Re: [Asterisk-Users] Quadbri bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?
Hi, I have problem with Quadbri and bristuffed Asterisk - I guess this is only configuration trick. I'd like Asterisk to respond only to 1 number on BRI interface and do nothing on other. Right now, even if I leave out that number in incoming context, Asterisk takes out and rejects call as number is non existant. I'd like that Asterisk wouldn't respond, so other ISDN phone I think a ugly trick is to do: exten = MSN_TO_BE_FREE,1,Wait(100) Matteo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_perl compile problem
Hi, gcc -c perlxsi.c -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing - D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm - I/usr/lib/perl5/5.8.0/i386-linux-thread-multi/CORE -o perlxsi.o gcc: perlxsi.c: No such file or directory gcc: no input files make: *** [perlxsi.o] Error 1 How can I get the file Perlxsi.c???/ nowhere, since is created automatically from res_perl Makefile. honestly I didn't have to modify * makefile. here I just did make clean ; make ; make install on res_perl dir, and all went ok. matteo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't Modprobe ztcfg
Hi, Il giorno ven, 15-04-2005 alle 18:33 -0400, Ian Pattison ha scritto: If I understand your question correctly ztcgf is not a module, it's merely a rudimentary diagnostic utility. Run ztcfg -vv to get info on your zaptel hardware. Not exactly. ztcfg is the tool that applies the zaptel.conf configuration to the kernel modules (ie sets up every channel to what you decide in zaptel.conf), and must be executed after module loading. The diagnostic utility is called zttool. Matteo. -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home ISDN BRI
Hi, Il giorno sab, 16-04-2005 alle 13:30 +0200, Robson Ribeiro ha scritto: Frtiz is a nightmare although it is cheap and I have seen it working. I have been trying to install it for some days without success but one thing is for sure: you have to use the right Kernel (they are available for 2.4.20 and 2.6something). You're wrong. I have it running on several kernels, including 2.4.26-29. To make it work you must change a define in the src code, dunno remember the right line now, but is easy to find out reading the compilation errors. Matteo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
Hi Il giorno ven, 08-04-2005 alle 10:24 +1200, Matt Riddell ha scritto: Matteo Brancaleoni wrote: I hate to say that, but the problem is that Digium doesn't do this. Ahh I beg to differ. I resell both Digium and Sangoma gear and provide full installation support for both. after a lot of words, the real story is that we too :) Matteo. -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Script Perl Autodialer
Hi, The problem is that when opening the zap channel, originate thinks that the call has been answered and send the call to the beginning of the context out. And what I really want is to make this but when the destiny person answered and not when the zap channel opens. as already in the docs, on analog zap interfaces you simply cannot do that, since on analog there's no way (apart dsp) to guess when the called party has answered So what can I do to solve it ou? go digital (isdn bri/pri, voip, whatever) Matteo -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on WRT54GS
Hi, Also, the issue i have with incoming calls is odd. I seem to get a timeout when dialing my SPA2000. Atleast that is the message. my incomeing context is [incoming] exten = s,1,Wait(10) exten = s,10,Dial(SIP/3518,20,tr) /me wonders why s,10 you should use next priority after 1 , ie 2 so: [incoming] exten = s,1,Wait(10) exten = s,2,Dial(SIP/3518,20,tr) Matteo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What happens if quadbri or octobri loses power - do they have power failure feature ?
Hi, I wonder if I use quadbri or octobri cards to insert Asterisk between ISDN PBX and ISDN line - if power of Asterisks fails - will those card connect PBX directly to ISDN line ? No, you need a isdn failover switch If not are there any other simple switching devices, that would do this (in power fail it will connect ISDN PBX to ISDN lines directly) ? Yes, klaus (author of bristuff) has/will have a solution for that. Hardware isdn failover switch. I don't know if I can reveal some details on this magic, so please contact him for further details Matteo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WM Wink timings for Nortel
Hi, Il giorno ven, 18-02-2005 alle 12:44 -0600, Eric Wieling ha scritto: The Digium Tx00P and TE*xxxP support EM Wink EM is analogue, not digital... digium cards support it over digital, like they supports fxs/fxo to a channel bank . same from EM The interface described here is analogue, afaik. Matteo. -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Accountcode and SIP Peers Part 2
hi, Il giorno gio, 17-02-2005 alle 22:19 +0100, Olle E. Johansson ha scritto: If you're anonymous, we propably can't match to a user/peer and set the accountcode from the configuration... Or? mmmh... but if the user authenticate itself, we can have an accountcode. I mean anonymous != auth and accountcode should depend on auth. or not? Matteo. -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID and anonymous SIP calls
Hi, Il giorno sab, 05-02-2005 alle 11:11 +0100, Marcello Lupo ha scritto: Hi to all, can you suggest to me the best way to avoid problems in the CDRs for anonymous sip calls? I have some peoples that set Send Anonymous : Yes in their Grandstream phones and i don't receive the username as phone number that i use to make billing. It is empty. The only place where there is the phone number is in the peer name where it write the name of the peer that in this case is the phone number. eh, you should not trust cid for billing, but accountcode. set it in your sip.conf file, so since the user authenticate the account code is set and no one can change it, beside you. matteo -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Not hanging up DS0 when number called is busy.
hi,
Il giorno mar, 01-02-2005 alle 13:30 -0600, James Sizemore ha scritto:
extensions.conf:
[trunk]
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = h,1,Hangup
try
extensions.conf:
[trunk]
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = _X.,2,Hangup
Matteo.
--
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VPN server for * and woad warriors using windows?
Hi, Il giorno dom, 23-01-2005 alle 10:33 +0100, Remco Barende ha scritto: What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. (Ofcourse call quality is important) best if ofcourse some ipsec-based solutions, but that leads to installing a client on winblow machines. You can use pptp, ok is not secure as ipsec but is built in in winblow 98,2k,xp... so on the client you must only create a new VPN connection (under connections manager) and you're done. On the linux side, go to http://poptop.sourceforge.net/dox/ to grab the server. I think that this is the easiest solutions for a decent encryption ad ease of use, when using m$ clients. (hoping you don't need to protect millions $$$ value data : ) of course ipsec is better, but needs more work to set it up, on client and on server side. just my 2 cents, Matteo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echocancellation in modem.conf
Hi, Il giorno mer, 19-01-2005 alle 18:45 +, Edin Kozo ha scritto: I have a ISDN BRI card with hisax module (w6692) and there is a lot of echo when I make calls to outside. Between the sip softphones the echo doesn't exist, but when I call to outside through the ISDN the echo exist. yup. modem support should be removed from asterisk. is useful only as a latency generator. Is there any sense to put echocancellation in modem.conf ? no. use an hfc-s pci card with bristuff drivers. Matteo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is an unregistered phone busy?
see ${DIALSTATUS} built in var.
can be chanunavail, or busy or what asterisk sets it to.
use it do do your switching.
Matteo.
Il giorno mer, 19-01-2005 alle 00:25 +0100, Rob Scott ha scritto:
Asterisk seems to regard an unregistered phone to be busy.
Is that correct? Is not an unregistered phone unavailable?
It is odd to me that if someone dials an unregistered extension, then
the dialplan jumps to busy and voicemail kicks in saying that the person
is on the phone, when clearly they can't be if the phone hasn't
registered.
Any way around this?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom hint for ZAP channels?
Hi Il giorno ven, 21-01-2005 alle 08:54 -0600, Justin Carlson ha scritto: is the hint 99,hint,ZAP/1 that works only for sip channels. if you want hint working also for zap, you should check very latest bristuff at junghanns.net website. Afaik he has added (among support for bri cards) extension states also for zap channels. Matteo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] not sharing IRQ's
Hi, Il giorno mer, 12-01-2005 alle 00:38 +0200, Shoval Tomer ha scritto: Only if you don't have Digium hardware installed. yes And only for MeetMe, I think. Correct me if I'm wrong on this, though... really, it works for zaptel timing, that's needed only by meetme and iax2 trunking. But it can work with anything that needs zaptel timings. Matteo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc problem
hi i have HFC supported ( Planet TA ) card installed on Redhat 9 and i have installed bristuff and i can load zaptel and load zaphfc module in TE mode . and unable to load ztdummy module properly you don't need ztdummy if you have a zaptel card installed here is my zaptel.conf loadzone=nz defaultzone=nz span=1,1,3,ccs,ami bchan=1-2 dchan=3 # here is my zapata.conf switchtype = euroisdn signalling = bri_net_ptmp ;this is for a peer to multipeer network ;pridialplan=local immediate=yes group = 1 context=default channel = 1-2 echocancel=yes put echocancel *before* channel definition, or you won't have echo can on channels 1-2 I think that the problem could be related on the fact that you try to load ztdummy. Just do that: unload all modules modprobe zaphfc ztcfg -vvv start asterisk no ztdummy here:) try and let us know. Matteo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call on hold disconnects...
hi, I receive a call at the extension. Press the hold button. Music on hold starts. When I place the handset back on the cradle, the call gets hung up/disconnected. The Phone is A GrandStream Budge Tone 100. this seems a phone problem. 2 solutions: * don't put the handset back on the cradle * get another type of phone Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI Card not recognized
Hi, Il giorno gio, 16-12-2004 alle 21:59 +0400, Muhammad Talha ha scritto: Dear all i am using Fedora Core 2 . i have Planet BRI TA with HFC chipset ( hisax ) i can easyly connect to internet using BRI but this card is still not recognized by asterisk i am using i4l driver . don't use i4l. is only a latency generator (ie you'll experience bad echo issues) some people suggest i should try bristuff from junghanns.net yes, go with that. We've bristuff running smoothly here with hfc based cards. Matteo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting Asterisk to GSM
Hi, Il giorno gio, 16-12-2004 alle 21:47 +, Jean-Michel Hiver ha scritto: I was wondering if there was any device I could use to connect * to GSM networks. I don't need much capacity, maybe 2-4 GSM channels. As usual, cheap is better :-) sure, mainly you can use gsm boxes with pstn to gsm interfaces. for example: for 1 gsm chan, you can use a box with an fxs interface on it, and can be connected to * via a single x100p (one fxo interface) Or for multi channels, you can go with a bri-gsm box, and interface it to * via a bri card (junghanns.net drivers) or even pri, with 16 or more channels (connected to the * with a pri card, ie te110p) or even sip... no card on the * box, but connected via a sip voip link. www.2n.cz has some of these products, but there're tons of them out there. prices? dunno exactly, the only that I'm aware of is that a bri - gsm (2 gsm chans) is something like 800 ¤ Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gap in priorities - what happens
Hi, Il giorno dom, 12-12-2004 alle 14:38 +0200, Warren Burstein ha scritto: When I first saw the priority numbers in extensions.conf, I thought BASIC, if a number is missing, * will fall thru to the next number. I learned that this is not so, if you have nothing between 1 and 3, you don't ever get to 3. that's true. But I'm wondering what does happen? Hangup and wait for next offhook? Undefined? Timeout is called. Ie if exists the exten t, after the timeout (default 5 secs, if I remember correctly) will be executed, otherwise hangup Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Card Error
Hi, Il giorno dom, 12-12-2004 alle 00:36 -0800, Charles S. Antrim ha scritto: I have success installing and compiling, but if I reboot I have to modprobe again to get he drivers loaded for the module I am using. I am using rhes31 and a tdm card with one fxo and one fxs. perhaps you have to build a script that loads modules on boot? see in zaptel src dir, there's a zaptel.sysconfig zaptel.init demo examples. Not installed by default. Matteo. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] last stable version
hi, Il giorno mer, 01-12-2004 alle 15:37 -0300, Listas ha scritto: Hi I would like to know which is the last stable version of asterisk and how to get it from the CVS, I mean rather than doing cvs checkout -r -v1-0_stable asterisk go on asterisk ftp site and look for 1.0.2 tgzs, or co via cvs with cvs checkout -r -v1-0 asterisk Greets, matteo P.S. search the ML. this is a echoing question ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple asterisk process
Hi Il giorno mer, 24-11-2004 alle 19:48 +0100, Ming-Wei Shih ha scritto: Hong Kim wrote: I'm running * on Redhat9 with E100P and ISDN PRI. When I executed asterisk, I could see about 25 asterisk processes. Did someone experienced this? Regards, Hong snip I only see one :) $ ps -ef |grep asterisk root 12536 1 0 Nov22 ?00:00:00 /opt/asterisk/sbin/asterisk xming 7486 7481 0 19:44 pts/000:00:00 grep asterisk $ let me guess, you are using 2.4.x kernel? In 2.4 kernel, all threads are listed ad processes not only kernel, but depends also on ps version. On rh9 I see only 1 proc, of FC1 (also kern 2.4, but newer ps) I see all the threads. Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Patch - estimated hold time announcements
Hi, Il giorno mar, 23-11-2004 alle 16:38 -0500, Jay Brussels ha scritto: I started out with the development branch then switched to the stable (as the entire company now runs on Asterisk). The stable branch (including 1.02) does not have the queue annoucements. I'm sorry to contradict you, but stable 1.0.2 has queue announcements. Looking into app_queue.c from asterisk-1.0.2.tar.gz we can confirm that. Eg. , on top of file: * These features added by David C. Troy [EMAIL PROTECTED]: *- Per-queue holdtime calculation *- Estimated holdtime announcement *- Position announcement *- Abandoned/completed call counters *- Failout timer passed as optional app parameter *- Optional monitoring of calls, started when call is answered then also into queues.conf.sample Matteo. PS we are running queues with announcements with stable version without any issue... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX issue at nufone
Hi, Y si te molesta no hubieras respondido OK ? because this isn't a nufone support ML. The next time, post your configs, not your complains about nufone. Without that, no one has divinatory powers and can help you. y mas BLA BLA BLA eres TU ... porque quizas cuando TU no habias nacido aun, ya yo llevaba años en el mundo de IT ... perhaps. but long time doesn't mean necessarily better. Y quizas yo no sepa configurar Asterisk, pero si puedo hacer un soft igual ... que te parece ? and so? what's the point? this is asterisk, not a software igual. and then, please answer in english, no other language is allowed here, in respect of other participants. Or you just wanted to say something remaining mostly hidden? Matteo. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error WARNING[-150101888] when starting Asterisk.
Hi Il giorno dom, 21-11-2004 alle 20:49 +, Mike Dent ha scritto: Ok, so I realised I was running a CVS version of * which might have been giving me the SIP problems. So I decided to get down 1.0.2. I followed the usual instructions, compiled and installed it. (FC2) [chan_zap.so]Nov 21 20:37:05 WARNING[-150101888]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_dump_info perhaps you forgot to move also to libpri 1.0.2 Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] block caller id
Hi, I have a PRI card. How do I block a caller id sent out to PSTN from a SIP client? I add a remote-party-id field privacy=full but still get caller id on a PSTN phone. I think that doing SetCIDNum() (with no args) before dial will do the trick. Matteo -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX issue at nufone
Hi, cut Nufone provide me some config examples ... I can dialout but I can't register my * Box, eg. whe I do iax show registry I got only a Request Sent and later I have a Timeout First of all, I'm (and many are) sick to see blah blah blah doesn't work with blah blah blah. This is * ML, not nufone, not any other provider. then... can you ping switch-2.nufone.net ? Have you the corrent register statement into iax.conf? eg: [general] register = user:[EMAIL PROTECTED] I hope that user:passwd has been substituted with you account data, right? perhaps doing iax2 debug on * cli will help. or even send your config files matteo. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd error at startup
hi Whenever I start asterisk with -gc, about 10 seconds passes and I get the following info: Nov 15 12:42:17 NOTICE[10369]: pbx_dundi.c:2841 destroy_trans: Peer '00:50:8b:f3:75:bb' has become UNREACHABLE! this is sample entry for digium dundi node in dundi.conf. comment it out on dundi.conf and see www.dundi.com to learn what is dundi :) Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: BRI in the US
Hi, One goal is to get BRI support in Zaptel if possible. I'm right now in the planning stage :P Plus BRI is much cooler than pots. Why invent the wheel again, what's wrong with bristuff from junghanns.net? US bri (afaik) is not EuroISDN, but NI or something like. funny mode Of course US people have their own standards : ulaw instead of alaw, NI instead of euroisdn, T1 instead of E1, miles instead of km and so on... :) /funny mode But since junghanns.net does already the cards (transport layer is the same for both, only layer-3 is different, afaik) perhaps adding to */libpri/zaptel euroisdn bri (from klaus) and us bri could be a great idea. is of course a bigger plus for * itself matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote answer not detected
hi Il dom, 2004-11-14 alle 00:13, DB ha scritto: Here's my a section of my simple extensions.conf snip exten = s,5,Dial(Zap/4/2326932|15) exten = s,6,Voicemail,u100 snip It works, but when the call is routed out on ZAP/4 (at priority 5), Asterisk seems to not realize the call is answered. After 15 seconds it proceeds to voicemail interrupting the call. Can anyone help? eh, perhaps with some details about your zap... ie what card? zaptel.conf? zapata.conf? matteo, still without divinatory powers -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail and MySQL 4.1.x
Hi, I'm using Asterisk-1.0.2 and voicemail linked with MySQL-4.1.7 library. I realised that asterisk is loosing connection with MySQL server and inform that user doesn't exist. Does anyone is using Asterisk voicemail linked with MySQL 4.1.x library? sure. never lost a connection. Using * + mysql 4.1 since when mysql 4.1 was in beta. also used asterisk + mysql cluster for a while, only in the lab, but never lost connection also in that case. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice asterisk patch
Hi, Il mer, 2004-11-10 alle 21:51, Michael Giagnocavo ha scritto: They send patches out by email? Who thought of this brilliant idea? Hmm, let's teach our users not to be cautious. the patch is pure c code. it took me 5 mins to read understand it. is very simple (but useful). Simply that patch (apart from adding some logs, comments and little code formatting) simply caches auth data AND let * manage 403 responses from the server, and this last one perhaps is the issue that was overloading BV . so, just read it (or let someone do for it) and understand that's not a problem :) Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice asterisk patch
mmmh Simply that patch (apart from adding some logs, comments and little code formatting) simply caches auth data snip too many simply here.. so, just read it (or let someone do for it) and understand that's not a problem :) or let someone do for you too late... my english is getting worse :( sorry for it. matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice asterisk patch
Hi, If you're joking, :). If you're serious, go read a primer on security. Do you patch your kernel the same way? No. I was speaking of THAT patch. that one is not so difficult, imho. a more difficult one, of course, must be understood before. or let someone that can do for you. Is not a binary file, don't you agree??? matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] example Monit control file
Hi,
Il ven, 2004-11-05 alle 05:44, David Harris ha scritto:
Can someone who is using monit to monitor asterisk post an example control
file ?
something like:
check process asterisk with pidfile /var/run/asterisk.pid
start program = /etc/init.d/asterisk start
stop program = /etc/init.d/asterisk stop
if 5 restarts within 5 cycles then timeout
alert [EMAIL PROTECTED]
you can also like a tcp monitor, if you want, like:
check process asterisk with pidfile /var/run/asterisk.pid
start program = /etc/init.d/asterisk start
stop program = /etc/init.d/asterisk stop
if failed port 5038 then restart
if 5 restarts within 5 cycles then timeout
alert [EMAIL PROTECTED]
the general section of monit can be:
set daemon 60 # Poll at 1-minute intervals
set logfile syslog facility log_daemon
set mailserver localhost
set mail-format
{ from: [EMAIL PROTECTED] }
set httpd port 2828 and
allow administrator:somepasswordhere
and on inittab file:
# monit
mo:2345:respawn:/usr/local/bin/monit -Ic /etc/monitrc
so spawning from init make you sure that monit never dies.
Also you will want to add checks for other services,
I currently use it to monitor also apache, mysql,
crond, sendmail, etc etc ect
matteo.
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia Srl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anybody explain the meaning of these messages?
Hi Am having some problems with a new asterisk installation. I get the following messages, if anybody can shed light on their meaning, I would very much appreciate it. 1- Unable to create formast pipe: Too many open files. 2- Unable to create toast pipe: Too many open files. check your ulimits (man ulimit), perhaps of that kernel max open fd is very low. To see how many fd you can have open, use ulimit -n On a standard vanilla kernel, this limit is set to 1024, that's ok for most uses (even with a full 4 pri board in use with 120 calls) you can also check how many fd has your asterisk open with something like lsof | grep pipe | grep asterisk Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New card - TE110P?
Hi, Il mer, 2004-10-27 alle 05:35, Anton Tinchev ha scritto: Will be there new card? I'm asking it, 'couse i'm going to buy 3-4 cards? Or i should wait for the new one? the card is here already. our latest shipment of E1 single span cards was of te110p Is a new card, that does E1 or T1 (like the quads) and has a new board / design (perhaps to be able to certify it like the quads) Matteo. P.S. I think that if you order now, you'll get the new cards. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reading global vars from AGI
Hi Il lun, 2004-10-11 alle 18:47, shabanip ha scritto: btw, can i read them from agi? how? use get variable agi command, like get variable foo Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Status of conference calls at Astricon ?
Ok, an italian link to nufone astricon conf room is up running. Connect it to: IAX2/[EMAIL PROTECTED]/meetme OR IAX2/[EMAIL PROTECTED]/meetmeq The first one is to listen speak. The second one is to listen only, use that if you wanna listen, perhaps with a speakerphone, in order to not send any noise and/or echo. I'll try to keep the link to nufone conf room up until astricon end. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice from one call carried on to next call
Hi Il dom, 2004-09-12 alle 10:05, Thor Atle Rustad ha scritto: I have set up asterisk with an ISDN card using i4l. When I place a call from ISDN to a SIP client, there is about a one-second delay from a word is spoken to it is heard at the other end. The funny thing, is that the last second or so of each call is saved somewhere in the depths of Asterisk and then played back at the beginning of the next call. old old problem. there's a problem with the kernel i4l driver, you must patch your kernel. search the ML for reference to that, I don't remeber where is, now. matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-Addons Changes
Hi Il gio, 2004-09-09 alle 18:18, Michael Workman ha scritto: I just downloaded it now off the CVS and it will no longer compile this kind of messages are only waste on bandwidth space. please: * don't send a message like this OR * paste the error into the email, if you need support OR * try to resolve the issue and inform the ml -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk-Addons Changes
seems that asterisk isn't installed Il gio, 2004-09-09 alle 18:48, Michael Workman ha scritto: Well this is what I am getting [EMAIL PROTECTED] asterisk-addons]$ make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cdr_addon_mysql.c:17:29: asterisk/config.h: No such file or directory cdr_addon_mysql.c:18:30: asterisk/options.h: No such file or directory cdr_addon_mysql.c:19:30: asterisk/channel.h: No such file or directory cdr_addon_mysql.c:20:26: asterisk/cdr.h: No such file or directory cdr_addon_mysql.c:21:29: asterisk/module.h: No such file or directory cdr_addon_mysql.c:22:29: asterisk/logger.h: No such file or directory cdr_addon_mysql.c:23:26: asterisk/cli.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/home/asterisk/asterisk-addons/format_mp3' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c common.c:1:29: asterisk/logger.h: No such file or directory common.c: In function `decode_header': common.c:93: warning: implicit declaration of function `ast_log' common.c:93: error: `LOG_WARNING' undeclared (first use in this function) common.c:93: error: (Each undeclared identifier is reported only once common.c:93: error: for each function it appears in.) make[1]: *** [common.o] Error 1 make[1]: Leaving directory `/home/asterisk/asterisk-addons/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo Sent: Thursday, September 09, 2004 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk-Addons Changes Hi Il gio, 2004-09-09 alle 18:18, Michael Workman ha scritto: I just downloaded it now off the CVS and it will no longer compile this kind of messages are only waste on bandwidth space. please: * don't send a message like this OR * paste the error into the email, if you need support OR * try to resolve the issue and inform the ml -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play audio into meetme conference?
Hi Il mar, 2004-08-03 alle 22:09, Paul Egger ha scritto: Is it possible to play and audio file into a meetme conference for both parties to hear? I thought I remembered reading something about it, but I can't find it now. Any help would be greatly appreciated. sure. use the call spooling file to connect to the meet me room and play any file with playback,background,mp3player,blah matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Linux for Asterisk
sorry, but I cannot resist... Il mer, 2004-07-28 alle 17:44, Mark Woods ha scritto: I'm not religous about any particular flavor of Linux, but I am highly partial to Debian, for multiple reasons. I agree. snip No, it won't be the absolute latest code, but the Debian community is pretty good about keeping packages updated. ah! ah! ah! really... oh oh, so why debian is eons later in releasing new packages... perhaps you're speaking of -unstable debian... that's wy too unstable. btw, I'm only joking... nothing serious here :) Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Sorry [Asterisk-Users] fonction Getvar
sorry, I misread your post. check from asterisk console: show manager commands if the function getvar is registered. here with rc1 works without probs. Matteo. Il mer, 2004-07-21 alle 19:13, Brancaleoni Matteo ha scritto: dialplan apps are not manager apps matteo. Il mer, 2004-07-21 alle 19:09, khady ha scritto: Hia i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown command does everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 dropping loop current 10 seconds after answer
Hi I have a TDM400 configured with 4 FXS ports, each connected to a caller-id analog trunk port on a Nortel system. Outgoing calls work great. But on incoming calls it appears that loop current is getting dropped momentarily about 10 seconds after the call is answered. Since the Nortel system is programmed to recognize this as remote party hangup it is causing all incoming calls to get dropped almost immediately. Changing from ks to ls in * doesn't make the problem go away. Any thoughts? perhaps the nortel drain too much current from the fxs card. on the bugtracker there's a patch that allows to raise loopcurrent on the proslic, feel free to test it. has resolved many issues with third party devices. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 card with R2
Hi Il mer, 2004-07-21 alle 19:37, Marcelo Rodriguez ha scritto: Hi, Does anyone know if there is a E1 pci card that can work with asterisk and support modified R2? Is this functionality of the card or the libpri driver ? the protocol (isdn,r2,whatever) is in userspace. isdnco is in libpri, r2 should be in libr2, but is far from being complete. Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Building Asterisk
Hi, Il mer, 2004-07-21 alle 20:19, Felippe Martins ha scritto: Hi I am kindda new to this mailing list. I have buit asterisk alrealdy once, but this time I am having a hard time to build it. Does anyone have anysuggestion why am I getting so many errors. unfortunately, this list doesn't have the divination plugin, so please report you errors. a mail like that is only annoying, thanks. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Large Enterprises using asterisk
Hi Why large enterprises (F500) are not shifting to asterisk as it is going to save them a lot of investment. why you say that? as far as I know there are very large * installation aroun the world. also mind that there are termination providers that are asterisk based or have asterisk in part of their network Are there some problems with asterisk ??? if used wisely, ie you know what you're doing , no Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RC1 Mirror, was Re: [Asterisk-Users] Asterisk-1.0 RC1
I uploaded every RC1 stuff to http://asterisk.espia-net.net/asteriskRC1/ just to help the digium slow link. Matteo Brancaleoni -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is there alist of codec by asterisk version?
Hi Il sab, 2004-07-10 alle 05:43, Kevin Walsh ha scritto: Joe Baptista [EMAIL PROTECTED] wrote: Is there alist of codecs asterisk actually has per version number - i.e. 0.7, 0.9 etc? if you have it installed, do show translation on the cli and you'll see all codecs supported, along with translation tims I believe Asterisk has the same codec list in all of its versions. Well, at least for the versions I've seen. I'm sure someone will rush to correct me if I'm wrong. as far as I know, the stable branch doesn't have g726 as in HEAD branch. matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP hackers gut Caller ID
hi... here in Italy is almost impossible to set an invalid cid, if is out of your allowed space. ie. if you have X numbers on your PRI, you can only set one of these. nothing more. on bri you simply cannot do nothing. just my 2 cents. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *8# into invalid extensions
the real sequence is only *8 Matteo. Il lun, 2004-07-05 alle 15:00, Vladyslav ha scritto: Hi All! Have a problem with remote call pickup via sip. When 1 sip phone is ringing and I'm trying to pickup a call from another sip phone by dialing *8# I'm getting: -- Sent into invalid extension '*8#' in context 'from-sip-post' on SIP/ciscok-8d39 such configs: zapata.conf -- context=inbound-analog callgroup=2 channel=2 -- sip.conf -- [ciscok] type=friend host=dynamic username=ciscok canreinvite=no callgroup=2 pickupgroup=2 mailbox=100 qualify=1000 dtmfmode=rfc2833 trunk=yes [ciscok2] type=friend host=dynamic username=ciscok2 canreinvite=no callgroup=2 pickupgroup=2 qualify=1000 dtmfmode=rfc2833 trunk=yes -- Please help. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Call Pickup
as far as I know, no. Il lun, 2004-07-05 alle 18:56, Adolfo R. Brandes ha scritto: I've looked in the obvious places but haven't found a definitive answer to the following: can an IAX extension (an Iaxy phone, for instance) do call pickup via *8? Adolfo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *8# into invalid extensions
Hi Il lun, 2004-07-05 alle 20:12, Brian K. West ha scritto: *8# works on sip that uses the # as the send key. sure, but since he gets -- Sent into invalid extension '*8#' in context 'from-sip-post'... means that he's sending *8# ... matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax or sip
hi snip trunking will save some bytes in flight iff one has four or more streams moving between two pbxes. you call -30% (more or less, depending on the codec) in bandwidth only some bytes ? reinvite rules, especially in a geographically distributed use scenario. that could be done with iax. see the notransfer flag in iax.conf you can move the entire call away, not only the rtp stream. now, i could see a network of iaxen if there was some way to negotiate call routing with costs etc. but trip looks a bit ugly and kinda far away. and it certainly is not part of current play. but is easy to add info to iax to carry what you need. what am i missing here? experience. btw, SIP is certainly needed 'cause of the clients... much more available than iax ones. but for server to server pov, iax is sure a better choice. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nat problem
add localnet=x.x.x.x/mask in the [general] section. where x.x.x.x/mask is the internal network mask of your * box. Matteo. Il dom, 2004-07-04 alle 05:33, Damian Minkov ha scritto: I have the following situation. My Asterisk Box is behind firewall ( for example 10.1.1.2 ) I have mapped 5060,1-10010 and in rtp.conf I have said this range of prots 1-10010. I'm tring to dial a PSTN from another PC with Sip phone in internet with external ip. I can hear the voice from the PSTN , but The Other Side can't hear me. I ran Ethereal and so that all rtp packets going from the calling phone are with destination 10.1.1.2. What to do to configured it right ? in Sip.conf [general] nat=yes externip=213.x.x.x [sipphone] [damencho] type=friend username=damencho host=dynamic nat=yes canreinvite=no ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P-E100P circuit board differences
hi Can someone please confirm that their E100P says T100P on the artwork? yes, the board is the very same. the only difference is in the framer chip. btw, just plug it, build zaptel load wct1xxp module and you'll see that is an E1 card. (in dmesg) Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile Error
checkout the libpri and compile install *before* asterisk Matteo. Il gio, 2004-06-24 alle 19:01, Joseph ha scritto: Just did a new cvs download and then tried to compile. I get this error message: chan_zap.c:59:2: #error You need newer libpri Then there are some more chan_zap.c errors. Here is the cvs command: export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout zaptel asterisk libpri And the make command #cd /usr/src/zaptel #make #cd /usr/src/asterisk #make And I did this after moving the current zaptel, asterisk, and libpri to archival. Where do I get this file? Or what am I doing wrong... -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zttool CLI
nothing ready, but is pretty simple to strip away all newt stuff from zttool and make the output go to stdout... On Wed, 2004-06-23 at 18:45, [EMAIL PROTECTED] wrote: Hello, I need to check red alarms status from the script, but asterisk CLI zap show channel 1 or pri show span 1 does not tell me this. zttool does, but I can run it only in interactive curses mode. Is there any ready solution? Best Regards, Ivan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HST Saphir with Asterisk
Hi Il dom, 2004-06-20 alle 00:21, Andrew Kohlsmith ha scritto: On Saturday 19 June 2004 17:59, Julian Pawlowski wrote: I would like to use my existing HST Saphir V S2M PCI with Asterisk. Unfortunately I could not find any information about the usage with Asterisk. snip The official recommendation is to purchase an X101P from Digium. If you're resourceful (hint: use google and search the mailing list archives) you will find that it is possible to obtain the same hardware for much more economically, although it is officially unsupported and others have had trouble buying X101P clones. the card in discussion is a isdn E1 pri card. HST is listed as capi hardware manufacturer under www.capi.org you should be able to use that with kapejod wonderful chan_capi and kernel drivers from http://www.hstnet.de/english/downloads/isdn/saphir_5_primary_pci/index.asp unfortunately they provide only precompiled kernel modules, and only for certain kernel versions... so you're stick to what they use. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions question
What's that ?
Dial(SIP/-083601e0, ZAP/g1/h) ?
why 'h' ?
don't use exten = _.,1,blah , but
try with exten =_X.,1,blah
Matteo
Il ven, 2004-06-11 alle 23:59, Christian Gatti ha scritto:
ser forwards a sip message with extension 9996 to asterisk which
plays my 'userisoffline' message and hangs up and should stop here but
instead asterisk continues to process the match everything extension ._
and dials out which is not what I want...
if I change the starting priority of the Dial app to a higher level
than 3 asterisk stops after the hangup but then doesn't accept any other
extension that should be dialed.
how can this be done?
part of extensions.conf:
...
exten = 9996,1,Wait(1)
exten = 9996,2,Playback(userisoffline)
exten = 9996,3,Hangup()
... other extensions with more than 3 priority levels
exten = _.,1,Dial,ZAP/g1/${EXTEN}
exten = _.,2,Hangup()
...
and the output in asterisk:
---
Connected to Asterisk CVS-05/26/04-02:55:14 currently running on sip (pid = 7472)
-- Remote UNIX connection
-- Executing Wait(SIP/-083601e0, 1) in new stack
-- Executing Playback(SIP/-083601e0, userisoffline) in new stack
-- Playing 'userisoffline' (language 'en')
-- Executing Hangup(SIP/-083601e0, ) in new stack
== Spawn extension (from-ser, 9996, 3) exited non-zero on 'SIP/-083601e0'
-- Executing Dial(SIP/-083601e0, ZAP/g1/h) in new stack
-- Called g1/h
-- Channel 1, span 1 got hangup
Jun 11 23:39:43 WARNING[491541]: app_dial.c:349 wait_for_answer: Unable to forward
voice
Jun 11 23:39:43 WARNING[491541]: app_dial.c:349 wait_for_answer: Unable to forward
voice
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
-- Executing Hangup(SIP/-083601e0, ) in new stack
== Spawn extension (from-ser, h, 2) exited non-zero on 'SIP/-083601e0'
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia Srl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seperate asterisk VM system possibility
Hi Il mer, 2004-06-09 alle 22:45, Steve Dolloff ha scritto: I would like to move voicemail to a dedicated server but I can't figure out how to make the MWI work if the ATA doesn't register to the voicemail server. The main reason for this is redundancy. I have two SIP registrars running and in the case of a failure from the primary, both the gateways and the ATAs switch over to the secondary, but since the voicemail is on the primary, it also fails. Anyone have any suggestions? I assume that the registar (are asterisk) are configured in the same way. so the easy way it to rsync /var/spool/asterisk/voicemail to the backup server, so if the primary goes away, you'll have your voicemails as before. of course when the primary is back a sync back must be done... matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router, Firewall, SIP Rewriter, and GnuGK
hi I am thinking of replacing the router box because hardware is getting flaky. I do not want to go through pain of assembling all this stuff together again. why don't u just replace the hardware clone the installation, with a simple disk copy? perhaps only changing the network cards modules will be what you'll need to do... Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: zaptel startup issues solved
Hi. cut Obviously I chose the custom option when I setup the OS. Is there someway to fix the zaptel code to not be so picky? I think it expects the kernel source tree to match the running kernel. If you had built a new kernel called 2.6.5-1.315custom and then booted from it, you would probably have built zaptel successfully. I think :-) sure, that happens because kernel-source package from fedora has the kernel version set to blahcustom, as long as many other rh versions. the solutions are 2: * use a plain, vanilla kernel * as Tony suggests, rebuild the kernel first, install it and then build zaptel Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
Hi I was planning to use the output of asterisk -rx show queues in a script when I noticed that sometimes asterisk only outputs the first line of the response. e.g: why don't you use the manager interface? it's much better... Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk firewall config
Hi. Il dom, 2004-05-23 alle 01:52, Tony Hoyle ha scritto: Surely it depends on who's calling me - if they're using a SIP phone it'll come in over the SIP port, and if they're using an IAX phone it'll come in over the IAX port - ie there's this context in the default iax.conf: [guest] type=user context=default callerid=Guest IAX User for letting unauthorized user to call you over IAX(2). Like a pstn call... everyone can call you if the have your number (or IP in Voip calls) If you don't want that, just delete that entry :) btw. how many rtp streams do I need? I only have 1 phone at the moment (max. will be about 4 I think). mmh... I dunno the values of that association, but bear in mind that: * are only UDP ports * are opened only during a RTP session, in a dynamic way so leaving open ports 1 to 2 UDP as in default rtp.conf isn't a problem, since there's not any port open... (unless you run any udp service on that interval :) ) and a portscan will detect these port as closed. only during a call, * and the phone will handshake an RTP port and use that. otherwise will be closed. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My TDM-400P FXO experience
Hi On Tuesday, May 18, 2004, at 07:50 PM, Leo Ann Boon wrote: f. Be careful about the zap channel naming. With the old XP101, the first channel (card) is Zap/1 and the second Zap/2. With the TDM, it's Zap/1-1, Zap/2-1 ... Zap/4-1 for the 4 ports on the first card and Zap/1-2 ... Zap/4-2 for the second card. You might need to update your dial plan. that sounds very strange are you sure? as far as I know each Zap channel is unique, so with 2 cards you should have from Zap/1 to Zap/8 The difference between Zap/1-1 and Zap/1-2 is (for example) when you have 2 calls on the same zap channel, ie when you have a call on the phone on Zap/1-1 and pressing the flash key, you create Zap/1-2 on which you can dial another exten. I don't think that Zap/1-1 and Zap/1-2 are first channels on different cards at all please double check that (as I'll do...) Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk firewall config
Hi Il dom, 2004-05-23 alle 00:11, Tony Hoyle ha scritto: The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? all depends on what you need to do. if you use only zap channels and no Voip, perhaps the only port you need to open is ssh (if using it, of course) if you plan to do only IAX, only port 4569 UDP needs to be opened. but if you plan to do only sip you need only port 5060 UDP and 1 to 2 UDP for sip rtp stream (configurable into rtp.conf) so... all depends :) I'm concerned as anyone that could guess an extension numberpassword could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there isn't one documented... mmmh... setting into the extension seems to me the same as setting into iax.conf (or sip.conf), or not? otherwise... use very strange passwords along with superstrange usernames I bet someone to get a login data like username : 2h729872pcnt with pw : inr2.f2f2232DDFW3r or not :) ? -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Wokflow diagram
I use callflow (callflow.sourceforge.net) works under linux with ethereal dump, and produces html+images pages, for viewing them via a web browser. Matteo. Il ven, 2004-05-07 alle 15:14, Ignace CARIA ha scritto: Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN WAN ISDN bridge possible?
sure. use * with: IAX2 for sending voice via WAN (I suppose is internet) and then for ISDN you can: if is PRI , get 2 digium cards if is BRI , get zapbri cards matteo Il lun, 2004-05-03 alle 19:41, Patrick Stuckenberger ha scritto: Hi list, is it possible to create something like a ISDN-WAN-WAN-ISDN bridge? We have to change our location, but our number and the telephone system should shoulb stay the same. kind regards, Patrick Stuckenberger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] nat friendly tftp server
Hi, I was searching along the net for a tftp server nat-friendly, in order to provide new firmware to our budgetones, which are 90% of time nat'ed. I came across this one: http://troja.ath.cx/~zond/jtftp/ works ok. is written is java, under linux. I'm using it now and seems ok. very simple, but does the job. Matteo. -- -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nat friendly tftp server
sure... but the one I suggested is meant to be run as a service on a linux server, maybe the same as your asterisk box. why having a winzoz machine to provide only tftp service ? :) Matteo. Il dom, 2004-05-02 alle 14:05, Ian Pilkington ha scritto: I personally download the TFTP server provided free by SolarWinds. I simply open up the required port and is very happy. http://support.solarwinds.net/updates/New-customerFree.cfm Regards, Ian. --- Outgoing mail is scanned but not certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.672 / Virus Database: 434 - Release Date: 28/04/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3com SIP phone working with asterisk
interesting... did you tried all the function? ie, can you put a call on hold, and more important do blind supervised transfer? what about the prices? more or less, just to have an idea... tnx, Matteo Il ven, 2004-04-23 alle 17:08, Lisa Xie ha scritto: Hello everyone, I just like to let you know that I tested Asterisk with 3COM SIP phones and it worked fine. The 3Com phones are old ones with the same look of NBX 2102 phone but different product number: P/N: 655005001 Rev B There is no special set up except that I have to specifically put allow=ulaw in sip.conf. Otherwise, there is codec unrecognized error. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=ulaw; Allow all codecs Lisa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 encoding of Monitor files
use lame Il ven, 2004-04-23 alle 17:33, Mike Machado ha scritto: I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call: Monitor(wav|test) 'file' output: test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Sox resample: sox test.wav -r 32000 newtest.wav Bladeenc call: bladeenc newtest.wav newtest.mp3 mpg123 newtest.mp3 # sounds like Im listening in fast-forward mode... Any suggestions on how I can get mp3 versions of files produced by Monitor? On Thu, 2004-04-22 at 15:49, Roscinante wrote: On Thu, 22 Apr 2004, Dennis Sorge wrote: Any recommendations for ripping my .wavs to MP3's? I'm running Mandrake 9.2 for a potential music server. Thank you in advance for your suggestions. I use bladeenc, I imagine there is some spiffy front end for it out there somewhere.. ___ Lug-nuts mailing list [EMAIL PROTECTED] http://felix.mikesoffice.org/mailman/listinfo/lug-nuts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One, två, tre, quatre, cinq ... International numbers in say.c
hi Olle. I have a patch for italian. should it be for plain say.c or for your modified say.c ? Also I have some the .it audio files, I'll ask if I can distribute them (perhaps with some credit to the company I work for, that payed them...) Matteo Il lun, 2004-04-19 alle 21:53, Olle E. Johansson ha scritto: http://bugs.digium.com/bug_view_page.php?bug_id=0001429 * Support for other language syntaxes in saynumber Accidentally I opened this can of worms to see if we can add support for other language syntaxes for saying numbers. Seems like Swedish, english and norwegian follow the same syntax. I've integrated existing patches for french, danish and soon portuguese syntax. The steps we're taking are: * First a quick-fix only for saying numbers * Adding documentation and sample sound files Many patches require additional sound files compared with the english set. * For a coming release we need a more general architecture that includes more phrases, time and date. This will be done with loadable modules for various languages. I need the original contributors of danish, french and portuguese to fax a disclaimer to Digium. See http://bugs.digium.com Also, I need users in these language territories to test the patch and add feedback to the bugtracker. I can try to put all this together into one unified patch, but not test everything for every language. If you have a patch for another syntax, please add it quickly to the bugtracker and fax in the disclaimer, so we can use it. If you have sound files for a language with decent quality that you can share to the community, please do so by adding them to the bug tracker. * If we all work on this together quickly, we may have a working say.c in the CVS soon. But to even ask a committer for support, I need test results up there on the bug tracker. * Thank you for your support! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream and stun
you don't need stun to make GS work under NAT with * Just set NAT=yes into the GS, and leave the stun server addr entry empty. And set nat=yes into the sip.conf entry. That's all Matteo. Il dom, 2004-04-18 alle 11:26, Richard ha scritto: Hi, I noticed some issues with how grandstream handles stun test. GS is running version 1.0.4.50. First I reset the NAT router. Then reboot GS, get results of restricted cone. Immediately reboot GS, get results full cone. I tried quite a few public and commercial stun servers. Also tried different model/version of linksys routers. I always got the same issue. Winstun on the PC doesn't have this issue. Some ngrep on the stund 0.91 on Fedora linux revealed winstun had about 20 UDP packets back and forward. However GS only had less than 10. Did anyone notice the same problem? Thanks, Richard __ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25 http://photos.yahoo.com/ph/print_splash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream and stun
the only difference with NAT=yes into GS is that enabling it the phone will send periodic (every 20secs @default) empty UDP packets to the SIP server, keeping the NAT hole open... so you don't have to do a dnat rule onto the nat'ing device. very useful :) Matteo. Il dom, 2004-04-18 alle 16:46, Ryan Thrash ha scritto: FYI, with 1.0.4.55 and NAT set to off (but with the * config set as nat=yes), I'm able to bypass stun servers completely with a GS phone as well. HTH, Ryan On Apr 18, 2004, at 5:08 AM, Brancaleoni Matteo wrote: you don't need stun to make GS work under NAT with * Just set NAT=yes into the GS, and leave the stun server addr entry empty. And set nat=yes into the sip.conf entry. Il dom, 2004-04-18 alle 11:26, Richard ha scritto: Hi, I noticed some issues with how grandstream handles stun test. GS is running version 1.0.4.50. First I reset the NAT router. Then reboot GS, get results of restricted cone. Immediately reboot GS, get results full cone. I tried quite a few public and commercial stun servers. Also tried different model/version of linksys routers. I always got the same issue. Winstun on the PC doesn't have this issue. Some ngrep on the stund 0.91 on Fedora linux revealed winstun had about 20 UDP packets back and forward. However GS only had less than 10. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intel 536ep as a FXO?
buy a x100p for 100 bucks and support digium. Matteo. P.S. and you'll have free installation support from digium and a rock solid hw made for asterisk. Il dom, 2004-04-18 alle 16:43, Marcin Mazurek ha scritto: Hi, I've seen some reports about ruuning Intel modem with 537 or MD3200 chipset running with Zaptel drivers as a FXO port. Did anybody managed to set up a PCI faxmodem based on Intel536ep chipset to work with * and Zaptel drivers? Modem seemd to work just fine with Linux, but the driver says no;) some more info: Linux 2.4.26 mazuchna:~# cat /proc/pci | grep 536 Communication controller: Intel Corp. 536EP Data Fax Modem (rev 0). mazuchna:~# lsmod Module Size Used byTainted: P ztdynamic 6692 0 (unused) zaptel177280 0 [ztdynamic] Intel536 876524 0 (unused) mazuchna:/lib/modules/2.4.26/misc# insmod wcfxo Using /lib/modules/2.4.26/misc/wcfxo.o /lib/modules/2.4.26/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg is there anything more I can do? tia mazek -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intel 536ep as a FXO?
flame mode Only a Digium zealot would call the X100/X101P 'rock solid hw' -- Digium is I'm not a zealot , nor endorsed in any way by digium.. for me , with a a lot of X100P installed, is rock solid. never missed a hit. They needed a cheap FXO interface for the masses and for now, that's what we have. It's certainly not a good solution, but it is *a* solution. Is a good solution. At least the combination callItAsYouWantcard + zaptel drivers... I am eagerly awaiting proper stocking of the IAXy and an FCC-certified FXO module for the TDM400P -- I think those should be Digium's flagship products, not a rebranded craptastic WinModem. Hope so. surely works better than the intel one, and I don't see any reason in loosing (your, of course) time into making it work under zaptel. Isn't it a craptastic WinModem also? even if made by Intel? /flame mode -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk database support
I hear some echo there :) simply, you can define sip friends from a database. just create the table, enable SIP_FRIENDS into channels Makefile and read chan_sip.c how to set db access (db access data must be into sip.conf) but, firstofall, you must be familiar with sip.conf and friends/user/peer definition in order to understand how it works... matteo Il sab, 2004-04-17 alle 17:21, gaillac harry ha scritto: Hello, Is it possible to use a database for provisionning sip clients? CVS provides sip-friends.sql in order to create tables (not database) what may i do with that tables? Regards Harry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 Admin Password
Wipeout, I bought the phone new about a year ago so its not provider locked.. I set the password to be nothing (I think) and then I set admin mode off, then when I tried to get into the admin area I couldn't, it would seem that either there is a bug that doesn't allow a blank password or it did not set it to be blank.. I will have to get hold of the distributor next week.. Later.. if I don't remember wrong, the default pw is . btw, just download the firmware from snom website, put it onto a tftp server and rename it as snom200.bin then reboot the phone, and as soon as it powers up (don't let the phone boot at all), press a key. it will prompt an ip addr,netmask,gw and tftp server addr. fill the values and go on. the phone will load the firmware, and everything will be set @ default values. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Missing vm feature - turn off voicemail
directly into voicemail I don't think that's possibile. but you can fake this function, simply using in the right way dbput / dbget and if conditions... Matteo. Il gio, 2004-04-15 alle 18:45, Iain Stevenson ha scritto: Listening to the options on the voicemail system it seems to be missing a feature for users to turn voicemail off completely. This seems a rather glaring omission. Does the feature of turning off message recording via the phone exist - or does it need a patch? Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small linux distro to run * in old boxes
I made a custom fedora mini distro, something like 350 megs, including apache,php,mysql webmin of course installable from a cd in 20 minutes, more or less :) at the end you have a fully working asterisk installations, along with some basic tools like webmin and a full webserver Matteo. Il ven, 2004-04-09 alle 18:02, Victor Perez ha scritto: Has anybody tried to install * in any of these minimalist linux distros like tinylinux? Which linux distro would you use to run * in old P2, P3 boxes? Regards, Victor Perez [EMAIL PROTECTED] (469) 221-4189 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small linux distro to run * in old boxes
Hi I made a custom fedora mini distro, something like 350 megs, including apache,php,mysql webmin of course installable from a cd in 20 minutes, more or less :) at the end you have a fully working asterisk installations, along with some basic tools like webmin and a full webserver Are you going to be making this available or is it something yo created for inhouse use only? dunno yet. is not to me. the whole packahe contains also our web manager for asterisk (configuration and several tools like call recording,contacts,manager view,blah blah blah) that's not open. as soon as I'll have a fully working stable installer, (now works good, but I have to polish some things) *perhaps* I could arrange to distribute at least a version without the web manager... hope so :) Matteo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] attendent transfer on ZAP channels
Hi Il mer, 2004-04-07 alle 20:28, Bartosz Jozwiak ha scritto: hello, Is it possible to make attendant transfer (not blind) with ZAP channels ? sure. just press the flash key on the phone (also known as the 'R' key, at least in EU), you will hear the dialtone, while the caller is put on hold. dial the extension you wanna transfer to, speak with the remote party and then: hangup to transfer to the dialled exten OR press R to be in a 3-way conference (of course the remote party should not hangup) OR just press R to get the call back (and the remote party should hangup) OR press R twice to get the call back is the remote party doesn't hangup immediately threewaycall and transfer must be enabled into zapata.conf matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk call manager
Hi. try adding a whitespace between ':' and the command. Eg. action: login enter blah blah Matteo. I am trying to setup the call manager and I configured the manager.conf file. When I try to telnet 0.0.0.0 5038 It says trying 0.0.0.0 Connected to localhost Escape character is '^]'. Asterisk Call Manager/1.0 Then I type Action:Login (enter) Username:sam Secret:sam Then I enter twice I get Response: error Message: missing action in request I am not sure what it means. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmgi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cron job to reboot GS101
sure, we do that with a cron job that fires up a script that connects to the GS web interface and reboots is. the job is launched every 4 hours. Also the GS web interface is down during a call, so there's no risk to hangup undergoing calls. (and the scripts also tries several times, before going to the next phone). fortunately, the GS reboots fast, so isn't ever noticed. I cannot release that code (is part of our asterisk based solution), but you can easily do that with php... Matteo. Il sab, 2004-04-03 alle 06:36, dkwok ha scritto: Does any one regularly reboot GS101? It sometimes lost registration with * and needs to be reboot. What is the best way to do it by cron? David Kwok ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie - Bashing head on wall! - RH9 - * - How do I install AVM C2 ISDN Pretty Please!
Use yast2 from suse and configure isdn: He's using redhat, not suse :) btw, I'm using capi with avm under redhat 7.x, 8.x , 9 and fedora. What're your problems? please be more descriptive and we'll help you. Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zyxel wifi sip phone
just found that over the net. looking forward to be able to try it with * :) http://www.zyxel.com/product/P2000W.html -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new2agi -php
Hi
The error message is generated by PHP... If your PHP version is higher
that v4.3.0
line 5: $stdout = fopen('php://stdout', 'w');
should be
$stdout = fopen(STDOUT, 'w');
Wrong... STDOUT is already open within php cli (4.3.0 and above)
so just do
fwrite(STDOUT,blah);
and you're done.
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA 200 Fax
hi It works, sort of. Basically, about 1 in 4 faxes are going out without errors. Of course, that's to an IAX peer, so I'm not sure if it's a problem with the IAX peer or with the Siupra. check you IAX connection. perhaps is using gsm and that could explain the failure Faxes must be sent uncompressed, ie with [u-a]law as codecs. I have 2 fax machines over SIP here (ulaw) and never missed an hit :) -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc bri with overlap sending/receiving
how is your outgoing dialplan?
tried into specifing something like
exten = _XXX.,1,Dial(blah/${EXTEN})
note the point : this rule will match
at least 4 digits, but also 5,6,7...N
matteo
Il sab, 2004-02-28 alle 22:34, Jan Baumann ha scritto:
Hi all,
I am currently testing Klaus-Peter Junghanns' zaphfc bri driver
0.0.2rc12 with two HFC ISDN cards in PtP setup - one connected to telco,
one to the legacy pbx - and try to dial from a pbx extension out to the
pstn through astersik.
This works perfectly as long as I dial on hook and pick up after dialing
the complete number.
Using the isdn phone (and any analog pbx extension which cannot prepare
dialing on hook) the way people are used to (first pick up, then dial)
results in dialing only the first few digits out to the Zap channel
connected to pstn and call setup to fail.
Obviously this is a problem with overlap sending/receiving in the zap
channels. Unfortunately we have a variable length numbering plan in
germany (local numbers can be anything between 4 and 9 digits long), so
putting more X in the regex doesn't seem to be an option.
Ideas how to get this work are greatly appreciated and very welcome. :)
Thank you and regards,
Jan Baumann
My current config:
extensions.conf:
; outbound dialing local calls
; try Enum, then PSTN
[local-pstn]
exten = _0[1-9]XX.,1,EnumLookup(49821${EXTEN:1})
exten = _0[1-9]XX.,2,SetCallerID(49821xx)
exten = _0[1-9]XX.,3,Dial(${ENUM},30)
exten = _0[1-9]XX.,4,Goto(102) ; Failure on SIP, fallback to PSTN
exten = _0[1-9]XX.,52,Congestion
exten = _0[1-9]XX.,102,SetCallerID(xx)
exten = _0[1-9]XX.,103,Dial(Zap/g1/${EXTEN:1},,tr)
exten = _0[1-9]XX.,104,Congestion
zapata.conf:
switchtype = euroisdn
; to/from ISDN PtP
signalling = bri_cpe
pridialplan=unknown
echocancel=no
immediate=no
group = 1
context=pstn-in
channel = 1-2
; to/from the PBX
signalling = bri_net
pridialplan=unknown
echocancel=no
immediate=no
group = 2
context=intern
channel = 4-5
zaptel.conf:
# PSTN DTAG
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
# PtP to PBX
span=2,0,3,ccs,ami
bchan=4-5
dchan=6
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] budgetones + G726
hi... I was playing with g726 and budgetones, here's my quick experience: * firmware 1.0.4.40 ... the phone just crash: as soon as you start a call in g726, only a squeeze is heard, all the display icons are lit and the phone is dead :) * firmware 1.0.4.46 : the phone survives, but the audio is only noise... no conversation is possible. Since g726 works ok with cisco sipura, I think that could be a phone bug... any other experience ? Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference and transfer
hi 1) Does Asterisk support a consulting transfer? E.g., call comes in, Mary answers, Mary presses Transfer and dials Joe, verifies that Joe answers and informs him who is calling, and then presses Transfer to complete the transfer? on zap channels yes on sip channels yes, depending if phone supports that too 2) How does one set up a 3-party conference? With a traditional phone system, you press the Conference button on the phone, dial the 3rd party, and press Conference again. This doesn't seem to work with Asterisk. see app_meetme Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] max asterisk load
hi. Assuming calls will be using G711, a little of g729 (max-15), 1000 SIP (multi-vendor: Cisco, GS, MOtorola, Dlink,etc), of which 80% clients are behind NAT, and server of I-P4 2GHz, 80GB HD and 4GRAM, will that work? depends on how many concurrent calls you have. you can have 10k users, but only 10 calls a time, so a little server is needed. think of that. supposing you have 30% of your users doing calls at the same time, I would say to to your question. perhaps a dual 2.4 gigs fits better. imho is to have several servers, and spread the load between them. the best way (but far from being perfect) is to test it. put up 2 asterisk servers, and generate calls between them. see when the load is high (or asterisk crashes : ) now you have an idea :) Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording
hi. mind that current CVS already has mixing support, if soxmix is installed into the sistem, so loligo.com exten file could be simpler. matteo. I would recommend checking out this link. http://www.loligo.com/asterisk/current/extensions.conf Darren Wiebe [EMAIL PROTECTED] kemal asad wrote: Is there a way to record phone conversations. I am using Asterisk with a IP phone and the digium hardware to make ouside calls.we need to have all outside calls One S100U USB FXS Interface (including the USB cable) One X100P PCI FXO Interface the system is working quite well. but we have got a new requirement can the asterisk server record all out going call. if yes please send me links on how to set it up. Thanks, Kemal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
