUI and Asterisk Realtime

Hello, Is there a GUI to manage sip users and voicemail with Asterisk Realtime . Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez

[Asterisk-Users] [Asterisk-Dev] hello

hello ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com __

[Asterisk-Users] hello

hello ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com __

Re: [Asterisk-Users] open letter

Could you tell me more please ? You understand than with host=dynamic in sip.conf asterisk use contact field in SIP HF Regards Harry --- "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> a écrit : > Sounds to me as what you want to do require 'a few' > code changes to &

Re: [Asterisk-Users] open letter

Could you tell me more please ? You understand than with host=dynamic in sip.conf asterisk use contact field in SIP HF Regards Harry --- "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> a écrit : > Sounds to me as what you want to do require 'a few' > code changes to &

Re: [Asterisk-Users] open letter (2)

. > > Have a look on the wiki (www.voip-info.org) for > Asterisk/SER consultants and > if you're lucky you might find someone who isn't > subscribed to the lists and > therefore may help you. I think Consultants have subscribed to these lists They could

RE: [Asterisk-Users] open letter (2)

my name is gaillac not giallac Harry --- Steve Totaro <[EMAIL PROTECTED]> a écrit : > New rule for email > Sender = harry giallac = deleted > > > > -Original Message- > > From: harry gaillac [mailto:[EMAIL PROTECTED] > > Sent: Wednesday, Novemb

Re: [Asterisk-Users] open letter (2)

You should read my mail so you would have an idea of my problem !!! Harry --- Patrick <[EMAIL PROTECTED]> a écrit : > On Wed, 2005-11-23 at 14:36 +0100, harry gaillac > wrote: > > What are your prices > > Don't have any since I have no idea what your > probl

Re: [Asterisk-Users] open letter (2)

What are your prices Harry --- harry gaillac <[EMAIL PROTECTED]> a écrit : > may be you > I agree > > --- Patrick <[EMAIL PROTECTED]> a écrit : > > > On Wed, 2005-11-23 at 10:34 +0100, harry gaillac > > wrote: > > > Advice me and I'll stop

Re: [Asterisk-Users] open letter (2)

may be you I agree --- Patrick <[EMAIL PROTECTED]> a écrit : > On Wed, 2005-11-23 at 10:34 +0100, harry gaillac > wrote: > > Advice me and I'll stop to mail my question. > > That almost sounds like a threat. Do you really > think you motivate > people to a

[Asterisk-Users] Re: [Users] open letter (2)

--- Klaus Darilion <[EMAIL PROTECTED]> a écrit : > Hi Harry! > > As this emails are on-topic you should cc: to the > list. > > harry gaillac wrote: > > In fact the problem is in contact sip header > field > > (private ip) > > agent send ReGISTER

[Asterisk-Users] Re: [Users] open letter (2)

ass through SER not directly to asterisk . I need nat support for sip agents behind nat. > Why do you use both? Asterisk can also do NAT > traversal. For how many > users is the setup? I think asterisk support 255 users > klaus > > harry gaillac wrote: > > Dear

[Asterisk-Users] open letter (2)

Dear users, This letter is addressed to the most experienced users for the ser openser and asterisk projects. Advice me and I'll stop to mail my question. How a session between two user agents behind nat could stay in the path ? Harry Kinds Regards |register || reg

[Asterisk-Users] RE: [Serusers] Re: open letter

Doug, You have ever post this mail. Harry > Others have tried to explain it too you, but I don't > think you fully > understand. Maybe it is a language issue. > > Your follow-up posts come across as demanding. When > I read your > posts, I feel like you are cr

[Asterisk-Users] RE: [Serusers] open letter

hello, Give me your price to enable my diagram ASAP --- harry gaillac <[EMAIL PROTECTED]> a écrit : > Hello open(ser) asterisk users > > Here is what i expect to do : > > Asterisk: registrar with public ip port=5050 > open(ser): outbound proxy with public ip port=506

[Asterisk-Users] Re: [Users] open letter

t be different in operation but I think its best > to find out howto > let ser do all the hardwork and let asterisk only > work when it needs to. They can work together ! thanks for help harry > harry gaillac wrote: > > >not exactly ! > > > >something like th

[Asterisk-Users] Re: [Users] open letter

is an outbound sip proxy which handle IM presence nat Harry > >>> One box > >>>--- > >

[Asterisk-Users] Re: [Users] open letter

asterisk. > > Paste config, in pastebin, and also a ngrep of the > call debug. > > Iqbal > > harry gaillac wrote: > > >Hello open(ser) asterisk users > > > >Here is what i expect to do : > > > >Asterisk: registrar with public ip port=5050 >

[Asterisk-Users] Re: [Serusers] open letter

> You lost me here. Was that a question or a > statement? > > I might not be able to help, since my SER usage is > totally diffent, > but let me see if I got this right: > - You want the SER to forward REGISTER messages to > the Asterisk. > - The user agents use private IP addresses. > - You w

[Asterisk-Users] open letter

box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans

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