[Asterisk-Users] H323/*/IAX - Firewall - IAX/*/H323
Hi Folks, I have two H323 Polycom video conference system with a Linux firewall Iptables in the middle. I am not getting to make H323 working in this setup and I was wondering to put two * servers as a bridge to jump the firewall using IAX. The idea basically is: h323 Polycom IPTABLES VideoConference Device -- *(LAN) --- *(WAN) H323 Polycom chan_h323 chan_iax chan_h323 or chan_oh323 or chan_oh323 Question before spending some time with it... should it work ? Thanks, Isamar ___ 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) no dialtone on a TDM400P card
Thomas Andrews wrote: On Sun, Nov 14, 2004 at 11:46:15AM +0100, Soren Rathje wrote: Can you post your actual configuration ? /etc/zaptel.conf fxols=1 #S100U fxsls=2 #X100P loadzone = us defaultzone=us Looks fine allthough the comments are wrong :-) /etc/asterisk/zapata.conf [trunkgroups] [channels] context=default Remove the next two lines, switchtype is related to ISDN circuits and the signalling you specify later in the file. switchtype=national signalling=fxo_ls rxwink=300 usecallerid=yes ; Type of caller ID signalling in use ; bell = bell202 as used in US, v23 = v23 as used in the UK, dtmf = DTMF as used in Denmark, Sweden and Netherlands ; cidsignalling=bell ; ; What signals the start of caller ID ; ring = a ring signals the start, polarity = polarity reversal signals the start ; cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no context=internal callerid=House Phone 1234 # Change to match your settings.. signalling=fxo_ls You may want to change Loop Start to Kewl Start (fxo_ks) channel=1 context=incoming callerid=asreceived signalling=fxs_ls You may want to change Loop Start to Kewl Start (fxs_ks) channel=2 /Soren ___ 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] Garbled sound - CPU or traffic problem?
Hi! I'm using asterisk on my local server. An analog phone is connected to a PhoneJack lite. The computer (PII-333) is connected to the internet via DSL. Sometimes the sound is really clear, as if it was a normal telephone call. But there are times too, when I hear the other person really garbled. That means the sound is disturbed with little breaks. The problem seems to occur on both sides. I used ULAW as speech codec. Is my cpu too slow? Could it be a problem with my connection? I do not use a donkey-client or something else. What can I do to analyse where this problem comes from? Could it be a problem with my SIP-Provider? (Sipgate) Bye! Michael ___ 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) no dialtone on a TDM400P card
Thomas Andrews wrote: Hi Cirelle, On Sun, Nov 14, 2004 at 07:28:56AM -0500, Cirelle Enterprises wrote: you might have to power the box down - no power for the modules to load (appears to be common for this card) if that is the case, do a search on tdm in the email archive as there is a fix for the reboot problem I'm afraid I don't understand this sentence. I did power the machine down to make sure that the power was indeed plugged in. Are you saying there's a startup problem with these cards. module 1 (closest to the top of the bracket (furthest from the pci connector) is for the phone line module 2 is for the handset I don't agree, but perhaps I'm wrong. The green module (fxs) is number 1 and the red module (fxo) is number 2. As I understand it you plug the handset into the green one (fxs). Not so ? Green is phone Red is line if we're using modem terminology.. :-) /Soren ___ 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] Elesign - ESC2420.
Hello list , Does someone uses ESC2420 ATA adapter?! http://www.elesign.com If positive , please let me know. Regards, -Jefferson Carvalho ___ 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) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 02:22:18PM +0100, Soren Rathje wrote: /etc/zaptel.conf fxols=1 #S100U fxsls=2 #X100P loadzone = us defaultzone=us Looks fine allthough the comments are wrong :-) Thanks Soren. I made all the changes you suggested, but do I have to change the above to ... fxoks=1 fxsks=2 ... if I changed to kewel-start in zapata.conf ? I assumed so, and went ahead and did so. Still no dial-tone though. Thanks, Thomas ___ 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) no dialtone on a TDM400P card
Thomas Andrews wrote: On Sun, Nov 14, 2004 at 02:22:18PM +0100, Soren Rathje wrote: /etc/zaptel.conf fxols=1 #S100U fxsls=2 #X100P loadzone = us defaultzone=us Looks fine allthough the comments are wrong :-) Thanks Soren. I made all the changes you suggested, but do I have to change the above to ... fxoks=1 fxsks=2 ... if I changed to kewel-start in zapata.conf ? My fault, you should change it in both files. I assumed so, and went ahead and did so. Still no dial-tone though. Hmm.. Does Asterisk load chan_zap ? /Soren ___ 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) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 03:01:28PM +0100, Soren Rathje wrote: Hmm.. Does Asterisk load chan_zap ? I believe so: [chan_zap.so] = (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXO Kewlstart signalling -- Registered channel 2, FXS Kewlstart signalling -- Automatically generated pseudo channel == Registered channel type 'Zap' (Zapata Telephony Driver) == Registered application 'CallingPres' == Manager registered action ZapTransfer == Manager registered action ZapHangup == Manager registered action ZapDialOffhook == Manager registered action ZapDNDon == Manager registered action ZapDNDoff == Manager registered action ZapShowChannels Thanks again! Thomas ___ 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] How to route all incoming call to the defines context in extensions.conf
I'm testing the Asterisk in a pure sip configuration, presently testing it with a number of sip phones, some registrations to a SER-server and with password protection for outgoing calls to the SER-server. I have a problem with incoming calls. When I get an incoming call, Asterisk finds a peer in sip.conf and tries to route the call to that peer. Apparently this happens because the address of the incoming call includes a domain name with the same ip address as the ip address of the domain name of the host entry in my [sipout] definition in sip.conf. I would like to route all incoming calls to my extension.conf [sipin] heading, even when a peer is found. If I delete the [sipout] definition in the sip.conf, I receive all incoming calls in the way I want, but I cannot make outgoing calls. If I could include a username and password in the dial command, I could do away with the [sipout], but I have found no way to include this in the dial command. I would appreciate any suggestions to solve the problem. Thanks, Jon Bruel ___ 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) no dialtone on a TDM400P card
Thomas Andrews wrote: On Sun, Nov 14, 2004 at 03:01:28PM +0100, Soren Rathje wrote: Hmm.. Does Asterisk load chan_zap ? I believe so: [chan_zap.so] = (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXO Kewlstart signalling -- Registered channel 2, FXS Kewlstart signalling -- Automatically generated pseudo channel == Registered channel type 'Zap' (Zapata Telephony Driver) == Registered application 'CallingPres' == Manager registered action ZapTransfer == Manager registered action ZapHangup == Manager registered action ZapDialOffhook == Manager registered action ZapDNDon == Manager registered action ZapDNDoff == Manager registered action ZapShowChannels Hang on... What line pair do you use on the phone; 1+4 or 2+3 ?? I believe the correct pair to use should be 2+3. /Soren ___ 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] SysMaster and GPL Violation
On Sun, 2004-11-14 at 08:30, Gilad Ben-Yossef wrote: Further, that really does seem to fly in the face of the spirit of the GPL, and this is touched on by the GPL FAQ: http://www.gnu.org/licenses/gpl-faq.html#TOCReleaseUnderGPLAndNF http://www.gnu.org/licenses/gpl-faq.html#TOCDeveloperViolate You may or may not be aware that to contribute code to FSF owned and maintained software one needs to assign copyright to the FSF in much the same way one is required to assign copyrights to Digium. True, I don't think the FSF are going to sell licenses to those programs under different terms and Digium does. Could you please go and read Marks (or my) mail again ? I don't know about the FSF (haven't checked) but with Digium you are keeping your copyright, but giving Digium a non-excluse, non-revocable license to your changes. Kind regards, Martin List-Petersen ___ 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) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 03:16:13PM +0100, Soren Rathje wrote: Hang on... What line pair do you use on the phone; 1+4 or 2+3 ?? I believe the correct pair to use should be 2+3. It's the middle pair. I assume that's 2+3 on an RJ connector ? ___ 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] H323/*/IAX - Firewall - IAX/*/H323
Hi, -Original Message- I have two H323 Polycom video conference system with a Linux firewall Iptables in the middle. I am not getting to make H323 working in this setup and I was wondering to put two * servers as a bridge to jump the firewall using IAX. The idea basically is: h323 Polycom IPTABLES VideoConference Device -- *(LAN) --- *(WAN) H323 Polycom chan_h323 chan_iax chan_h323 or chan_oh323 or chan_oh323 Question before spending some time with it... should it work ? It should, but as far as I have tested this, it won't. Someone commented to me that Video RTP is not passed through the Asterisk H323 channel driver, and therefore it won't work. I consider anything H323 a major pain, but that might just be me. Some good documentation on how this would (have to) work would be nice, but is currently lacking. Florian ___ 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) no dialtone on a TDM400P card
Thomas Andrews wrote: On Sun, Nov 14, 2004 at 03:16:13PM +0100, Soren Rathje wrote: Hang on... What line pair do you use on the phone; 1+4 or 2+3 ?? I believe the correct pair to use should be 2+3. It's the middle pair. I assume that's 2+3 on an RJ connector ? Correct.. Just for verification, do you have any green led's lit on the back of your card ?? /Soren ___ 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 190/220 dialplan strings?
Have a look at the Snom FAQ page: http://www.snom.com/faq_en.php Joris. On Nov 11, 2004, at 6:58 PM, Rich Adamson wrote: Anyone have an example dialplan string as to what is valid for these phones. Their admin manual doesn't cover it. ___ 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) no dialtone on a TDM400P card
I think I know what the problem is. I think that asterisk cannot generate dialtone because it had a problem with the soundcard. [chan_oss.so] = (OSS Console Channel Driver) Nov 14 16:35:49 WARNING[2078]: chan_oss.c:994 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Nov 14 16:35:49 WARNING[2091]: chan_oss.c:240 sound_thread: Read error on sound device: Resource temporarily unavailable I had to put this in modules.conf to get rid of the error: noload = chan_oss.so So I assume now that it's not capable of making dialtone ? Regards, Thomas ___ 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) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 03:35:06PM +0100, Soren Rathje wrote: Just for verification, do you have any green led's lit on the back of your card ?? Yes, and I have tested with a different telephone and cable that I know works. ___ 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) no dialtone on a TDM400P card
On Sun, 2004-11-14 at 15:16 +0100, Soren Rathje wrote: Hang on... What line pair do you use on the phone; 1+4 or 2+3 ?? I believe the correct pair to use should be 2+3. Just an idea because I'm also trying to get an old (pre power connection) TDM400 working, and have no dial tone. Try cat /proc/interrupts a number of times, do the interrupts on wctdm show an increase? Can you dial the extension from another? Watch out mine rings on the rack but nothing from the phone then if I try again it locks completely and requires a complete power reset. -- Dave Cotton [EMAIL PROTECTED] ___ 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] ODBC Message Waiting Indicator
Hi, everyone: I was playing with the ODBC configuration to pull sip and voicemail config info from a MSSQL2000 server. Everything works great except for the message waiting indication on the Polycom phones (all three models). If I move the sip registration info to sip.conf, the MWI starts working again. I'm just asking if this is a bug or should I poke around some more to get it to work. Any pointers in the right direction are appreciated. Thanks... -rb ___ 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) no dialtone on a TDM400P card
Hi Dave, On Sun, Nov 14, 2004 at 03:38:27PM +0100, Dave Cotton wrote: Try cat /proc/interrupts a number of times, do the interrupts on wctdm show an increase? They do! What also bothers me is that the interrupt is shared: 16661397 IO-APIC-level ohci1394, wctdm I have no idea what ohci1394 is. I don't have any infra-red devices connected, but I assume this (Intel) motherboard has support, hence this driver ?? Can you dial the extension from another? Watch out mine rings on the rack but nothing from the phone then if I try again it locks completely and requires a complete power reset. Sorry to be so dumb, but how would I do that ? I only have one FXS module. Or is it possible to simulate a call from the *CLI console ? Thanks, Thomas ___ 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] manager api: how to handle failed calls
Hello,
Comments inline..
The question is how to correctly handle failed calls.
In my application I want to make hundreds of outgoing calls automatically.
When the callee pick up the phone he gets a playback message and give an
acknowledge by means of dtmf code.
I make use of manager command originate, something like
Action:originate
channel: ZAP/g1/
Variable:X|Y|Z
extension: test
the extension test is something like
[test]
exten s,1 , wait ()
exten s, 2 , answer ()
exten s, 3 playback(XX)
The problem is since I don't use the application dial inside the extension
I cannot get any value from
DIALSTATUS or HANGUPCAUSE variable
I tried several strategies:
1)
change the logic and use local pseudo channel
In the originate command if I use channel: local/[EMAIL PROTECTED]/n
where test1 is:
[test1]
exten = _.,1,Dial(ZAP/g1/g${EXTEN})
exten = _.,2,NoOp( 2 HANGUPCAUSE is ${HANGUPCAUSE})
exten = _.,3,NoOp( 2 DIALSTATUS is ${DIALSTATUS})
exten = _.,4,NoOp( number is ${number})
exten = _.,5,Hangup
I got the correct HANGUP value ( ie BUSY) but unfortunately I cannot see
the variables set on the originate command.
I wonder why not?
Maybe, (just maybe, I did not try it myself) the originate variables
are passed using asterisk CVS-HEAD and variable names prefixed with
underscore... Eg: Use variable _X instead of X in the originate
command. Let me know if it works.
Regards,
--
Nicolás Gudiño
Buenos Aires - Argentina
___
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) no dialtone on a TDM400P card
On Sun, 2004-11-14 at 16:52 +0200, Thomas Andrews wrote: They do! What also bothers me is that the interrupt is shared: 16661397 IO-APIC-level ohci1394, wctdm I have no idea what ohci1394 is. I don't have any infra-red devices connected, but I assume this (Intel) motherboard has support, hence this driver ?? Firewire, either disable it from the BIOS or move your cards around, Digium cards do not like shared interrupts. Can you dial the extension from another? Watch out mine rings on the rack but nothing from the phone then if I try again it locks completely and requires a complete power reset. Sorry to be so dumb, but how would I do that ? I only have one FXS module. Or is it possible to simulate a call from the *CLI console ? Yes, but from your earlier post you cast doubt on the sound card. Can't you use a softphone from another machine? -- Dave Cotton [EMAIL PROTECTED] ___ 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) no dialtone on a TDM400P card
Thomas Andrews wrote: I think I know what the problem is. I think that asterisk cannot generate dialtone because it had a problem with the soundcard. [chan_oss.so] = (OSS Console Channel Driver) Nov 14 16:35:49 WARNING[2078]: chan_oss.c:994 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Nov 14 16:35:49 WARNING[2091]: chan_oss.c:240 sound_thread: Read error on sound device: Resource temporarily unavailable I had to put this in modules.conf to get rid of the error: noload = chan_oss.so So I assume now that it's not capable of making dialtone ? I have noload'ed both chan_oss and chan_alsa and I still get a dialtone. OK, Excercise 1; Stop Asterisk, Stop Zaptel. (I'm using FC1 so do the equivalent for Debian) modprobe zaptel debug=1 insmod wctdm debug=1 /sbin/ztcfg Now you can tail -f /var/log/messages and see hookstate. Already at this point I get a dialtone on my FXS port. /Soren ___ 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) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 04:06:37PM +0100, Soren Rathje wrote:
modprobe zaptel debug=1
kernel: Zapata Telephony Interface Registered on major 196
insmod wctdm debug=1
kernel: Setting FXS hook state to 0 (00)
last message repeated 3 times
kernel: Registered Span 1 ('WCTDM/0') with 4 channels
kernel: Span ('WCTDM/0') is new master
kernel: Freshmaker version: 71
kernel: Freshmaker passed register test
kernel: ProSLIC on module 0, product 0, version 5
kernel: ProSLIC on module 0 seems sane.
kernel: ProSLIC on module 0 powered up to -72 volts (c2) in 20 ms
kernel: Loop current set to 20mA!
kernel: Post-leakage voltage: 25 volts
kernel: ProSLIC on module 0 powered up to -72 volts (c0) in 10 ms
kernel: Loop current set to 20mA!
kernel: Calibration Vector Regs 98 - 107:
kernel: 98: 10
kernel: 99: 11
kernel: 100: 11
kernel: 101: 0f
kernel: 102: 07
kernel: 103: 64
kernel: 104: 09
kernel: 105: d7
kernel: 106: 07
kernel: 107: 08
kernel: Init Indirect Registers completed successfully.
kernel: Proslic module 0 loop current is 20mA
kernel: Module 0: Installed -- AUTO FXS/DPO
kernel: ProSLIC on module 1, product 0, version 0
kernel: VoiceDAA System: 04
kernel: ISO-Cap is now up, line side: 03 rev 03
kernel: Module 1: Installed -- AUTO FXO (FCC mode)
kernel: ProSLIC on module 2, product 0, version 0
kernel: Module 2: Not installed
kernel: ProSLIC on module 3, product 0, version 0
kernel: Module 3: Not installed
kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
kernel: NO BATTERY on 1/2!
/sbin/ztcfg
kernel: Setting FXS hook state to 0 (00)
kernel: Registered tone zone 0 (United States / North America)
kernel: Power alarm on module 1, resetting!
last message repeated 9 times
asterisk -vvvgc
kernel: Setting FXS hook state to 0 (00)
kernel: Setting FXS hook state to 0 (00)
I don't like the look of that NO BATTERY message. What do you think
Soren ?
-Thomas
___
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] SysMaster and GPL Violation
Joe Greco wrote: I'm struggling to think of another free software project where contributed code bearing an identical GPL or BSD license would require any such additional disclaimer. How about any softwaer owned by the FSF, MySQL, SleepCat DB, QT. I can continue if you want... :-) Really? Has the FSF really lowered itself to forcing people to sign away future acquired patent/IP rights? I wonder if IBM has signed such an agreement, because that'd have an interesting effect on some of their technology patents. I'm going to have to start echoing someone else's comments here who called it software communism. I probably wouldn't have been quite that extreme, but hey, you learn something new and interesting every day. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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) no dialtone on a TDM400P card
Thomas Andrews wrote: kernel: NO BATTERY on 1/2! I don't like the look of that NO BATTERY message. What do you think Soren ? NO BATTERY applies to FXO ports and says that Span 1/Card 2 does not receive power from the line, i.e. it is not plugged into the wall socket. (if I read the source correctly) BTW. Does the hookstate change change is you lift the handset ?? /Soren ___ 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 setup for BudgeTone
Wilson Pickett wrote: I'd like to know what's most reliable configuration for BudgeTone 101 in snip The .16 firmware is beta and it has been found to work poorly for several people, including me. I went back to .5.11 I would try to check that first Exactly, .16 has several bugs like message button not working, but .5.11 has a *nasty* bug with not-reregistering after the timeout period, which leads to phone not ringing on incoming calls - you have to power cycle the phone to get it working. ___ 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) no dialtone on a TDM400P card
- Original Message - From: Thomas Andrews [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 5:58 AM Subject: Re: [Asterisk-Users] (newbie) no dialtone on a TDM400P card | On Sun, Nov 14, 2004 at 11:46:15AM +0100, Soren Rathje wrote: | | Can you post your actual configuration ? | | /etc/zaptel.conf snip| | /etc/asterisk/zapata.conf | | [trunkgroups] snip | echocancel=yes | echocancelwhenbridged=yes | rxgain=0.0 | txgain=0.0 | group=1 | callgroup=1 | pickupgroup=1 | immediate=no snip try changing the immediate parameter in your Zapata.conf to immediate=dialtone greg ___ 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] SysMaster and GPL Violation
Joe Greco wrote: The GPL is fundamentally flawed in that it's never been functionally tested and challenged in court, and many IP lawyers believe that there are challenges that it would not survive. The fact that some lawyers may have found further legal loopholes to exploit is not shocking, given the holes in the current implementation. Actually, this is not true. The GPL was tested in a Germen court and survived very well thank you very much. I'm not so worried about courts where a straightforward reading of a license may be interpreted without many complications by an impartial judge. (I apologize for having forgotten that large parts of the rest of the world have a sane legal system. Look at us, we finally got rid of Ashcroft...) I'm much more interested in the U.S. system, where case law often has an unexpected and interesting effect on rulings, and frequently the party with more money to throw at a problem can win anyways. IOW, I wouldn't want IBM to try breaking the GPL in a courtroom, because I believe there'd be a large chance that they'd find a way to succeed. But this is not the most improtant point. The important point is this: The target of a good license (or any legal document for that matter) is not to survive in court. The purpose of a good license is to be so iron clad clear that it never ever gets into court in the first place. Well, there, that's the BSD license for you. Short. Sweet. Ironclad. That's *not* the GPL, which is a myriad maze of twisty turns and various requirements and obligations, all of which represent attack vectors against the litigants and against the license itself. Until they've all been tested in court, I'm not really convinced that it is ironclad. And this, my friend, is something the GPL has done *very well*. Mostly because people have been afraid to bring it to court, because their IP lawyers are staring at it in horror. Things like the preamble are completely stupid, because it talks about the goals of the license. A court is allowed to consider that additional data when evaluating a case, and if there were to be a conflict between that and the actual license terms, it is ambiguous (and up to the court) which of the preamble and the terms would actually win out. Yuck. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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
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 By the way, is anyone actually working on this? I would contribute to a bounty for solid US BRI support within Asterisk (preferably under FreeBSD, but I can deal with Linux if I absolutely have to). ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 04:36:21PM +0100, Soren Rathje wrote: NO BATTERY applies to FXO ports and says that Span 1/Card 2 does not receive power from the line, i.e. it is not plugged into the wall socket. (if I read the source correctly) ok. I connected it to the PABX and I got this so I assume that ports ok Nov 14 17:49:30 zorg kernel: 63761 Polarity reversed (0 - -1) Nov 14 17:49:30 zorg kernel: BATTERY on 1/2 (-)! BTW. Does the hookstate change change is you lift the handset ?? In /var/log/messages ? nothing happens as far as I can see. ___ 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 setup for BudgeTone
On Sun, 14 Nov 2004, Vahan Yerkanian wrote: Exactly, .16 has several bugs like message button not working, but .5.11 has a *nasty* bug with not-reregistering after the timeout period, which leads to phone not ringing on incoming calls - you have to power cycle the phone to get it working. Urk. I'm about to deploy 70 phones at a client and was intending to use .5.11. Can't say I've noticed this problem in testing. What is the current blessed and recommended version then? Steve ___ 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) no dialtone on a TDM400P card
Hi Dave, On Sun, Nov 14, 2004 at 04:05:04PM +0100, Dave Cotton wrote: Firewire, either disable it from the BIOS or move your cards around, Digium cards do not like shared interrupts. Yes firewire :) I couldn't disable it in the BIOS, so I took your advice and swapped cards. Now it's not shared: 22:1043486 IO-APIC-level wctdm The interrupt count is still steadily increasing. (and asterisk isn't running at the moment.) Yes, but from your earlier post you cast doubt on the sound card. Can't you use a softphone from another machine? If there is something I can install on either linux or windows I'm happy to try. What would you suggest as the easiest softpone to install ? Thanks, Thomas ___ 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] SysMaster and GPL Violation
It is also being used by IBM against SCO. And so if IBM attorneys think it's good, there's good chance it is. Actually my comment was not a serious this is why reply, it was intended as humerous reply to this silly discussion. (As there really is no problem with the soundness of the GPL.) Nor do I give a heck what any layman say on the subject as THAT is the real joke. FSF has designed and used the GPL Very Very effectively without needing to go to court, for years. Against numerous violators. Which is really the way it should be. (So tight that other lawyers don't even bother to challange it in court.) It was designed by some very competent license attorneys and has been acknowledged as a very good license by other outstanding attorneys. Of which I don't see one single one on this list. Listening to discussions about law by people who are not layers, which at that would be practicing in the appropriate areas, is like townspeople getting together shooting the shit. It's keeps them busy and entertained, and sometimes rallied up over nothing. Since this is an area which seem to keep peoples imagination going on forever maybe someone should start a small server (Yahoo offers this) to discuss the GPL. Should be very busy and entertaining. Having long since gotten bored with this thread I only dipped in to indicate the futility of this discussion. The thread started out with a honest attempt to put attention on someone that appeared to be GPL violator. Digium put an end to the discussion but the thread refused to die. If someone thinks he has found a valid problem with the GPL why not DO something about it and send off an email to FSF. These discussions can at this point only result in upsetting people who buy into arbitraries conjugated by laymen. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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 setup for BudgeTone
[EMAIL PROTECTED] wrote: On Sun, 14 Nov 2004, Vahan Yerkanian wrote: Exactly, .16 has several bugs like message button not working, but .5.11 has a *nasty* bug with not-reregistering after the timeout period, which leads to phone not ringing on incoming calls - you have to power cycle the phone to get it working. Urk. I'm about to deploy 70 phones at a client and was intending to use .5.11. Can't say I've noticed this problem in testing. What is the current blessed and recommended version then? Steve .5.11 ___ 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] 3 - TDM31B Card Installation Difficulty
Hi, o I purchased 3 TDM31B cards and fixed them in my computer (in 3 PCI slots) o I downloaded the latest Zaptel source from CVS, compiled it and loaded modules zaptel.o and wctdm.o. o I successfully configured them from /etc/zaptel.conf as shown in the information below. ztcfg returned no errors - see the report below. o I successfully configured /etc/asterisk/zapata.conf (see info below). o I configured an X-Lite phone to test with an analog phone plugged into one of the channels. The problems are: o I cannot make a call from the analog phone (Saachi phone, KX-T3223) connected to one of the FXS ports. When I pick up the receiver, I hear the dialtone but when I press the buttons, asterisk seems not to get the numbers dialled, both using pulse and touch tone dialling. o I can call the analog phone from X-Lite however on receiving, I cannot hear much voice. What I hear is choppy sound corresponding to whatever I say from the analog side. When someone speaks from the X-lite side, nothing is heard from the analog phone. o There are three FXS ports where there is no dialtone - but the phone is actually powered - I can hear touch tone / pulse when I dial. o There are three FXS ports that give neither power nor dialtone. What could the problem be? Any help will be highly appreciated. Please find below abit of information I thought may be useful. Please let me know if more is needed. EXTRA INFORMATION - linux:/usr/src/new # uname -a Linux linux 2.4.20-4GB #1 Mon Mar 17 17:54:44 UTC 2003 i686 unknown unknown GNU/Linux linux:/usr/src/new # linux:/usr/src/new # ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXO Loopstart (Default) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Slaves: 02) Channel 03: FXO Loopstart (Default) (Slaves: 03) Channel 04: FXS Loopstart (Default) (Slaves: 04) Channel 05: FXO Loopstart (Default) (Slaves: 05) Channel 06: FXO Loopstart (Default) (Slaves: 06) Channel 07: FXO Loopstart (Default) (Slaves: 07) Channel 08: FXS Loopstart (Default) (Slaves: 08) Channel 09: FXO Loopstart (Default) (Slaves: 09) Channel 10: FXO Loopstart (Default) (Slaves: 10) Channel 11: FXO Loopstart (Default) (Slaves: 11) Channel 12: FXS Loopstart (Default) (Slaves: 12) 12 channels configured. linux:/usr/src/new # /etc/zaptel.conf: fxols=1-3 fxols=5-7 fxols=9-11 fxsls=4,8,12 loadzone = us defaultzone=us /etc/asterisk/zapata.conf: [channels] signalling=fxo_ls echocancel=16 echocancelwhenbridged=yes is in milliseconds pulsedial=yes group=1 context=default callprogress=yes busydetect=1 busycount=7 relaxdtmf=yes channel = 9-11 channel = 1-3 channel = 5-7 signalling=fxs_ls group=2 context=incoming channel= 4,8,12 Gerald. ___ 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] Does Music On Hold not work on Debian???
Hi all, Is there anyone who has music on hold working on a Debian 2.6 kernel, Asterisk 1.0.1 and mpg123 0.59r, all installed with apt-get? Whatever I do I keep getting a WARNING[1130572720]: Unable to start music on hold (class 'default') on channel SIP/softel1-15b2 with no additional info. I can use mpg123 on the command prompt to listen to mp3 files using any account I can think. Groeten, Joost Kraaijeveld Askesis B.V. Molukkenstraat 14 6524NB Nijmegen tel: 024-3888063 / 06-51855277 fax: 024-3608416 e-mail: [EMAIL PROTECTED] web: www.askesis.nl ___ 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) no dialtone on a TDM400P card
Thomas Andrews wrote: On Sun, Nov 14, 2004 at 04:36:21PM +0100, Soren Rathje wrote: NO BATTERY applies to FXO ports and says that Span 1/Card 2 does not receive power from the line, i.e. it is not plugged into the wall socket. (if I read the source correctly) ok. I connected it to the PABX and I got this so I assume that ports ok Nov 14 17:49:30 zorg kernel: 63761 Polarity reversed (0 - -1) Nov 14 17:49:30 zorg kernel: BATTERY on 1/2 (-)! BTW. Does the hookstate change change is you lift the handset ?? In /var/log/messages ? nothing happens as far as I can see. May I suggest you call the nearest medicine man and have him drive out the gremlin... Or, look for contact problems in the sockets/connectors, you may have a faulty FXS module since the FXO module and the base card seems to function as expected. /Soren ___ 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] Simple Question
Hi there, I was wondering if youd be able to answer a question for me. I want to run an asterix system in my house. My main goal is for it to pick up my landline (via a modem) and then have a push button system i.e. push one for luke push two for johnetc and then divert it to the desired voip phone which will run off my Ethernet lan Is this quite simple to set up and can I attach asterix to my landline via a standard modem? In the future I want to use a service provider so I can have a tru internet phone Many thanks Luke Sheldrick ___ 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] Asterisk and Digium
After seing things readily get out of hand on some subjects I offer this data: How do you bring down a group like Asterisk? You split it up. You create friction and fractions : ) within the group. Now you have the group fighting itself. Anyone who has a valid concern about f.ex. the license Digium has, should take good care in how he spreads that view. I see people who spread this kind of misinformation as a threat to the group. What is your actual intention? And what effects are you creating? If you are continuing creating friction in a group that don't have a problem with the licensing in the first place, then your intentions must be to bring down this group. And even if that's not your intention, that's the direction of your actions. People should be aware that Asterisk DOES pose a real threat, together with the rest of the Open Source, against entrenched businesses. They have a REAL good motivation not to let this cat out of the box. Some people don't really understand what they are doing and help undermining the group by pushing angles and views that breaks up the unity of the group. For example the programmers that are contributing code to Asterisk do so of free will. They have each one of them agreed to the licensing with Digium. If you don't want your code inserted with the main code you don't need it. ANYONE WHO IS NOT CONTRIBUTING CODE BUT WHO IS SPREADING THE WORD ON HOW WRONG SOMETHING ABOUT IT IS, IS UNDERMINING THE GROUP! And most likely, that is their intentions too. This may seem harsh and unfriendly but is nevertheless true. Engaging in a discussion, beyond simply pointing to the FACTS, is aiding such a person. If someone have an honest concern with such issues, they should pursue that in a manner that was not destructive to the group. Why unsettle people just because you don't yet know if you even have a valid point? Get it validated with the proper sources. Check with an attorney and or FSF. Share the result with Digium. Whom, if you found a proper problem, would act to resolve it. If Digium refused to deal with this new problem, then and only then would it be proper to inform the group with ALL the data, so they can educate themselves and see if They care. It should be a CLEAN FACTUAL MESSAGE. Crying generalities or arbitraries does not help anyone as it cannot be acted upon. They are simply destructive or at best a waste of peoples time. A percentage of the population seem bent on being more destructive than helpful. They seem unable to do something without causing more damage than help. We have all seen such people in our lives. Let's not help such people keep a foothold in this group! -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] SysMaster and GPL Violation
Martin List-Petersen wrote: You may or may not be aware that to contribute code to FSF owned and maintained software one needs to assign copyright to the FSF in much the same way one is required to assign copyrights to Digium. True, I don't think the FSF are going to sell licenses to those programs under different terms and Digium does. Could you please go and read Marks (or my) mail again ? I don't know about the FSF (haven't checked) but with Digium you are keeping your copyright, but giving Digium a non-excluse, non-revocable license to your changes. I stand corrected. AFAIK the FSF is a full copyright assigment. BTW, I actually signed and faxed the Digium disclaimer and already contributed code under it thinking it was a copyright assigment. I guess I just trust Mark, which in the end - this is what all this is about. Gilad ___ 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] 3 - TDM31B Card Installation Difficulty
Begumisa Gerald M wrote: o I can call the analog phone from X-Lite however on receiving, I cannot hear much voice. What I hear is choppy sound corresponding to whatever I say from the analog side. When someone speaks from the X-lite side, nothing is heard from the analog phone. If you call from X-Lite to the demo menus can you hear them clearly (no choppy sound)? Given the problems you are having this might point to a bad TDM100P card. I recently had to swap out a TDM100P card (rev H) for a replacement (rev G) card because the card apparently wasn't supplying correct timing for Asterisk. This even affected X-Lite to demo calls where the FXO and FXS modules weren't even being used. The way to check is to switch to the ztdummy driver instead of the TDM100P driver and see if the X-Lite - demo calls become clear. I don't know if was a defective card or a REV H issue, but now at least working. The other symptoms in my case were no dialtone through the FXS card and the FXO card not answering incoming calls. Steve ___ 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: Top posting
Whatever. I find it frankly more annoying to have people bottom post. I use Outlook Express for my mail (as do millions of others), and the way OE formats it's mail lends itself to top posting.When you bottom post, I need to scroll way down the message to see your response, while when you top post I can see the response right away. If I want to see the source message *then* I'll scroll down, but chances are I've already been reading the thread so this isn't necessary. Professional? That's a matter of opinion, I don't think it's any less professional to top post, it's purely a question of what's convenient for different readers. Besides, as has already been commented on before, people should just be happy that everyone's willing to spend their time offering their advice on this forum rather than being concerned about how their message is formatted... just my 2 cents... Paul - Original Message - From: Tracy R Reed [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 4:15 AM Subject: Re: [Asterisk-Users] Re: Top posting On Fri, Nov 12, 2004 at 06:57:05PM -, Kevin Walsh spake thusly: up properly. There is no excuse at all for lazily top-posting. As a businessman I also see it as a matter of professionalism. I see top posting and not trimming etc as just unprofessional. I regularly do get poorly formatted emails with no trimming and top posting and such emails always strike me as unprofessional and amature. To some degree email is not all that unlike traditional written communications. You would not send a client such a poorly formatted letter. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig ___ 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] 3 - TDM31B Card Installation Difficulty
Hi Steve, If you call from X-Lite to the demo menus can you hear them clearly (no choppy sound)? Actually I can't - the sound is still choppy! Interesting. When I unload the zaptel and wctdm modules the problem goes away (I can hear the demo files quite clearly from the X-Lite phone). Given the problems you are having this might point to a bad TDM100P card. Mmm. I have a spare one. I'll replace the one that doesn't give dialtone and see what happens. Thanks alot Steve. I'll fix the card and let you know what happens. Rgds, Gerald. ___ 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
By the way, is anyone actually working on this? I would contribute to a bounty for solid US BRI support within Asterisk (preferably under FreeBSD, but I can deal with Linux if I absolutely have to). I'm going to try to assist in this also. ;) bkw ___ 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] 3 - TDM31B Card Installation Difficulty
Begumisa Gerald M wrote: Mmm. I have a spare one. I'll replace the one that doesn't give dialtone and see what happens. I'm not jumping to any conclusions, but please note the revision of the card you're taking out and the rev of the card you're putting in. In my case the modules on the card were perfectly fine - it's the backbone card itself that supplies the timing and had the problem. Thanks alot Steve. I'll fix the card and let you know what happens. I'm very much a newbie myself, but I seem to be (only) a couple of days ahead of you :-) Rgds, Gerald. Steve ___ 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 setup for BudgeTone
Vahan Yerkanian a écrit : Wilson Pickett wrote: I'd like to know what's most reliable configuration for BudgeTone 101 in snip The .16 firmware is beta and it has been found to work poorly for several people, including me. I went back to .5.11 I would try to check that first Exactly, .16 has several bugs like message button not working, but .5.11 has a *nasty* bug with not-reregistering after the timeout period, which leads to phone not ringing on incoming calls - you have to power cycle the phone to get it working. Well, 1.0.5.16 is the official version on the grandstream site: http://www.grandstream.com/y-downloads.htm. I only installed it last friday, so I'm not sure it is better or worse now. I was using 1.0.5.11 before, and was not aware of the non-reregistering problem, which would explain why the phone would not ring. Now using the static IP the phone no longer need to register, so I may safely go back to 1.0.5.11, right? Thanks, Jean-Denis ___ 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] SysMaster and GPL Violation (lets think before we jump)
Hey All, Isn't it possible that part of the commercial licenses that is offered is that you (the buyer) are not required to advertise, disclose, or even admit that your products offerings are based on an open source project? What other reason would one have for buying a commercial license for an OS piece of software? ~c ___ 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] Voicemail shorter then (ex) 2sec - don't accept
In which configuration file I can specify that I don't want to accept messages for example shorter then 2sec. ? I've looked in voicemail.conf but I couldn't find any setting that will support this option. In most cases message shorter then 2 or 3sec will not contain any message and I don't want system to record them and sending an email to me. -- #Joseph ___ 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 setup for BudgeTone
Now using the static IP the phone no longer need to register, so I may safely go back to 1.0.5.11, right? Yes, this is exactly what I did. Since my ISP gave me a fixed ip, I haven't bothered with registration so didn't notice if there were problems. .16 wouldn't even completely load for me and others had this problem as well. My motto with Budgetone is whatever works! ___ 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] Dial Plan Pattern Matching
Hi,
We are trying to figure out how to block certain calls via the Dial
Plan. For example we want to block any calls to XXX5551212. We tried
the simple approach below but it did not work. The second line gets
picked up when we dial 1301212.
exten = _1XXX5551212,1,Congestion
exten = _1305XXX,1,Goto(pri4,${EXTEN},1)
Any tips?
--
Andres
Network Admin
http://www.telesip.net
___
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] SendText
Hello, I was trying to use SendText to send a message to an extension, but it seems as if the message is being sent to the caller instead of the callee... e.g.: exten = 123, 1, SendText(hello world) Does anyone have any suggestion on how to override the behavior? Many thanks, Alex ___ 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] spandsp problem
Steve Underwood wrote: Hi Michael, What happens with 0.0.2pre4? For most people that version gives better results than 0.0.1k. It seems to fix most of the quirks people have had. spandsp says we have more pages to send. The remote fax machine sends a disconnect message and hangs up. It sounds like the remote fax machine is at fault. Does it always fail with one machine, and succeed with another? Some machines can be set to receive a maximum of X pages. Could it be soemthing like that. Try to get 0.0.2 working. It works well enough for other people than it is about to loose its pre status. With 0.0.2pre4, the new HP5510 is able to receive a multiple page fax from spandsp. However, the HP3150 does not connect. I'm not getting the debug messages that I got with 0.0.1k. How can I enable these messages? Also, I'm running asterisk-1.0.2. Should I be using HEAD with -0.0.2? Thanks, Mike -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ 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] Re: Dial Plan Pattern Matching
Andres [EMAIL PROTECTED] writes:
We are trying to figure out how to block certain calls via the Dial
Plan. For example we want to block any calls to XXX5551212. We tried
the simple approach below but it did not work. The second line gets
picked up when we dial 1301212.
exten = _1XXX5551212,1,Congestion
exten = _1305XXX,1,Goto(pri4,${EXTEN},1)
Split them up. Place each in its own context, and include those
contexts in a higher context, in the order you want them evaluated.
-tih
--
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
___
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] H323/*/IAX - Firewall - IAX/*/H323
I use Polycom video conferencing systems through both Smoothwall and Astaro Security linux with no problems or special settings what so ever. They are both using the h323 NAT and connection tracker modules for IP Tables. If you have those in your firewall you should be all set. As far as user Asterisk and IAX to bridge it. Asterisk wont pass Video in its H323 only audio. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Isamar Maia Sent: Sunday, November 14, 2004 8:21 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H323/*/IAX - Firewall - IAX/*/H323 Hi Folks, I have two H323 Polycom video conference system with a Linux firewall Iptables in the middle. I am not getting to make H323 working in this setup and I was wondering to put two * servers as a bridge to jump the firewall using IAX. The idea basically is: h323 Polycom IPTABLES VideoConference Device -- *(LAN) --- *(WAN) H323 Polycom chan_h323 chan_iax chan_h323 or chan_oh323 or chan_oh323 Question before spending some time with it... should it work ? Thanks, Isamar ___ 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] Dial Plan Pattern Matching
y
On Sun, 14 Nov 2004, Andres wrote:
exten = _1XXX5551212,1,Congestion
exten = _1305XXX,1,Goto(pri4,${EXTEN},1)
Any tips?
Sure - you need to give Asterisk a hint about priority, like so:
[area-305]
exten = _1305XXX,1,Goto(pri4,${EXTEN},1)
[default]
include = area-305
exten = _1XXX5551212,1,Congestion
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Dial Plan Pattern Matching
Nice...Thanks a lot Tom!
Andres
Network Admin
http://www.telesip.net
Tom Ivar Helbekkmo wrote:
Andres [EMAIL PROTECTED] writes:
We are trying to figure out how to block certain calls via the Dial
Plan. For example we want to block any calls to XXX5551212. We tried
the simple approach below but it did not work. The second line gets
picked up when we dial 1301212.
exten = _1XXX5551212,1,Congestion
exten = _1305XXX,1,Goto(pri4,${EXTEN},1)
Split them up. Place each in its own context, and include those
contexts in a higher context, in the order you want them evaluated.
-tih
___
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 to sip gw
El dom, 14-11-2004 a las 19:30, alex chua escribi: hi AS we have the sip running live may be you can consider to visit http://www.yestalk.net Nice. Now, do you know how to forward my incoming calls to a SIP through an ISDN ISDNLink Asustek card? Thanks regards alex --- FuturaHost.Com Lists [EMAIL PROTECTED] wrote: Hello Im trying to have my normal incoming calls automatically forwarded to my SIP phone, or even better, directly to a given number thorough my SIP service provider. Example: Im visiting the office in Argentina or Spain, someone call to our office in Italy (a 'normal' PSTN call), then the Asterisk forward the call, thorough SIP, to the 'normal' PSTN number of the office in Argentina or Spain. What minimum hardware i need? I have Linux, some ISDN cards and some SIP (Budgetone 102) phones. Thanks for the help out there -- Pablo Povarchik Quality Colocation and Dedicated Servers services Colocation facilities include Fremont California, London UK and Trento Italy +--- FuturaHost.Com - Industrial Business Class ISP + | Web Hosting - Dedicated Servers - Colocation | [EMAIL PROTECTED] - http://futurahost.com/ - (+39) 0461 592710 | Get a high quality full cabinet with 5Mbps full burst included |for only 700/month, availability also in London +-+ ___ 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 __ Do You Yahoo!? Download the latest ringtones, games, and more! http://sg.mobile.yahoo.com -- Pablo Povarchik Quality Colocation and Dedicated Servers services Colocation facilities include Fremont California, London UK and Trento Italy +--- FuturaHost.Com - Industrial Business Class ISP + | Web Hosting - Dedicated Servers - Colocation | [EMAIL PROTECTED] - http://futurahost.com/ - (+39) 0461 592710 | Get a high quality full cabinet with 5Mbps full burst included |for only 700/month, availability also in London +-+ ___ 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] Cisco ATA and G729
Hi * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a00801e0eb0.html Kido --- Dzajro ___ 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 to sip gw
May be my post wasn't gentle enough? May be colisters are expecting some more detailed info about my config? Sincerely, my need is not critical, in fact im simply playing with this, but i think a merit some answer? Thanks Pablo El dom, 14-11-2004 a las 00:50, FuturaHost.Com Lists escribió: Hello Im trying to have my normal incoming calls automatically forwarded to my SIP phone, or even better, directly to a given number thorough my SIP service provider. Example: Im visiting the office in Argentina or Spain, someone call to our office in Italy (a 'normal' PSTN call), then the Asterisk forward the call, thorough SIP, to the 'normal' PSTN number of the office in Argentina or Spain. What minimum hardware i need? I have Linux, some ISDN cards and some SIP (Budgetone 102) phones. Thanks for the help out there -- Pablo Povarchik Quality Colocation and Dedicated Servers services Colocation facilities include Fremont California, London UK and Trento Italy +--- FuturaHost.Com - Industrial Business Class ISP + | Web Hosting - Dedicated Servers - Colocation | [EMAIL PROTECTED] - http://futurahost.com/ - (+39) 0461 592710 | Get a high quality full cabinet with 5Mbps full burst included |for only ¤700/month, availability also in London +-+ ___ 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] SuperValetPark Call Disconnect
After a call is retrieved with the command SuperValetUnparkCall, I am unable to either re-park the call or transfer the call. Has anyone else experienced this situation? Thanks, Kevin ___ 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) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 05:42:36PM +0100, Soren Rathje wrote: May I suggest you call the nearest medicine man and have him drive out the gremlin... Local tradition dictates that I slaughter the whitest goat :) Or, look for contact problems in the sockets/connectors, you may have a faulty FXS module since the FXO module and the base card seems to function as expected. I agree. I looped the fxo back to the fxs to see if battery voltage was being supplied by the fxs: it looks fine because /var/log/messages shows the change of battery state when I (un)plug it, so at least the connectors are ok. I'm going to get hold of the supplier and see if he can test the module for me. Thanks so much for your help Soren. I have *really* appreciated it! Regards, Thomas ___ 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] SysMaster and GPL Violation (lets think before we
Hey All, Isn't it possible that part of the commercial licenses that is offered is that you (the buyer) are not required to advertise, disclose, or even admit that your products offerings are based on an open source project? What other reason would one have for buying a commercial license for an OS piece of software? In this industry? Lots. Let's start with linking it to a non-GPL- compatible codec, move on to linking it with a propietary configuration and management system, and end up at creatively finding a reason to fund the development of an open source software project while simultaneously obtaining a licensing model for your company that doesn't make lawyers cringe. That's three good reasons that have nothing to do with it, and I didn't even think much about it. There are plenty of reasonable reasons that one might do it. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Asterisk update
What is the easiest way to update a CVS-HEAD? I remember seeing a post a while back about an update script that updated everything, but I cannot find it now? ___ 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] MacOS/x softphone and g729a
So being at the IETF in DC last week was fun - and the first real test of my * set up. (Yay! for softphones and SIP) One issue, though - is there *really* no soft phone for the Mac with G.729A codec support? 64kbps UDP was 'patchy in places' in a very congested IP network. I spoke with Xten whilst over in the US and there was a bit of confusion over G729A and X-PRO on OSx. One sales person said that it shipped with it, so i bought it, only to find that it didn't come with it. They've been good about it, but not forthcoming with if/when/why/how G729A on os/x will happen. I'd willingly part with about 50ukp for a commercial product - a 8kbps codec on the desktop really is worth *something*, just not the 10,000 * 50usd that Xtel seem to want (!) - [I assume that they are just trying to cover programmer costs for implementing a new version and getting the necessary license, but, still, 'sheesh'!] The other alternative, I guess, is to take my Zyxel on the road with me (ho ho ho) ___ 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 update
http://www.szmidt.org/asterisk/asterisk-update.sh - Original Message - From: Henry Devito [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 2:30 PM Subject: [Asterisk-Users] Asterisk update What is the easiest way to update a CVS-HEAD? I remember seeing a post a while back about an update script that updated everything, but I cannot find it now? ___ 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] Asterisk update
Thanks, I was just getting ready to reply to cancel the message. I found it right after I posted. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, November 14, 2004 1:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk update http://www.szmidt.org/asterisk/asterisk-update.sh - Original Message - From: Henry Devito [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 2:30 PM Subject: [Asterisk-Users] Asterisk update What is the easiest way to update a CVS-HEAD? I remember seeing a post a while back about an update script that updated everything, but I cannot find it now? ___ 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 ___ 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] Asterisk using the wrong peer in sip.conf
I'm having a problem with Asterisk choosing the wrong peer entry from the sip.conf file. Based off the debug below Asterisk should see the message coming from TNT3 (see the sip.conf below) and not from SER_FAX (which it shows in the debug below Found peer 'SER_FAX'). Based off what I have here it seems to me that since the From is 198.88.216.30 it should match with the TNT3 entry in my sip.conf which is what I want it to do, instead it is matching with the SIP proxy that is proxying it the SIP message. Is there a way to get Asterisk to lookup based of the originator of the INVITE instead of by who last proxied it the INVITE? Basically here is my setup: 198.88.216.84 = SER 198.88.216.30 = TNT3 (My PSTN Gateway) 198.88.216.85 = Asterisk TNT3 --- SER Proxy --- Asterisk - Here is the debug from Asterisk: Sip read: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Record-Route: sip:[EMAIL PROTECTED]:5060;ftag=46cae657-1bf8d0c3-1ed858c6;lr=on To: sip:[EMAIL PROTECTED]:5060;user=phone From: sip:[EMAIL PROTECTED]:5060;user=phone;tag=46cae657-1bf8d0c3-1ed858c 6 Remote-Party-Id: sip:[EMAIL PROTECTED]:5060;user=phone;screen=yes;id-type=subscriber ;party=calling;privacy=off Proxy-Require: privacy Call-ID: [EMAIL PROTECTED] CSeq: 93145626 INVITE Via: SIP/2.0/UDP 198.88.216.84;branch=z9hG4bK4aca.dc7b3842.0 Via: SIP/2.0/UDP 198.88.216.30:5060;rport=5060 Max-Forwards: 69 Contact: sip:[EMAIL PROTECTED]:5060;user=phone Supported: replaces Supported: 100rel Content-Type: application/sdp Content-Length: 272 v=0 o=TNT3 469291203 469291203 IN IP4 198.88.216.30 s=Session SDP c=IN IP4 198.88.216.30 t=0 0 m=audio 44518 RTP/AVP 18 0 8 96 a=silenceSupp:off a=ecan:b on g168 a=rtpmap:96 telephone-event/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 16 headers, 12 lines Using latest request as basis request Sending to 198.88.216.84 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 96 Peer audio RTP is at port 198.88.216.30:44518 Found description format telephone-event Found description format PCMA Found description format PCMU Found description format G729 Capabilities: us - 0x10c(ULAW|ALAW|G729A), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x10c(ULAW|ALAW|G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'SER_FAX' Looking for 44 in FROM_PSTN list_route: hop: sip:[EMAIL PROTECTED]:5060;ftag=46cae657-1bf8d0c3-1ed858c6;lr=on list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 198.88.216.84;branch=z9hG4bK4aca.dc7b3842.0;received=198.88.216.84;rport=506 0 Via: SIP/2.0/UDP 198.88.216.30:5060 From: sip:[EMAIL PROTECTED]:5060;user=phone;tag=46cae657-1bf8d0c3-1ed858c 6 To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as62f171b3 Call-ID: [EMAIL PROTECTED] CSeq: 93145626 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 - Here is my sip.conf: [general] port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind to context=FROM_PSTN ; Default for incoming calls rtptimeout=30 ; Terminate call if 60 seconds of no RTP activity canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw [SER] type=friend host=198.88.216.84 port=5060 qualify=no nat=yes disallow=all allow=g729 allow=ulaw allow=alaw [SER_FAX] type=friend host=198.88.216.84 port=5060 qualify=no nat=yes disallow=all allow=ulaw [TNT3] type=friend host=198.88.216.30 port=5060 dtmfmode=rfc2833 qualify=no canreinvite=no nat=no context=FROM_PSTN deny=0.0.0.0 permit=198.88.216.30/255.255.255.255 disallow=all allow=g729 allow=ulaw allow=alaw Michael Shuler ___ 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] Point to Point VOIP
How are the two offices connected? In terms of an Asterisk solution, at a high level you are looking at an Asterisk machine on each end, each of which is connected to the existing office phone system or the local PSTN via TDM cards (or T1/E1 with channel banks, etc). Without more details it's hard to be more specific, but you should get an idea there. Greg - Original Message - From: Jacob Arthur [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November 07, 2004 10:59 PM Subject: [Asterisk-Users] Point to Point VOIP I am looking for a setup something like the following. I have two offices, one located in the US and one in Australia. I would like to implement a solution whereby I would install a gateway in each of the two offices. When calls are made to a few numbers in the US, the calls would be routed over the gateway to the one in Australia. The gateway in Australia would dial out to a pre-defined number/set of numbers to complete the call. What is the minimum hardware/software configuration I would need to complete this sort of setup? I am relatively new to the concepts behind VOIP, so any help would be greatly appreciated. Is there anyone with a similar setup to this that has any suggestions/tips? Thanks, Jacob Jacob Arthur, MCP ATS [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ 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] Best setup for BudgeTone
Jean-Denis Girard wrote: Well, 1.0.5.16 is the official version on the grandstream site: http://www.grandstream.com/y-downloads.htm. I only installed it last friday, so I'm not sure it is better or worse now. I was using 1.0.5.11 before, and was not aware of the non-reregistering problem, which would explain why the phone would not ring. Now using the static IP the phone no longer need to register, so I may safely go back to 1.0.5.11, right? Unfortunately that's not correct. Try this (with static IP): Set up the phone's re-register delay to a say 5 minutes. Save Reboot. Call once. Phone rings. Wait 6 minutes. Call again. Phone doesn't ring and if you have something other than Dial() for that extension, say voicemail, it activates. My solution was to put a large value for the timeout, and reboot reboot reboot. This is true with 1.0.5.11, not sure about 1.0.5.16, as I rolled back from it as the message button wasn't working, sending only 'INVITE:' instead of the full SIP message to call the voicemail extension. ___ 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] SysMaster and GPL Violation (lets think before we
Joe Greco wrote: Hey All, Isn't it possible that part of the commercial licenses that is offered is that you (the buyer) are not required to advertise, disclose, or even admit that your products offerings are based on an open source project? What other reason would one have for buying a commercial license for an OS piece of software? It is no doubt a Nazy plot. Can this thread please end now? Gilad ___ 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] SysMaster and GPL Violation
I just trust Mark, which in the end - this is what all this is about. Gilad The most sensible comment made in this thread. ___ 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] problem with zyxel prestige 2002
Hi I tried to use Zyxel Prestige 2002 VoIP Analog Telephone Adapter with asterisk. Device registers both phones and i can call out. But incoming calls are not working. Asterisk - sip show peers shows zyxel, zyxel web interfce shows that devices are registered. But when i do incoming call to zyxel, phones do not ring and if voicemail is configured, calls go directly to voicemail. Any suggestions ? Mihkel ___ 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 using the wrong peer in sip.conf
First, please do not use friends. It will confuse you. We receive calls from type=user and send calls to type=peer (that is the general idea, anyway... :-) For an incoming call, we first match on users based on the username part of the From: sip address. So if the call comes from sip:[EMAIL PROTECTED], we'll match on oej. If we can't find a matching user, we will match on peers based on IP address, which is your problem. Make sure you have a type=user with the correct user name and you should be all set. /O ___ 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] ERROR: retrans_pkt: Maximum retries exceeded on call
Can anybody provide any input on an error:
retrans_pkt: Maximum retries exceeded on call
My extension 11 will not ring. I can dial out without any problem but
when I call in it goes straight to voicemail box.
Goto (office-open,s,1)
-- Executing Wait(SIP/pstn-spa3k-71e2, 2) in new stack
Nov 14 14:37:15 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
seqno 102 (Request)
Goto (office-open,1,1)
-- Executing Dial(SIP/pstn-spa3k-71e2, SIP/11|20) in new stack
-- Called 11
Nov 14 14:37:33 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
seqno 102 (Request)
My sip.conf.
[11] ; outgoing call on fxs port
type=friend
host=dynamic
context=internal
secret=
port=5064 ; port on Line 1
username=demo1
mailbox=11
dtmfmode=rfc2833
canreinvite=no
nat=no
[pstn-spa3k] ; incoming calls on FXO port
type=friend
secret=spa3k
username=demo2
host=dynamic
port=5065 ; port on Pstn line
dtmfmode=rfc2833
nat=no
context=incoming
extension.conf
[globals]
sales_support=SIP/11
;accounting=SIP/12
pstn-spa3k=10.0.0.150:5065 ;the pstn line on the spa3k
[incoming]
; First, let's do the holidays
exten = 888,1,GotoIfTime(*|*|1|jan?holiday,s,1)
exten = 888,2,GotoIfTime(*|*|1|jul?holiday,s,1)
exten = 888,3,GotoIfTime(*|*|11|nov?holiday,s,1)
exten = 888,4,GotoIfTime(17:00-23:59|*|24|dec?holiday,s,1)
exten = 888,5,GotoIfTime(*|*|25|dec?holiday,s,1)
exten = 888,6,GotoIfTime(17:00-23:59|*|31|dec?holiday,s,1)
; these are the days we're open
exten = 888,7,GotoIfTime(10:00-17:59|mon-sun|*|*?office-open,s,1)
; if we're not open, we're closed (duh!)
exten = 888,8,Goto(office-closed,s,1)
[office-open]
exten = s,1,Wait(2)
exten = s,2,Answer()
exten = s,3,BackGround(welcome) ; Play a congratulatory message
exten = s,4,Goto(1,1)
exten = *,1,Goto(s,3)
exten = 1,1,Dial(${sales_support},20)
exten = 1,2,Voicemail(u11) ; Right to voicemail
exten = 1,3,Hangup()
--
#Joseph
___
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] asterisk ser setup consulting needed
my company is deploying a metaswitch vp3510 and we are looking to add some additional services via voip. ive messed around briefly with asterisk (had it working a time or two with some cisco 79xx phones and such) but dont have the time required to really get in and dig around on how to get it working for what we currently need. Once its setup, Im sure I can maintain the system, but Ive had trouble with the extensions.conf file getting it to work. here are our needs: 1. voicemail integration using asterisk with the metaswitch - the metaswitch supports sip and i have configured it to recognize my asterisk server as the voicemail server. - i have asterisk installed on a box (suse 9.0) but havent done anything beyond the compile of asterisk and a few modules 2. sip proxy using ser for sip clients to connect to and place calls through the metaswitch vp3510. anyone interested, please contact me via email with your hourly rate. -- Chad Whitten ___ 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] Marconi Sys X/TE410P configuration
On Wed, 10 Nov 2004 00:19:58 +, Steve Kennedy [EMAIL PROTECTED] wrote: Hi Steve, Hi, name rings a bell for some reason ? Hmm, yes, I have the same feeling about your name. Ever been to borstal? (Just kidding...) I would have answered sooner but last week turned nasty when someone delivered 1.2M calls in 10 minutes when they had forecast 30k calls over 3 weeks; messy, very messy - always carry a spatula! Couldn't find any config data on the lists, though did pick out the EuroISDN stuff. Hopefully it's just a switch misconfiguration. Here are some configs from a production server using a TE4XXP card connected to a Marconi System X switch via ETSI Q.931 PRI (ISDN110): http://www.comgate.tv/Marconi_Star/zaptel.conf http://www.comgate.tv/Marconi_Star/zapata.conf I hope that these files help. If you still have problems why don't you let us take a look at the configs you are using? Regards Darren -- Darren Storer Comgate Telco|Internet|Broadcast ___ 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] SysMaster and GPL Violation (lets think before we
Hi folks, Might I propose a new mailing list ?? Asterisk-bitch Thus discussions such as the one with this topic could be moved to it rather than clutter up an already very busy list. All those in favour ? . ___ 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] SIP Packets stuck in queue
This is weird. None of my SIP clients will authenticate to my development system. They just timeout trying. When I look at the network stats I see the packets are queued but they dont go anywhere. Could this be a problem in the ethernet driver or chan_sip?? Running Asterisk v1.0.11 (CVS Stable) == [EMAIL PROTECTED] root]# netstat -ua Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State udp10600 0 *:5060 *:* == -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. - http://www.linuxsys.com Located in beautiful Tallahassee, Florida Office 850-224-5737 Office 850-575-7213 Mobile 850-294-7567 ___ 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] (no subject)
Is there any problem with the compilation of channel h323 in asterisk 1.0.2? I get the following error. /asterisk-1.0.2/channels/h323# make g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I../../include -I/root/pwlib/include/ptlib/unix -I/root/pwlib/include -I/root/openh323/include -Wno-missing-prototypes -Wno-missing-declarations ast_h323.cpp cc1plus: warning: -Wno-missing-declarations is valid for C/ObjC but not for C++ ast_h323.cpp: In member function `void MyH323Connection::SendUserInputTone(char, unsigned int)': ast_h323.cpp:725: error: invalid conversion from `char' to `const char*' ast_h323.cpp: In member function `virtual void MyH323Connection::OnUserInputTone(char, unsigned int, unsigned int, unsigned int)': ast_h323.cpp:735: error: invalid conversion from `char' to `const char*' ast_h323.cpp: In member function `virtual void MyH323Connection::OnUserInputString(const PString)': ast_h323.cpp:746: error: invalid conversion from `char' to `const char*' /usr/include/c++/3.3.4/istream: At top level: chan_h323.h:31: warning: `sockaddr_in bindaddr' defined but not used make: *** [ast_h323.o] Error 1 And this is the Makefile # include the Makefile of OpenH323 ifndef OPENH323DIR OPENH323DIR=/root/openh323 endif ifndef PWLIBDIR PWLIBDIR=/root/pwlib endif ifndef ASTERISKDIR ASTERISKDIR=/usr/local/asterisk endif ifndef ASTETCDIR ASTETCDIR=/usr/local/asterisk/etc/asterisk endif _ http://www.mailbox.gr ÁðïêôÞóôå äùñåÜí ôï ìïíáäéêü óáò e-mail. http://www.thesuperweb.gr Website ìå ÁóöáëÝò Controlpanel áðü 6 Euro êáé äþñï ôï domain óáò! ___ 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] Snom 220 Problem
Hi all, I'experiencing problems with my Snom 220 telephones. I bought 2 of them, and as I receivedthe phones I started them up and I used the web interface to configure them. Both telephones, after first boot, are network unreachable: they don't answer to ping requests and they, obviously, don't work. After talking with Snom support I tried to move then in a completely new network environment and I had the same behaviour, so I askedto my reseller to replace my telephones. I just received my two new devices, and I plugged the firs one, configured it with web interface and, after first reboot the same orrible behaviour !!! I can say that I am not configuring advanced options, nothing more than language and SIP settings, but maybe I'm doing something wrong. I can say that I own also Cisco 7940 and Budgetone phones, I got them working in a couple ofhours.Anyone experienced the same problem ? Thanks Marco Vescovi --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.793 / Virus Database: 537 - Release Date: 10/11/2004 ___ 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
RE: [Asterisk-Users] Re: Top posting
Paul Fielding [EMAIL PROTECTED] lazily top-posted: Whatever. I find it frankly more annoying to have people bottom post. I use Outlook Express for my mail (as do millions of others), and the way OE formats it's mail lends itself to top posting. As you seem to find it difficult to move the cursor on your own, perhaps this utility will help: http://home.in.tum.de/~jain/software/oe-quotefix/ You could install it to fix your broken mail reader - if it's not too much effort. When you bottom post, I need to scroll way down the message to see your response The effort involved is clearly too much for you to handle. Are you really that lazy? If I want to see the source message *then* I'll scroll down, but chances are I've already been reading the thread so this isn't necessary. Your laziness will make life difficult for people who find your followups in a future Google search. Just because you've read the entire thread, doesn't mean that someone else will have done the same next year. Then again, the chance of you posting useful information for someone to find in Google does seem to be a bit remote. just my 2 cents That might be all your time is worth. Others get paid a little more than that. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: getting callerid from spa3k to asterisk
Randy Bush [EMAIL PROTECTED] wrote: if i have two sip contexts for my spa3k, on inbound and one outbound, e.g. [spa3k-in] type=friend host=dynamic port=5061 auth=md5 secret=pfui qualify=1000 canreinvite=yes context=ext-in42 [spa3k-out] type=peer auth=md5 secret=pfui username=outpass fromuser=outpass host=spa3k.bogus.com port=5061 nat=no canreinvite=yes context=ext-in42 and the spa3k's PSTN / Subscriber Information / User ID: = spack-in, the incoming connection from spa3k to * is being routed to the spa3k-out context, not the spa3-in context. see appended. i suspect this is a bug in * 1.0.1. i found the problem, or at least a work-around. if i reverse the order of the above two sip contexts, the incoming call is properly routed to the spa3k-in sip context as opposed to the wrong one, spa3k-out. my guess is that * is traversing a list and taking the first context which has the ip address and port it wants without checking the context name against the name which was received over the wire. so it depends on what order the contexts are inserted in the list. Yes, but it's not a bug. :-) You have type=friend and type=peer. Change friend to user for the incoming definition. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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: Top posting
Dude...you seriously need either to relax, or remove yourself from this and all mailing lists if it is that bothersome to you. I consciously changed my Thunderbird formatting to insert replies at the top. I prefer it. So do many others. Get over it, and yourself. Jesus... Greg Kevin Walsh wrote: Paul Fielding [EMAIL PROTECTED] lazily top-posted: Whatever. I find it frankly more annoying to have people bottom post. I use Outlook Express for my mail (as do millions of others), and the way OE formats it's mail lends itself to top posting. As you seem to find it difficult to move the cursor on your own, perhaps this utility will help: http://home.in.tum.de/~jain/software/oe-quotefix/ You could install it to fix your broken mail reader - if it's not too much effort. When you bottom post, I need to scroll way down the message to see your response The effort involved is clearly too much for you to handle. Are you really that lazy? If I want to see the source message *then* I'll scroll down, but chances are I've already been reading the thread so this isn't necessary. Your laziness will make life difficult for people who find your followups in a future Google search. Just because you've read the entire thread, doesn't mean that someone else will have done the same next year. Then again, the chance of you posting useful information for someone to find in Google does seem to be a bit remote. just my 2 cents That might be all your time is worth. Others get paid a little more than that. ___ 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 Patch issues
I revered back. Problem solved, hope they dont suspend me :) Jerry Geis wrote: After the broadvoice patch I am getting busy messages also on call in. Is anyone else experiencing a lot of busy signals after this patch? ie Broadvoice becomes disassociated with asterisk.. __ NOD32 1.922 (20041112) Information __ This message was checked by NOD32 antivirus system. http://www.nod32.com ___ 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 __ NOD32 1.922 (20041112) Information __ This message was checked by NOD32 antivirus system. http://www.nod32.com ___ 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: Top posting
Kevin Walsh wrote: As you seem to find it difficult to move the cursor on your own, perhaps this utility will help: http://home.in.tum.de/~jain/software/oe-quotefix/ Wow! If only all people using outlook would use this! Seriously people, even if you are pissed off at Kevin, this link is really good. It gives you some of the features from Thunderbird in Outlook Express. Check it out. Now we just need an OE plugin that will only let people post in plaintext and we're sorted! -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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: Top posting
On Nov 15, 2004, at 00:18, Matt Riddell wrote: Now we just need an OE plugin that will only let people post in plaintext and we're sorted! ... or to mandate legible quote structuring and plain-text in the list charter ;-) -- Mark/ ___ 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: Top posting
Feel free to debate and argue, but to litter your response with personal insults to me simply tells everyone that your response is worth even less than my measely 2 cents. If you want to make it personal, take it to email rather than this forum so the others don't have to waste their time with it... Sorry everyone, this is the last public comment I'll make on the issue... :( regards, Paul - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 4:45 PM Subject: RE: [Asterisk-Users] Re: Top posting Paul Fielding [EMAIL PROTECTED] lazily top-posted: Whatever. I find it frankly more annoying to have people bottom post. I use Outlook Express for my mail (as do millions of others), and the way OE formats it's mail lends itself to top posting. As you seem to find it difficult to move the cursor on your own, perhaps this utility will help: http://home.in.tum.de/~jain/software/oe-quotefix/ You could install it to fix your broken mail reader - if it's not too much effort. When you bottom post, I need to scroll way down the message to see your response The effort involved is clearly too much for you to handle. Are you really that lazy? If I want to see the source message *then* I'll scroll down, but chances are I've already been reading the thread so this isn't necessary. Your laziness will make life difficult for people who find your followups in a future Google search. Just because you've read the entire thread, doesn't mean that someone else will have done the same next year. Then again, the chance of you posting useful information for someone to find in Google does seem to be a bit remote. just my 2 cents That might be all your time is worth. Others get paid a little more than that. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Top posting
On 11-Nov-2004, George Gardiner wrote: So that I can understand the almost religious fervour on this point could someone please explain to me why top posting is so hated!! Hopefully this isn't just further fanning the flames, but here's the page I like to point people to which does a great job discussing both views: http://mailformat.dan.info/quoting/ I realize that I've tipped my hand on my own preference simply by bottom- posting my reply. :) -- David McNett [EMAIL PROTECTED] http://slacker.com/~nugget/ ___ 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] SysMaster and GPL Violation (lets think before we
Only if we can move the top-post discussion there too Gary wrote: Hi folks, Might I propose a new mailing list ?? Asterisk-bitch Thus discussions such as the one with this topic could be moved to it rather than clutter up an already very busy list. All those in favour ? Only if we can move the top-post discussion there too ___ 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] Hangup Phone
Hi, Asterisk does not hangup automatically after caller leave a voicemail message and hangup.! Asterisk does not hangup automatically after the caller hangup in the Auto attendant menu system! What variables should I change to have * automatically hangup if the caller hangup? Right now, I have a variable set to a maximum of 60 seconds to hangup. All comments are greatly appreciated. Darly ___ 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] SysMaster and GPL Violation (lets think before we
On Sun, 14 Nov 2004 17:30:23 -0800, Bruce Ferrell wrote: Only if we can move the top-post discussion there too Gary wrote: Hi folks, Might I propose a new mailing list ?? Asterisk-bitch Thus discussions such as the one with this topic could be moved to it rather than clutter up an already very busy list. All those in favour ? Only if we can move the top-post discussion there too Sure, any of those types of debate. In fact, i just wished people would NOT use the list for debates !! . ___ 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] SysMaster and GPL Violation (lets think before we
Does that mean I have to put my posts there when I'm PMSing? On Sun, 14 Nov 2004 17:30:23 -0800, Bruce Ferrell wrote: Only if we can move the top-post discussion there too Gary wrote: Hi folks, Might I propose a new mailing list ?? Asterisk-bitch Thus discussions such as the one with this topic could be moved to it rather than clutter up an already very busy list. All those in favour ? Only if we can move the top-post discussion there too Sure, any of those types of debate. In fact, i just wished people would NOT use the list for debates !! . ___ 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] Polycom 500 software?
On Mon, 8 Nov 2004, Rich Adamson wrote: That's not right. New phones come loaded with the current relevant firmware. Upgraded f/w is only available to/from certified resellers. Or look on the wiki for where it is freely available. The two new 500's that were purchased from a Polycom reseller actually came with no firmware installed at all; only the bootloader (or whatever its called). Someone on this list pointed me to a souce for downloading the sip image, and now I've got the phone running, but it won't register with *. Not sure what the registration problem is as yet, but doing a sip debug indicates the registration failure. I double checked the Auth UserID and Password and they appear to be correct. Seems others on the list have had the same issue, but I've not found any responses resolving the problem as yet. Anyone have any suggestions? I just purchased a pair of SoundPoint IP 300's for use with Asterisk. Came with the SIP firmware loaded on it. Registered fine against Asterisk. I was able to register my phone at Polycom's website and setup an account to download firmware and manuals, but did not find the latest firmware there, only an older release. Found the firmware from a link on the Wiki. Loaded it, and it solved a couple of minor bugs. Phones work good. No major issues. However, I'm a little pissed that Polycom advertises the phones as supporting Power Over Ethernet, when in fact the phone has no POE chipset in it. You need to purchase an additional $40 cable if you want to plug it into a POE setup. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 to sip gw
El dom, 14-11-2004 a las 23:54, Martin List-Petersen escribió: On Sun, 2004-11-14 at 19:00, FuturaHost.Com Lists wrote: El dom, 14-11-2004 a las 19:30, alex chua escribió: hi AS we have the sip running live may be you can consider to visit http://www.yestalk.net Nice. Now, do you know how to forward my incoming calls to a SIP through an ISDN ISDNLink Asustek card? If that is the internal card, i actually think it's HFC-S based card, which is supported either by isdn4linux/chan_modem_i4l or bristuff (the latter is preferred, because it gives you real zaptel devices and H*** of a lot better sound quality). Check the logo on the chipset. It should be the towers of the Cologne Domchurch the chip says HFC-S or alike. Be aware though, that either of these solution only support European DSS1 (or isdn4linux would also implement 1TR6). That was very helpfull info. Thanks Pablo Kind regards, Martin List-Petersen -- Pablo Povarchik Quality Colocation and Dedicated Servers services Colocation facilities include Fremont California, London UK and Trento Italy +--- FuturaHost.Com - Industrial Business Class ISP + | Web Hosting - Dedicated Servers - Colocation | [EMAIL PROTECTED] - http://futurahost.com/ - (+39) 0461 592710 | Get a high quality full cabinet with 5Mbps full burst included |for only ¤700/month, availability also in London +-+ ___ 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] Service Providers With Caller ID Name??
Does anyone know any Asterisk friendly VoIP providers that offer caller id with NAME besides Broadvoice??? 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
Re: [Asterisk-Users] isdn to sip gw
On Mon, 2004-11-15 at 02:26, FuturaHost.Com Lists wrote: El dom, 14-11-2004 a las 23:54, Martin List-Petersen escribió: On Sun, 2004-11-14 at 19:00, FuturaHost.Com Lists wrote: El dom, 14-11-2004 a las 19:30, alex chua escribió: hi AS we have the sip running live may be you can consider to visit http://www.yestalk.net Nice. Now, do you know how to forward my incoming calls to a SIP through an ISDN ISDNLink Asustek card? If that is the internal card, i actually think it's HFC-S based card, which is supported either by isdn4linux/chan_modem_i4l or bristuff (the latter is preferred, because it gives you real zaptel devices and H*** of a lot better sound quality). Check the logo on the chipset. It should be the towers of the Cologne Domchurch the chip says HFC-S or alike. Be aware though, that either of these solution only support European DSS1 (or isdn4linux would also implement 1TR6). That was very helpfull info. For the US you would need to use chan_capi and a Eicon Diva Server Card. Nothing else working so far on US BRI's. /Martin ___ 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 setup for BudgeTone
I've been using 1.0.5.10 on 25 phones since August and I've only had to reboot 2 phones the entire time. Craig - Original Message - From: Vahan Yerkanian [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 15, 2004 3:51 AM Subject: Re: [Asterisk-Users] Best setup for BudgeTone Jean-Denis Girard wrote: Well, 1.0.5.16 is the official version on the grandstream site: http://www.grandstream.com/y-downloads.htm. I only installed it last friday, so I'm not sure it is better or worse now. I was using 1.0.5.11 before, and was not aware of the non-reregistering problem, which would explain why the phone would not ring. Now using the static IP the phone no longer need to register, so I may safely go back to 1.0.5.11, right? Unfortunately that's not correct. Try this (with static IP): Set up the phone's re-register delay to a say 5 minutes. Save Reboot. Call once. Phone rings. Wait 6 minutes. Call again. Phone doesn't ring and if you have something other than Dial() for that extension, say voicemail, it activates. My solution was to put a large value for the timeout, and reboot reboot reboot. This is true with 1.0.5.11, not sure about 1.0.5.16, as I rolled back from it as the message button wasn't working, sending only 'INVITE:' instead of the full SIP message to call the voicemail extension. ___ 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
[Asterisk-Users] ResponseTimeout problem
I'm trying to implement ResponseTimeout to give a customer a few extra seconds before ringing the phone. But it doesn't work or I'm doing it the wrong way. exten = s,3,BackGround(welcome) exten = s,4,ResponseTimeout,15 exten = s,5,Goto(1,1) After playing welcome message it goes straight to 1,1 and ring the phone. How do I pause for 15sec and give customer some time to enter an option? -- #Joseph ___ 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 setup for BudgeTone
Jean-Denis Girard wrote: Well, 1.0.5.16 is the official version on the grandstream site: http://www.grandstream.com/y-downloads.htm. I only installed it last friday, so I'm not sure it is better or worse now. I was using 1.0.5.11 before, and was not aware of the non-reregistering problem, which would explain why the phone would not ring. Now using the static IP the phone no longer need to register, so I may safely go back to 1.0.5.11, right? Unfortunately that's not correct. Try this (with static IP): Set up the phone's re-register delay to a say 5 minutes. Save Reboot. Call once. Phone rings. Wait 6 minutes. Call again. Phone doesn't ring and if you have something other than Dial() for that extension, say voicemail, it activates. My solution was to put a large value for the timeout, and reboot reboot reboot. This is true with 1.0.5.11, not sure about 1.0.5.16, as I rolled back from it as the message button wasn't working, sending only 'INVITE:' instead of the full SIP message to call the voicemail extension. FWIW, I just received a 100 phone and its running v1.0.5.10, and everything _seems_ to work just fine. Registers fine and has been working for several days without a miss (so far). Our limited testing has not seem any issues thus far. Rich ___ 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
