Re: [asterisk-users] WiFi SIP phones
I must confirm Also we have modified the firmware to have more than 1 sip account and have pre-encoded sip proxies, works fine. Olivier Alberto Pastore a écrit : After trying painfully many many many flawed devices we eventually found a very good solution for our wi-fi network: Pirelli DP-L10. It's a dual GSM/Wi-Fi SIP triband (800/1800/1900). As a cellphone, no special note about it, but as a wifi phone, it has two main features which make it worth using: - excellent battery life - excellent roaming support between different APs even with no L2 fast roaming support or WPA key caching We now have 30 phones registered on an asterisk 1.2 server, connected through a 14 access point network (using wpa-psk). We've reached a DECT-level quality! Alberto Sagredo (M) ha scritto: You could try, N80, N95 devices. It cost arround 300 dollars and works fine with SIP , Wifi and GSM. I have been trying for several weeks with Truphone, Gizmo, Asterisk and other providers my N80 IE, and it works perfectly Regarsd 2007/5/23, Chris Bagnall [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since then. What models are currently out there people would recommend I look at? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 License
you can register twice after that you'll have to explain the reasons of changes to Digium Olivier Arun Kumar a crit: HI I bought 20 license from Digium and install in my server and b'coz of some problem I've to change my server is it possible that I can use those lice and register again in my new server ? Is it possible that I'll be able to use those lice in my old box also ? thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
well, the best I got is the tc300/arcor/twintel a gsm/wifi from pirelli - http://www.pirellibroadband.com/en_IT/browser/attachments/pdf/DPL10.pdf tried many wifi phones, that's the best we got. Long lifetime for the battery, good reception, roaming between Access points with the same network... good enough for us and cheap. livier Andrew Kohlsmith a crit: On Monday 04 June 2007 10:28 am, Paul Hayes wrote: Looking at the OP's requirements list in the first post, there is nothing currently on the market which will cover anything like all those features (and do it well!). I've got the WIP300 and 330 on my list, with the latter being the more likely candidate, as I can throw up custom apps once I figure out how it's done. :-) I like the idea of a wifi phone running Linux though, so both of these options will have to be investigated. but I'm yet to test any of these. The main problem is they have a habit of constantly losing connection with my access points. Even the F1000G and F3000 phones I have here don't do that. My F1000G phones *CONSTANTLY* lost connection with my WRT54, and it had nothing to do with signal strength, as the access point was less than 10 feet away from my desk, with nothing between to interfere. :-( I'm yet to be convinced that wifi in it's current state is any use for telephony at all. DECT works so much better, it just needs someone to make a fully functioning SIP DECT phone. The Siemens is good but they need to work on more SIP functions, although proper transfers should be possible soon. I am also slowly coming to this conclusion. Polycom recently acquired SpectraLink, who've got many years in the wireless phone business. They've got both Wifi and DECT offerings, but nothing with bluetooth, so the search continues. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Message d'un membre eBay sur l'objet #130110909566
? Membre eBay: zzolivier a crit: Hello, I'm wandering how can I make voicemail notification when i got a messages in asterisk mailboxes. For the moment i have e-mail notifications, but I readed that I can do also a sms notification to local sip accounts. Also I'm wandering if i can make something like callback from asterisk to sip account, and play voicemail check, when the user log in. Is there someone that use this feature? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: TR: TR:
?
KOUCH RACHID a crit:
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, December 20, 2006 6:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
"DG" == Douglas Garstang [EMAIL PROTECTED] writes:
DG So, in the event that the logic flows beyond
DG coo1_OnNet, we want to reset the caller id of say, 3254001 Doug,
DG to 3254000 Widgets Inc.
DG exten = 3254101,1,Dial(SIP/3254101,20,tr)
DG exten = 3254102,1,Dial(SIP/3254102,20,tr)
DG exten = 3254103,1,Dial(SIP/3254103,20,tr)
[coo1_CallStart]
include = coo1_OnNet
You want something which executes here, if coo1_OnNet didn't match?
exten = _.,1,Set(CALLERID(all)=Widgets Inc 3254001)
will do that.
If you then want to continue in priority 1 instead of 2, you just do
exten = _.,n,Goto(coo1_CallStart2,${EXTEN},1)
[coo1_CallStart2]
include = syst_OnNet
include = syst_OffNet
That won't do it. Processing will continue in the current extension priority. I need it to continue looking for an extension to match against. Once Asterisk has matched the dialled number against an extension in the dialplan, your stuck in an extension you can never get out and get Asterisk to go back to looking for extensions to match against.
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Re: [asterisk-users] Power requirements on the TDM-400 card
I use a a400p(tdm400p clone) on a soekris, 2 fxs and 2 fxo, some soldering needed but the Soekris power supply is enough. Only the fxs need power, Fxo doesn't. 18v 800 ma hope it could help Olivier Bob Chiodini a crit: Gustavo, Take a look at this thread http://lists.digium.com/pipermail/asterisk-users/2006-October/169627.html Presumably the supplemental 12v supply is for ringing voltage. I did not see anything on Digium's support pages about the card itself. Maybe a call to tech support may help. Bob... On Mon, 2006-12-11 at 13:09 +, Gustavo Felisberto wrote: I have a TDM-400 from digium with 2FXO+2FXS ports. Any idea on how much power will this drain from the 12 and 5 V connector when all ports are in use? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: good Linux references
Strange idea to switch from freebsd to another OS, Freebsd is very stable with asterisk, I must say, rock solid... What's the reason? Olivier ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk: SIP Gateway or Proxy
Not a proxy, of course, or proxy for a very small amount of users. Pbx and gateway is the best usage. David Thomas a écrit : Asterisk can actually act as a Gateway and a SIP Proxy. This is where a lot of confusion comes in. It can do pretty much any voip function you throw at it. Definitely search the archives if you still have questions. try site:lists.digium.com keyword in google to search the mail archives. David On 12/1/06, yusuf [EMAIL PROTECTED] wrote: Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Germany VOIP provider
we can provide a german did with unlimited inbound calls and give you a prepaid account for outbound, but no unlimited. Have a look at www.phonext.com Olivier Thameem Ansari a crit: I would like to get some details about voip providers in local germany. I am moving to germany and looking for some unlimited land+mobile minutes from provider. I also need a german DID with unlimited inbound and flat monthly rate. If anyone know anything, please reply. Thanks, Thameem ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Germany VOIP provider
also, sorry, I forgot to say that we are located in Belgium, very close to germany :) Olivier Thameem Ansari a crit: I would like to get some details about voip providers in local germany. I am moving to germany and looking for some unlimited land+mobile minutes from provider. I also need a german DID with unlimited inbound and flat monthly rate. If anyone know anything, please reply. Thanks, Thameem ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Negotiation
I must agree, we use 2 Ser in front of 4 asterisk sharing the same database cluster. Olivier Brian Capouch a écrit : Douglas Garstang wrote: Would you like me to dig up the posts from Keving Fleming stating that this is known not to work Brian? As I recall those posts have to do with the way your particular setup required ARA to work with a failover/redundant cluster system you were building. Beyond that I'm not really interested in getting into a pissing contest. I have ONE SQL table called extensions_table on ONE SQL server, but have maybe 20 SIP phones using that same database, placing calls from 10-12 separate Asterisk instances. I was calling into question your presenting a well-known fact that appears to be incorrect. If Kevin sees this and wants to chime in to support your statement and tell me that my experience is somehow an illusion, he's certainly welcome to do so. I have experienced the taste of crow, and eat it when needed. You? Can certain situations be construed where ARA will not do exactly what the administrator wants? Apparently, from reading some of your posts, true. Can multiple Asterisk servers be set up to use a single database instance to store common configuration information? Certainly true, from my and many other people's experiences. The thrust of my post was to refute the fact, and to suggest you perhaps adopt a little less inflammatory rhetoric when you post to this list. B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: where is the error?]
---BeginMessage---
Identifier 0, identifier_type 2 not found in identifier list given when
sql query is :
SELECT\ left(Customer.balance,instr(Customer.balance,'.')-1)\ FROM\
Customer\ Inner\ Join\ subscriber\ ON\ subscriber.customer_id\ =\
Customer.id\ WHERE\ subscriber.username\ =\ ${CALLERIDNAME}
query works on Mysql...
same error when I use Truncate...
Any ideas are welcome :)
Olivier
begin:vcard
fn:Olivier Taylor
n:Taylor;Olivier
email;internet:[EMAIL PROTECTED]
tel;work:+3227470340
tel;fax:+3227470397
note;quoted-printable:MailScanner is like deodorant...=0D=0A=
You hope everybody uses it, and=0D=0A=
you notice quickly if they don't
version:2.1
end:vcard
---End Message---
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Re: [asterisk-users] [Fwd: where is the error?]
thx mate,
but also ' must be escaped ' has to become \'
I got it, thanks for the help, u got me to the right way :)
Olivier
trixter aka Bret McDanel a crit:
On Mon, 2006-07-17 at 15:17 +0200, olivier.taylor wrote:
email message attachment (where is the error?)
SELECT\ left(Customer.balance,instr(Customer.balance,'.')-1)\ FROM\
Customer\ Inner\ Join\ subscriber\ ON\ subscriber.customer_id\ =\
Customer.id\ WHERE\ subscriber.username\ =\ ${CALLERIDNAME}
asterisk translates , to | then processes it. try \, instead see if
that cures your errors.
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Re: [asterisk-users] intel vs amd motherboards
ooops, sorry, you right, forgot to mention it... It was to be compared with AMD 64. Olivier C F a écrit : Olivier can you please do a cat /proc/cpuinfo and post it here? I think you have a 64 bit cpu. On 7/9/06, olivier.taylor [EMAIL PROTECTED] wrote: Fyi, Double Intel Xeon 3Ghz performance below g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 410 2914 ulaw - 2 - 1 2 2 1 410 2914 alaw - 2 1 - 2 2 1 410 2914 g726 - 2 2 2 - 2 1 410 2914 adpcm - 2 2 2 2 - 1 410 2914 slin - 1 1 1 1 1 - 3 9 2813 lpc10 - 3 3 3 3 3 2 -11 3015 g729 - 3 3 3 3 3 2 5 - 3015 speex - 3 3 3 3 3 2 511 -15 ilbc - 3 3 3 3 3 2 511 30 - Olivier Tzafrir Cohen a écrit : On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote: Tzafrir, are you trying to tell me that I can realy do double on the intel becuase the second CPU will do it? In the ideal case you'll get double performance with two CPUs. In theory. A case of many concurrent calls is basically something that can be easily parallelized. So in theory nothing stops you from getting something closer to double performance. I don't know how close reality is to that nice theory. I only remarked that 'show translations' totally ignores the second CPU. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intel vs amd motherboards
Fyi, Double Intel Xeon 3Ghz performance below g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 4 10 29 14 ulaw - 2 - 1 2 2 1 4 10 29 14 alaw - 2 1 - 2 2 1 4 10 29 14 g726 - 2 2 2 - 2 1 4 10 29 14 adpcm - 2 2 2 2 - 1 4 10 29 14 slin - 1 1 1 1 1 - 3 9 28 13 lpc10 - 3 3 3 3 3 2 - 11 30 15 g729 - 3 3 3 3 3 2 5 - 30 15 speex - 3 3 3 3 3 2 5 11 - 15 ilbc - 3 3 3 3 3 2 5 11 30 - Olivier Tzafrir Cohen a crit: On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote: Tzafrir, are you trying to tell me that I can realy do double on the intel becuase the second CPU will do it? In the ideal case you'll get double performance with two CPUs. In theory. A case of many concurrent calls is basically something that can be easily parallelized. So in theory nothing stops you from getting something closer to double performance. I don't know how close reality is to that nice theory. I only remarked that 'show translations' totally ignores the second CPU. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Most stable Asterisk version
Same for me here, freebsd ports and same usage. Running from months on a Dell 1850(biXeon 4Gb ram) with no problems. Olivier Jean-Michel Hiver a écrit : shadowym a écrit : Hi there, I am getting ready to set up a production Asterisk system. It needs to be stable. Upgrading, patching, rebooting, troubleshooting etc. are pretty much NOT an option once this thing is deployed. Like any phone system, it is expected to just work. Try FreeBSD's Asterisk port. It has been working rock-solid for me so far. It's been a few weeks now with no issues (fingers crossed)... But I admit that it does _JUST_ softswitching (i.e. call routing, load balancing and database CDR collection) and hence has the smallest possible feature set (search google: voip-info asterisk slimming). Another option is to buy Digium's commercial edition of Asterisk, which is supposed to be just what you describe. Best Regards, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Ok, on peut parler français alors ;) Olivier Jean-Michel Hiver a écrit : Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? Quel bordel, sacrebleu! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Multi Call Generation
sipsak is ok for that Olivier Abdul Lateef a écrit : Hi all, Is there any such as tools for multi call generation to test, how much call can be done via Asterisk? _ Best Regards, --- Abdul Lateef Nepal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to Hire: PHP Programmer for PhoneCALL
here is an example of the code I procuced, ?php echo Asterisk Users Mailing List - Non-Commercial Discussion ; ? It should work if php-cli is on the path, just try. Olivier Dustin Wildes a écrit : Hello all! It's come time where I need to add another programmer to our team. You should have at least 3 years of work experience with PHP/MySQL. Please send me your resume and a few code samples if you can. If you can only work part-time or full-time, please include that in your response. Along with your salary requirements. You'll be working with PhoneCALL, so be sure to look over the code first before applying. http://www.vecsector.com/phonecall Thanks everyone! --- Dustin Wildes President VecSector, LLC 1.912.422.7082 x101 email: [EMAIL PROTECTED] web: http://www.vecsector.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP codec preference order ineffective
just buy G729 codec licences from Digium, 10$ per channel, Asterisk is just G729 pass-thru. G729 is not a free codec (: Olivier Patai Tams a crit: Hi, I set a preference order of the codecs to my sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls of not registered phones disallow = all allow = g729 allow = g723 allow = alaw allow = ulaw Connected a 'Sipura SPA' sip phone to asterisk with g729 as its preferred codec. Problem: asterisk cannot make the connection: set_format: Unable to find a codec translation path from alaw to g729 Connection only can be made if the Sipura's preferred codec is alaw/ulaw. Any help appreciated Tamas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom replacement handset
heya Cory, did u receive and test the siemens handsets? Olivier Cory Andrews a crit: Ryan Shoot me an email off list, I can help you out with a replacement handset. Thanks Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Ryan Stark Sent: Tuesday, May 30, 2006 8:29 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom replacement handset Does anyone know where I can get replacement handsets for the Polycom SoundPoint IP phones? Or does anyone have any they want to sell? From the looks of it you have to buy a whole new phone to get a new handset. My vendor, TriaTechCOA, told me I had to buy a whole new phone to get a handset, which is pretty ridiculous. Maybe there is a more sane vendor I should be buying from? Thanks, -Ryan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Centos cause Asterisk crash
use freebsd, not just a kernel as linux, just a real complete os :) Sean Kennedy a écrit : chan, Run each script seperately to determine which one causes the crash. From there, check your logs to see any error messages. There should be something. My hunch is that prelink will cause the crash. chan (Alpha Trilogies Networks) wrote: Hi, Can some one who experience that does those file necessary for the CentOS and Asterisk installation /etc/cron.daily/00-makewhatis.cron /etc/cron.daily/slocate.cron /etc/cron.daily/prelink /etc/cron.daily/rpm /etc/cron.weekly/00-makewhatis.cron I experience that those file cause my Asterisk Server crash. Can I just disable them and run the Asterisk stable? Any reply will be appreciated. Thank you in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 hardware for asterisk
seems to be very good cards, but also, very expensive, isn't it? Olivier Armin Schindler a écrit : On Tue, 30 May 2006, Tristan wrote: I'm interested too to know about a quad E1 card... I need to connect it to 2 differents ISDN providers in Europe and to establish a third connection with a Matra PBX. The server ( IBM XSeries 346 ) has to serve about 60-70 simultaneous calls ( IVR and max 30 conferences... ) I will also need ( but later, as I think I'll have to write it ) support for videoconferencing over ISDN using different protocols like h320 or h324m... What would you recommend ? Digium TE411P, Sangoma A104D, Eicon Diva Cards ? I cannot tell anything about the Digium or Sangoma cards, but the Eicon Diva Server Cards are active cards, which means they do the ISDN protocol stuff including digital-signal-processing (if needed) on board without using the hosts CPU. So in a setup as you described above, I recommend to use the Eicon cards. Armin Armin Schindler a écrit : On Tue, 30 May 2006, olivier.taylor wrote: Hi all, I need your lights :) There are many hardware provider for E1 cards on the market, what's your exeperience with E1 and what's the preferred provider for Asterisk out of Digium? I prefer Eicon Diva Server cards, they have good features and are very reliable. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 hardware for asterisk
Hi all, I need your lights :) There are many hardware provider for E1 cards on the market, what's your exeperience with E1 and what's the preferred provider for Asterisk out of Digium? Olivier ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..
Clearwire(wimax) blocks voip in Belgium and did in the Usa. just google 'voip clearwire', that's interesting... Olivier Joao Pereira a écrit : Hello Just 2 ideas: How cares about GSM WiFi handovers? I just want to make free VoIP calls. About the ISPs blocking VoIP: I believe they will not block VoIP because a lot of theire services are VoIP based, like the webcasts, the TV shows over the Internet, and all the multimedia stuff they want us to buy. In Portugal I already did 3G VoIP calls from TMN and Vodafone. I would really like to try this phone :) Regards Joao Pereira Steve Kennedy wrote: On Tue, May 23, 2006 at 02:50:33AM +0800, Sam Tam wrote: Well it is incorrect to say that. In places like USA or London, a lot of areas are covered by local wifi providers, some are free, some aren't. You then can use them to drop some of your local or international calls cheaply by using wifi. But the point is without operator cooperation, there's no seamless handover between GSM and WiFi, and the operators don't want to lose the revenue on the voice, so they are unlikely to support it. BT have an arrangement with Vodafone for their Fusion service (using an in-premise Bluetooth basestation and a phone with GSM/Bluetooth), but they're big enough to force an operator's hand. For general GSM/WiFi UMA, it's unlikely the (UK) operators will allow other providers access to their networks, as it reduces their revenues. They're already p*ssed off enough that they're being forced to reduce roaming charges (currently on voice - but the EU is likely to look at data charges which can be extremely costly). They are desperate to keep revenues. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soekris hadware
more kindly : http://www.astlinux.org/ Olivier Christopher Snell a crit: Google and voip-info.org will have answers to all of your questions. On 5/17/06, Jonathan Gonzalez [EMAIL PROTECTED] wrote: Hi group, i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: [...] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi VoIP Handsets..
just have to say WOW I got a new voip wifi handset. Not yet on the market, but the constructor promised me international versions if I can have a 50+ order. Well, the constructor is a very well known handset provider in the Pstn/Isdn world and Skype world :( ... It's a german one... Smell good, go on, it's S.s, they will put the handset on the market around september. Honestly, I gave a few calls with it, dunno about battery life and so on, quality is very very good, but what I can say is that the specs are WONDERFULL. Linux based :) Olivier ps: public price will be around 199 taxes includes. If I have 50+ orders, I promise to do my best to have the best price for all, this is NOT a commercial offer, just an offer for asterisk users (also ser users). Kind of open source hardware offer ;) If any of you can host the specs, I will send a pdf The VoIP Connection a crit: According to all of my sources, the UIP1868 has been discontinued. Kind of a shame, it was a neat product. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Colin MacMillan [mailto:[EMAIL PROTECTED]] Sent: Wednesday, May 17, 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. I know for a fact that the Aastra 480i-CT is not available in the UK/Europe at the moment. There is no program in place to get in over into Europe however I think it could happen in the next 4+ months. Does anyone know if the UNIDEN UIP1868 is available in the UK? If so how do I get my hands on one ...? On 5/17/06, Andrew Latham [EMAIL PROTECTED] wrote: Cory The 480i-CT does not state DECT to my knowlege as the EU DECT standard uses reseved frequency space in the US.I have heard rumblings about a US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity. Andrew On 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi VoIP Handsets..
nope, that's this one : http://www.slashphone.com/86/2493.html Olivier Cory Andrews a crit: Could it perhaps be this iPod looking jammie right here http://www.engadget.com/2006/03/10/fritz-mini-wifi-phone-mp3-player-and-more/ Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of olivier.taylor Sent: Wednesday, May 17, 2006 12:47 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. just have to say WOW I got a new voip wifi handset. Not yet on the market, but the constructor promised me international versions if I can have a 50+ order. Well, the constructor is a very well known handset provider in the Pstn/Isdn world and Skype world :( ... It's a german one... Smell good, go on, it's S.s, they will put the handset on the market around september. Honestly, I gave a few calls with it, dunno about battery life and so on, quality is very very good, but what I can say is that the specs are WONDERFULL. Linux based :) Olivier ps: public price will be around 199 taxes includes. If I have 50+ orders, I promise to do my best to have the best price for all, this is NOT a commercial offer, just an offer for asterisk users (also ser users). Kind of open source hardware offer ;) If any of you can host the specs, I will send a pdf The VoIP Connection a crit: According to all of my sources, the UIP1868 has been discontinued. Kind of a shame, it was a neat product. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Colin MacMillan [mailto:[EMAIL PROTECTED]] Sent: Wednesday, May 17, 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. I know for a fact that the Aastra 480i-CT is not available in the UK/Europe at the moment. There is no program in place to get in over into Europe however I think it could happen in the next 4+ months. Does anyone know if the UNIDEN UIP1868 is available in the UK? If so how do I get my hands on one ...? On 5/17/06, Andrew Latham [EMAIL PROTECTED] wrote: Cory The 480i-CT does not state DECT to my knowlege as the EU DECT standard uses reseved frequency space in the US.I have heard rumblings about a US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity. Andrew On 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update
Re: [Asterisk-Users] Audio problems 50% of the time.
if your connection is also used for web, email, and the worst, p2p, you better to have qos on your router. just be aware that g711 will use 80Kb up and down... gsm and g729 wil use 30/40Kb then : disallow all allow = gsm allow = g729 Olivier kurt x a écrit : I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via DSL through a Linksys router. You can see the my sip.conf for the phone blow. The problem is each time the phone rings I can hear/be heard 50% of the time. Any suggestion on what to look for. I do have my reg time set for 180 seconds on the cisco ATA186. [72459] type=friend username=XX secret=X host=dynamic context=voice-mail dtmfmode=rfc2833 ;canreivet=yes nat=yes qualify=yes Kurt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi VoIP Handsets..
siemens wifi voip phone specs are available at : http://www.gdlicanet.net.mx/voip/sl75.zip Thanks to Otto Krumm Hernndez [EMAIL PROTECTED] Olivier olivier.taylor a crit: nope, that's this one : http://www.slashphone.com/86/2493.html Olivier Cory Andrews a crit: Could it perhaps be this iPod looking jammie right here http://www.engadget.com/2006/03/10/fritz-mini-wifi-phone-mp3-player-and-more/ Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of olivier.taylor Sent: Wednesday, May 17, 2006 12:47 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. just have to say WOW I got a new voip wifi handset. Not yet on the market, but the constructor promised me international versions if I can have a 50+ order. Well, the constructor is a very well known handset provider in the Pstn/Isdn world and Skype world :( ... It's a german one... Smell good, go on, it's S.s, they will put the handset on the market around september. Honestly, I gave a few calls with it, dunno about battery life and so on, quality is very very good, but what I can say is that the specs are WONDERFULL. Linux based :) Olivier ps: public price will be around 199 taxes includes. If I have 50+ orders, I promise to do my best to have the best price for all, this is NOT a commercial offer, just an offer for asterisk users (also ser users). Kind of open source hardware offer ;) If any of you can host the specs, I will send a pdf The VoIP Connection a crit: According to all of my sources, the UIP1868 has been discontinued. Kind of a shame, it was a neat product. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Colin MacMillan [mailto:[EMAIL PROTECTED]] Sent: Wednesday, May 17, 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. I know for a fact that the Aastra 480i-CT is not available in the UK/Europe at the moment. There is no program in place to get in over into Europe however I think it could happen in the next 4+ months. Does anyone know if the UNIDEN UIP1868 is available in the UK? If so how do I get my hands on one ...? On 5/17/06, Andrew Latham [EMAIL PROTECTED] wrote: Cory The 480i-CT does not state DECT to my knowlege as the EU DECT standard uses reseved frequency space in the US.I have heard rumblings about a US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity. Andrew On 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew
Re: [Asterisk-Users] WiFi VoIP Handsets..
sorry my scanner was upside down ;) Olivier JP Carballo a écrit : Lacy Moore - Aspendora wrote: Had to turn my monitor upside down to read them :-) -- Lacy Moore Aspendora, Inc. You must have one of those rotating monitors huh? ;) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Cheapest provider for Philippine route
Can be as low as 15€cents from us on fix and 20€cents for mobiles We don't have dids yet for Philipine -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Johnathan Corgan Envoyé : mardi 28 février 2006 18:07 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Cheapest provider for Philippine route Sam Tam wrote: Do anyone know who can provide some cheap PH routes/.’ I've been looking myself. Cheapest DIDs in Metro Manila I've seen are $27.50/month; cheapest termination to same (non-mobile) from US I've seen is $0.23/minute. Expensive chismis :-) -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Beer meeting at Fosdem
Hi Olle, Will u be there for the speech of Jan Janak? If yes, you will find a guy, 1m83, with a bear and a red suit, it's me. You also can call me on my mobile to fix the voip beer (0032495283361). We will try to have Jan and other guys Olivier ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] How to Get SIP Header : To Field ?
Title: Message
SIPGetHeader(var=headername)
Olivier
-Message d'origine-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
Jean-Marc SalsaEnvoyé: lundi 13 février 2006
11:40À:
asterisk-users@lists.digium.comObjet: [Asterisk-Users] How to
Get SIP Header : To Field ?
Hi,
I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch.
When forwarding a call to Voicemail, here is somehow what the softswitch
sends to Asterisk :In INVITE : Vm Phone Number ( to route the call )In
To : Person who has been called !In From : Person who was calling !
Of course, I need to send the call into the "Called User" Mailbox (Thus To
SIP header) !
So Basically, filed in INVITE is "EXTEN", From field can be obtained from
the function ${SIPCHANINFO(from)}But how to get the "To" field ?
I have tried to add some code line into the chan_sip.c ...It works
partially ... meaning that, I can add this "to" in SIPCHANINFO
funciton,but the result is null.
Here is what I have added in chan_sip.c :in structure sip_pvt ( "to"
field same as "from" )in sipchaninfo_function added "to" Line same as
"from"function_sipchaninfo_read added "to" line same as "from"
So I believe that I have enabled somehow Asterisk to read the value to from
the channel ...But how to get the value and put it inside the channel ???
I think this would be my real question !
Thanks in advance for anybody who could help me ...
Yours,
JM
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[Asterisk-Users] Oh323, opengk and asterisk
Hi all, Well, i need h323 and asterisk working together. I have asterisk with oh323 working I have opengk installed on the same server (working too). I have a h323 handset (swissvoice ip10s) The swissvoice register with opengk (don't ask me how).. I need opengk register with asterisk to have the opportunity to relay the calls to a sip pstn gateway. I googled a lot but didn't find any solution or samples. Just to avoid wasted time, does any of you have an opengk config file and an asterifk config file making possible to have a working solution? Thanks for all, Olivier ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] lists problem, Gmail????????
Pfff, What for an answer :( I use gmail and have no problems. Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Martin Joseph Envoyé : lundi 13 février 2006 20:36 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] lists problem, Gmail On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote: C F ha scritto: Am I the only one having trouble with this list? Since the begining of the week I have not been receiving mail from the list like I used to, is this a gmail problem? or is it subscription problem? or is something wrong with the list? anybody else using gmail having any problems? Yes, I'm also getting some lag sometimes, one or two days without receiving mails get a real mail server and it works great! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Asterisk 1.2.x + oh323 on Debian Sarge.
Welcome to the club, same here with freebsd :( -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jarek Jarzebowski Envoyé : vendredi 10 février 2006 23:01 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Asterisk 1.2.x + oh323 on Debian Sarge. Hello, is anybody there who successfully compiled Asterisk 1.2.4 with oh323 on Debian Sarge? I tried severel versions of oh323 and pwlib and there is no results... only errors. -- Jarek ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Dialing part of the extension
Exten = 9XX,1,Dial(Zap/4/${EXTEN:1},40,r)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Cosmin
Prund
Envoyé : samedi 11 février 2006 12:19
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Dialing part of the extension
I know this one must be easy but I'm an newbye so please help. In my
extensions.conf I want to have a line like:
Exten = 9XX,1,Dial(Zap/4/${SOMETHING},40,r)
Ie: I want to dial all the XXX-es, but not the 9;
How do I do that? What do I write in place of ${SOMETHING}? Navigating
the wiki didn't provide any usefull advice...
Thanks.
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[Asterisk-Users] asterisk 1.2.4,addons 1.2.1, ooh323 and freebsd
Title: Message is there a way to compile ooh323 on freebsd, I have tried many solutions, nothing works :( Any good idea is welcome. Kind regards, Olivier ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
