Re: [Asterisk-Users] Bristuff / Junghanns / Customer Service

after some while (since TEI:s lost by the terminal equipment will not be reused until asterisk is restarted). Scheduled asterisk restarts every week seems to solve the problem. With other ISDN phones this problem does not occur. -- Regards, Tobias Jönsson, Lund

Re: [Asterisk-Users] zaphfc syslog flooding

hfc card working before adding more.) 5. Your hardware might have problems with the many interrupts the zaphfc card generates. Some motherboards simply cannot take care of the 8000 interrupts per second these cards generate. Good luck! -- Best Regards, Tobias Jönsson, Lund

Re: [Asterisk-Users] Time sync on PRI

it, but have not had the time. It's already there, in bristuff patches. Please encourage Digium to add Junghanns' patches to the asterisk code :) -- Best Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] PRI and echocancel

server the delays may be so short on a bridged call that the echo would not be noticable even if it is there. That is the way ordinary pstn calls work - there is usually echo cancellation only on cellular calls or long distance calls. -- Regards, Tobias Jönsson, Lund SE

Re: [Asterisk-Users] PRI and echocancel

Eric Wieling wrote: Tobias Jönsson wrote: If you need echo cancelling on the PBX PRI depends on the phones used on that PBX. If they are all digital and they do not introduce any echo you will not need any echo cancelling. The outside phone will still (usually) be analog and cause echo. So

Re: [Asterisk-Users] Bristuff ZapHFC and Loosing D-Channel

. Bristuff will activate layer 1 and layer 2 again immediately. What will happen if there comes a call (from asterisk or from ISDN network) during the short times when D-Channel is down? Will they retry or will the call be dropped? -- Regards, Tobias Jönsson, Lund

Re: [Asterisk-Users] BUSY-tone on incoming calls?

,Wait(1) exten = 123437,5,Voicemail(su21) exten = 123437,6,Hangup exten = 123437,110,SetVar(PRI_CAUSE=17) exten = 123437,111,Hangup -- Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com

Re: [Asterisk-Users] BUSY-tone on incoming calls?

On Tue, 25 Jan 2005, Peter Svensson wrote: On Tue, 25 Jan 2005, Tobias Jönsson wrote: No, PRI_CAUSE works great at least in 1.0.2, probably in the earlier 1.0 releases too. Busy() may play a busy tone to the caller instead of signalling busy so using PRI_CAUSE is much better in PRI or BRI

Re: [Asterisk-Users] Meetme conf and Shoutcast

On Sun, 16 Jan 2005, Mike wrote: We would like to know if there is a way to broadcast (in realtime) a conferance. http://www.voip-info.org/wiki-Asterisk+cmd+Ices I haven't tried it though. -- Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing

Re: [Asterisk-Users] Cannot open /dev/dsp

at your /etc/asterisk/modules.conf and be sure not to load them (noload = chan_oss.so, noload = chan_alsa.so). -- Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Authenticate or DISA?

On Tue, 16 Nov 2004, Peter Svensson wrote: On Tue, 16 Nov 2004, Tobias Jönsson wrote: On Mon, 15 Nov 2004, Jason Williams wrote: After the Authenticte why not do a Playtones(Dial) this will give dialtone The dialtone won't stop after pressing first digit then. If course you can have an X extension

Re: [Asterisk-Users] Authenticate or DISA?

be used for extension matching in further contexts. -- Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] CHANUNAVAIL = CHANUNAVAIL doesn't eval properly

then goto extension 'i' else goto priority 103. Why don't you just use the bristuff package? There is a Dial application which goes to n+201 if channel is unavailable. -- Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list [EMAIL PROTECTED] http

Re: [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()

, Tobias Jönsson, Lund SE___ 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: Compressing a dialplan

the lines above: 0462[35] 04624[0-4] 04624[5-9]XXX 04626X That is the conversation I would like a macro/script to do. What I thought about was if there are any kind of regexp compression programs or something like that. -- Regards, Tobias Jönsson, Lund

Re: [Asterisk-Users] GrandStream HT-486 ATA as VoIP Gateway

be quite confusing to europeans. -- Regards, Tobias Jönsson, Lund SE___ 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] Compressing a dialplan

give me a hint please? Regards, Tobias Jönsson, Lund SE___ 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

Re: [Asterisk-Users] Re: Compressing a dialplan

On Wed, 25 Aug 2004, Maron Kristófersson wrote: Tobias Jönsson wrote: Does anyone have a program that could be used to compress the dialplan? I have lots of numbers in a list, for example if the file contains 12300 12310 123113 12320 12330 12340 12350 12360 12370 12380 12390 they all could have

Re: [Asterisk-Users] Finding operator from ISDN signalling?

:( That information is, unfortunately, not included in signalling. In Sweden you can get the information for free (restricted number of queries) by a web service at Swedish Number Portability Administrative Center. I do not know if there is somethink similar in Norway. Regards, Tobias Jönsson, Lund SE

Re: [Asterisk-Users] call-back example

}.call) exten = h,9,System(mv /tmp/${UNIQUEID}.call /var/spool/asterisk/outgoing) Regards, Tobias Jönsson, Lund SE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] Zaptel module loading (was: Another small suggestion patch)

scripts. Regards, Tobias Jönsson, Lund SE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] call-back example

On Thu, 19 Aug 2004, Peter Svensson wrote: On Thu, 19 Aug 2004, Tobias Jönsson wrote: exten = h,2,System(echo Channel: SIP/[EMAIL PROTECTED] /tmp/${UNIQUEID}.call) exten = h,3,System(echo MaxRetries: 2 /tmp/${UNIQUEID}.call) exten = h,4,System(echo RetryTime: 60 /tmp/${UNIQUEID}.call

[Asterisk-Users] Re: Pingtel registration failing

is violating the RFC 3261 by ignoring the Contact header. As far as I understand, Asterisk will always ignore addresses in the SIP header if nat is enabled in sip.conf. Change the setting to nat=no and asterisk should follow the standard. Regards, Tobias Jönsson, Lund SE

Re: [Asterisk-Users] disable console channels

the sound card is grabbed. How can I disable those channels? Just put these lines in your modules.conf: noload = chan_alsa.so noload = chan_oss.so Regards, Tobias Jönsson, Lund SE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

Re: [Asterisk-Users] Help - is voip good for in-house calls?

, the other end will introduce echo so that the ip side will be hearing himself speaking with a small delay. I have that problem with my home BRI running zaphfc. Regards, Tobias Jönsson, Lund SE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

Re: [Asterisk-Users] No incoming audio on incoming SIP calls

-users/2004-July/055598.html and http://lists.digium.com/pipermail/asterisk-users/2004-June/051930.html. BTW, watch out for the difference between t and T options. Regards, Tobias Jönsson, Lund, Sweden ___ Asterisk-Users mailing list [EMAIL PROTECTED

Re: [Asterisk-Users] zaphfc hardware sound trouble

(in the zaptel directory) and watch the output. I still have problems with this one running zaphfc-0.1.0-RC2k on my new Intel Pentium 4 2,4 GHz system but timing is excellent on my old AMD K6-233 MHz system... Regards from Sweden, Tobias Jönsson ___ Asterisk

[Asterisk-Users] zttest never get 100% accurancy

% 99.548340% --- Results after 40 passes --- Best: 99.987793 -- Worst: 98.950195 Regards, Tobias Jönsson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] outgoing MSN on zaphfc

I should add I have pridialplan=local; I couldn't dial numbers like 9545046370544 (9545 for choosing service provider) when using pridialplan=national. Regards, Tobias Jönsson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

[Asterisk-Users] zaphfc: All DTMF tones are doubled

: DTMF (1) SUBCLASS: 1 (49) ] [Zap/5-1] [ TYPE: DTMF (1) SUBCLASS: 1 (49) ] [Zap/5-1] Regards, Tobias Jönsson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] Rejecting Calls (SIT Tone/Invalid) Across PRI

will be rejected immediately. Regards, Tobias Jönsson ___ 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] D-Channel on span 1 up/down + frame slips with zaptelBRI

I have installed two HFC PCI A-cards running zaphfc from bristuff-0.0.2, which seems to work quite fine, but I continously receive the messages D-Channel on span 1 up followed by D-Channel on span 1 down with a few seconds interval. Why is that? Bri intense debug log and configuration files below.