[Asterisk-Users] Re: Call queues
Hi Jeremy, What about this in extentions.conf exten = 5000,1,Dial(SIP/phone1SIP/phone2SIP/phone3,50,r) -- mvh. Hans-Henrik Andresen Jeremy Kenney [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hello I am new to asterisk I want to setup the call queues where it will ring multiple devices at the same time and send the call to the first one that is picked up. There doesn't need to be an agent login for this I don't think I just want setup so no login is required. Please help -Jeremy ___ 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] Re: MYSQL_FRIENDS and IAX problem
Hi, Are there realy no-one who can help here -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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] Re: Re: MYSQL_FRIENDS and IAX problem
hmm - this is the bad thing about open source etc. Should we make a bugreport ? or are we just doing something wrong ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- usedcanon [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] It seems that way, I asked the same question about a month ago, and no one cared to answer. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 18 July 2004 07:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem Hi, Are there realy no-one who can help here -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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 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] MYSQL_FRIENDS and IAX problem
Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from IP-ADRRESS When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. Any clue ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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] zapras - and kernel ??
Hi, I'm trying to get zapras do work, I had downloaded the pppd-source and the 2 patches. I succefull compiled and install the patched version of pppd, but got this error in message-log Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized option 'active-filter' Jul 15 11:43:57 voip1 pppd[9299]: Plugin zaptel.so loaded. Jul 15 11:43:57 voip1 pppd[9299]: Zaptel Plugin Initialized Jul 15 11:43:57 voip1 pppd[9299]: Using zaptel device 'stdin' Jul 15 11:43:57 voip1 pppd[9299]: pppd 2.4.1b2 started by root, uid 0 Jul 15 11:43:57 voip1 pppd[9299]: Zaptel device is 'stdin' Jul 15 11:43:57 voip1 pppd[9299]: Unable to put device 'stdin' into HDLC mode Jul 15 02:43:57 voip1 kernel: Zaptel: Zaptel PPP support not compiled in Jul 15 11:43:57 voip1 pppd[9299]: Exit. It's thrue, I have'nt pathced and compiled the kernel, but Can't find anything about it - no reademe. Any clue ? or any how-to for the zapras ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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 on 64bit ?
Hi, A'm about to set up a asterisk for 5000 users, and the customer had a 64bit server - can asterisk compile on that ? I will use a digium X100P for timing use will that do on a 64bit ? (I'm using SUSE91 kernel 2.6) What else ? Is it posible to have only one server for 5000 users ? I gues that it will be 5-700 sim. users only talking sip, and IAX2 to my PSTN-Gateway. The system is suposed to scale to 15000 users. I'm ready to receive input :) /Hans-Henrik Andresen ___ 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: Asterisk on 64bit ?
Hi, Thanks for your answer: Yes it works, in theory... (You *might* run into some hardcoded limits but they are usually easy to fix) I was just wondering if someone HAS it running, could tell me it WONT work at all. It'll be really interesting to see 15000 registrations on one server... Sory, misunderstanding, no - I was thinking on 5000 users per server Just a small piece of advice... try using more and (physically) distributed servers. I will do There are basicly 1000 ways to solve this and each member of this list Yea. I read on the wiki-pages that it was not good to use agi, cause of heavy load on the server, but an extensions.conf with 5000+ entrys is that good ? I would preffer agi and a very littlte and simple extensions.conf. Any experience with asterisk and 5000-15000 users ? -- mvh. Hans-Henrik Andresen ___ 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 compiling festival
Hi, I had followed the installation-guide to festival http://www.voip-info.org/wiki-Asterisk+festival+installation speech-tools compiles OK, but I got this error when compiling asterisk if I compile without the patch it compiles, but of cause did'nt work with asterisk. any clue ? /Hans-Henrik Andresen Making in directory src/modules/base ... making dependencies -- modules.cc module_support.cc parameters.cc ff.cc pos.cc p hrasify.cc word.cc postlex.cc phrinfo.cc g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend -Wno-depre cate d -I../include -I../../../src/include -I../../../../speech_tools/include modules .cc In file included from ../../../../speech_tools/include/ling_class/EST_Relation.h :43, from ../../../../speech_tools/include/EST_wave_aux.h:54, from ../../../../speech_tools/include/EST_speech_class.h:44, from ../../../../speech_tools/include/EST.h:60, from ../../../src/include/festival.h:44, from modules.cc:41: ../../../../speech_tools/include/EST_THash.h:292: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:294: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:303: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:304: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend -Wno-depre cate d -I../include -I../../../src/include -I../../../../speech_tools/include module_ support.cc In file included from ../../../../speech_tools/include/ling_class/EST_Relation.h :43, from ../../../../speech_tools/include/EST_wave_aux.h:54, from ../../../../speech_tools/include/EST_speech_class.h:44, from ../../../../speech_tools/include/EST.h:60, from ../../../src/include/module_support.h:45, from module_support.cc:41: ../../../../speech_tools/include/EST_THash.h:292: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:294: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:303: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:304: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend -Wno-depre cate d -I../include -I../../../src/include -I../../../../speech_tools/include paramet ers.cc In file included from ../../../../speech_tools/include/ling_class/EST_Relation.h :43, from ../../../../speech_tools/include/EST_wave_aux.h:54, from ../../../../speech_tools/include/EST_speech_class.h:44, from ../../../../speech_tools/include/EST.h:60, from ../../../src/include/module_support.h:45, from parameters.cc:41: ../../../../speech_tools/include/EST_THash.h:292: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:294: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:303: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:304: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend -Wno-depre cate d -I../include -I../../../src/include -I../../../../speech_tools/include ff.cc In file included from ../../../../speech_tools/include/ling_class/EST_Relation.h :43, from ../../../../speech_tools/include/EST_wave_aux.h:54, from ../../../../speech_tools/include/EST_speech_class.h:44, from ../../../../speech_tools/include/EST.h:60, from ../../../src/include/festival.h:44, from ff.cc:41: ../../../../speech_tools/include/EST_THash.h:292: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:294: warning: ` EST_TStringHashV::IPointer' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:303: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename ../../../../speech_tools/include/EST_THash.h:304: warning: ` EST_TStringHashV::IPointer_k' is implicitly a typename g++ -c -fno-implicit-templates -O3 -Wall -Wno-non-template-friend -Wno-depre cate d -I../include -I../../../src/include -I../../../../speech_tools/include pos.cc In file included from ../../../../speech_tools/include/ling_class/EST_Relation.h :43, from ../../../../speech_tools/include/EST_wave_aux.h:54, from
[Asterisk-Users] 'Answered' at wrong time.
Hi, When I make a call from my asterisk and it is passed thru another astrisk eg. iaxtel, I got 'Answered' in my astrisk, and bill-sec is start counting as soon I get connected to the other asterisk, and not if the party on the other asterisk server pick up the phone. So IF the other party are not aswering the call at all, I still get Answerd and billsec in cdr. Whats wrong ? Can I do something about it ? /HHA ___ 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] notransfer=yes but still tryin to bridged
Hi, Another one. I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get this in my logfile Attempting native bridge of [EMAIL PROTECTED]/5 and IAX2[dialout]/6 Asterisk Version is CVS-04/19/04-22:17:41 What's wrong ? I gues it has somethnig to do withe my bilsec-problem as well. /HHA ___ 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: 'Answered' at wrong time.
Problem at partner site, some perl-problem with answer-command /HHA Whats wrong ? Can I do something about it ? /HHA ___ 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 response 404 Not Found AND circuit-busy ??
I have a dlink dvg-1120s voip-router. I can make calls out from the router, but when calling the router I got -- Executing Dial(SIP/2010-b437, SIP/2021|30|r) in new stack -- Called 2021 -- Got SIP response 404 Not Found back from 62.79.78.74 -- SIP/2021-473b is circuit-busy What does this meen ? Or what can I do ? The router is behind nat, but if I put the router on the same network as asterisk it work ok /Hans-Henrik Andresen ___ 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] iax2 reload - how ?
Hi, My asterisk fails and stops after running the reload command ~20 times (I'm testing) - is this a kown problem ? Therefor I wil reload only sip, extensions and iax, it works with sip and extensions, but it seem that there are no reload for iax - or what ? -- mvh. Hans-Henrik Andresen -- Telefon for en flad 20'er - www.telefin.dk -- ___ 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: Newbie....
If you want MusicOnHold and Conferencing however, you will need one card for the timing. Why - I had used only ztdummy that works for MOH and conf. uncomment ztdummy in the makefile for zaptel and compile. /Hans-Henrik Andresen ___ 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 connection to Cisco Call Manager
Hi, At my company we have a large CCM-installation, is it possible to / how to connect between asterisk and CCM. I'm quit shure that the CCM only use Skinny. Any idea of the hardware-size for 1000 users ? /Hans-Henrik Andresen ___ 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: Re: Re: Limit on call in minuttes.
Tanks This was exatly what I needed, /Hans-Henrik Andresen Nicolas Gudino [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi Hans, http://bugs.digium.com/bug_view_page.php?bug_id=773 This patch plays a tone 40,30,20 and 10 seconds before absolutetimeout. ___ 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] Limit on call in minuttes.
Hi, I saw somewhere that it was possible to set a limit for how long time a call could be, for an extension in extension.conf. But I can't find it anymore. Can someone please help. Calls to '411' an operator may max. be 5 min. I have this in extension.conf. [shortcuts] exten = 411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) [operator] exten = 0,1,Dial(SIP/operator,30,tr) ___ 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: Help Newbie: TDM Development Kit
Did you compile the zap and lipri and installed ? /HHA app_dial.c:533 dial_exec: Unable to create channel of type 'ZAP' ___ 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] peer is UNREACHABLE when using XLITE
Hi, I have 3 friends trying to connect to my Asterisk using x-lite, all of them are using 3 dif. adsl-provider. For each of them I got this in sip.conf: disallow=all allow=ulaw allow=alaw allow=ilbc allow=g729 allow=g723.1 [seholm] type=friend secret=** auth=md5 nat=yes host=dynamic reinvite=no canreinvite=no qualify=1000 dtmfmode=inband callerid=Svend Erik Holm 60 context=sip They can all connect, but in asterisk I got this *CLI Mar 7 09:20:26 NOTICE[278546]: chan_sip.c:5846 sip_poke_noanswer: Peer 'henrikoglone' is now UNREACHABLE! And sip show peers show this seholm 83.88.89.122(D) 255.255.255.255 5060 UNREACHABLE They can make calls TO me, but of cause the pickup wont be sent to them, so asterisk shut down the channel after 5 sec. or so. If they are using sjphone it works, test with gs. analog adaptor also works. If I use a x-lite on same lan as asterisk, and if they make a VPN to my asterisk it works. (We had tried to use stun-server as well) Any clue ? /Hans-Henrik Andresen ___ 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: GS HandyTone-286 Transfer Problem, can anyone confirm?
I have no problem transfer from one GS adaptor to another GS adaptor. /Hans-Henrik Andresen Can anyone confirm that this problem exists? ___ 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: peer is UNREACHABLE when using XLITE
Hi I tried to raise it to 5000, but still unreachable. But as I wrote earlyer, for the same config, sjphone and a Grandstream 286 works. /HHA qualify=1000 If the client turns UNREACHABLE, you might want to change the qualify= setting to qualify=yes, that defaults to two seconds, instead of one second that you have here. ___ 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: Limit on call in minuttes.
Hi, Thank you, but this I cant get to work. /HHA so that should enable you to do the following: Call timeout = 20 sec Max Call Duration = 300 sec = 5 min. exten = 411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20,S(300)) however, I have not tried it yet so someone correct me if I am wrong.. ___ 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: Limit on call in minuttes.
Thank you This works, but. It just cut the line, I had hoped for some bip bip bip to remind that now your about to be disconected, is this possible as well ? /Hans-Henrik Senad Jordanovic [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] exten = 1,AbsoluteTimeout ($SECONDS) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Limit on call in minuttes.
HMM - This wont work :(
exten = 10,1,Dial(SIP/hha1,20,S(10))
exten = 10,2,VoiceMail,u10
exten = 10,102,VoiceMail,b10
Soren Rathje [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Ok, it actually works fine here..
Asterisk CVS-03/06/04-14:35:21, Copyright (C) 1999-2004 Digium.
From extensions.conf:
[pstn-out-nat]
;
ignorepat = 0
; NOT USED
exten = _0XX0X,1,Congestion
; Local eight-digit dialing accessed through trunk interface
exten = _0NXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},20,S(20))
exten = _0NXXX,2,Congestion
From * console:
-- Executing Dial(SIP/1000-4d25, Zap/1/4060|20|S(30)) in new
stack
-- Setting call duration limit to 30 seconds.
-- Called 1/4060
-- Zap/1-1 answered SIP/1000-4d25
-- Hungup 'Zap/1-1'
== Spawn extension (default, 04060, 1) exited non-zero on
'SIP/1000-4d25'
cdr_odbc: Query Successful!
-- Søren
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi,
Thank you, but this I cant get to work.
/HHA
so that should enable you to do the following:
Call timeout = 20 sec
Max Call Duration = 300 sec = 5 min.
exten =
411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20,S(300))
however, I have not tried it yet so someone correct me if I am wrong..
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[Asterisk-Users] Re: Re: Re: Limit on call in minuttes.
arhh - I did a checkout the 4th of marts - I will do a new checkout /HHA When did you checkout your version of Asterisk from CVS ?? This feature was put into CVS on the 6'th as a fix for bug #1107 but I have not seen it in v1-0_stable. ___ 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: Re: Re: Limit on call in minuttes.
What checkout name should I do ? Just asterisk ? # cd /usr/src# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot# cvs login - the password is anoncvs.# cvs checkout asterisk /HHA This is a new feature, that's why it is NOT in 1.0-stable. Only bugfixes go into -stable. New feautres - in CVS. ___ 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] danish voice
Hi, Anyone got danish voice-files who wants to share ? /Hans-Henrik Andresen ___ 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 compiling zaptel
Hi, On my Suse90-out of box I had downloaded from CVS asterisk. I'm running kernel 2.4.21-99-smp4g with 4cpu's, and the kernelsource is in /usr/src/linux Asterisk compiles with no problem. But when compiling zaptel I got this error .. zaptel.c: In function `zt_ec_chunk': zaptel.c:4981: error: parse error before unsigned zaptel.c: In function `process_timers': zaptel.c:5500: error: parse error before unsigned zaptel.c: In function `zt_timer_poll': zaptel.c:5513: error: parse error before unsigned zaptel.c: In function `zt_chan_poll': zaptel.c:5545: error: parse error before unsigned zaptel.c: In function `zt_transmit': zaptel.c:5728: error: parse error before unsigned zaptel.c: In function `zt_receive': zaptel.c:5820: error: parse error before unsigned zaptel.c:5842: error: parse error before unsigned zaptel.c:5852: error: parse error before unsigned zaptel.c:5878: error: parse error before unsigned zaptel.c:5889: error: parse error before unsigned zaptel.c:5892: error: parse error before unsigned make: *** [zaptel.o] Error 1 Any help on this ? ___ 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: Error compiling zaptel
Greate - /usr/src/linux/include/linux/version.h:6:2: #error The kernel sources in /usr/src/linux are not yet configured. /usr/src/linux/include/linux/version.h:7:2: #error Please run 'make cloneconfig make dep' in /usr/src/linux/ /usr/src/linux/include/linux/version.h:8:2: #error to get a kernel that is configured like the running kernel. /usr/src/linux/include/linux/version.h:9:2: #error Alternatively, you can copy one of the config files /usr/src/linux/include/linux/version.h:10:2: #error arch/$ARCH/defconfig.* to .config, and run /usr/src/linux/include/linux/version.h:11:2: #error 'make oldconfig make dep' to configure the kernel /usr/src/linux/include/linux/version.h:12:2: #error for that configuration. h - I done the 'make cloneconfig make dep' and do I have say that i works now - thank you. you left out the important part of the error log. I guess the compiler complains about a header file it doesn't find before throwing all those parse errors. ___ 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] Maillinglist as newsgroup ?
Hi, I was thinking if it was possible to get this list as news ? It would be much easier that 'hotmail-account' /HHA _ Scope out the new MSN Plus Internet Software optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1 ___ 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: Maillinglist as newsgroup ?
Greate - it works. Thank you /HHA http://www.gmane.org offers many mailinglists as a newsfeed. ___ 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] configuration to Grandstream via tftp
Hi, Anyone know how to set up tftp server for grandstream. I gues it should be somethink like tftpserver-dir mac-address firmware.bin config.txt Is this correct ? And how should the config-file look like. ? I had search sipphone.com but did'nt find anything. /HHA _ Rethink your business approach for the new year with the helpful tips here. http://special.msn.com/bcentral/prep04.armx ___ 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] configuration to Grandstream via tftp
Thanks. How is the directory structure ? or do you add all you phone to the one file cfg.txt and have it in the root of your tftp-dir ? /HHA Attached is the config file I send to my Grandstream. Change IP address Phone ID to suite. _ Find high-speed net deals comparison-shop your local providers here. https://broadband.msn.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] configuration to Grandstream via tftp
Thank your for the link - now I wil try it :) /Hans-Henrik Andresen This is the URL I got the config file from, http://www.plugndial.com/ it's on a link from the SipPhone URL. _ Learn how to choose, serve, and enjoy wine at Wine @ MSN. http://wine.msn.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] GS Handytone Echo-problem
Hi, Yesterday I finaly got my handytone sip adaptor. It works But when dialing to and from ISDN I got echo in both ends, I had tried diff. codecs, but then the GS wont work at all - It can do a call, but after 3 'ring' it disconnect. Any hints ? _ Scope out the new MSN Plus Internet Software optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1 ___ 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] ISDN30 - HW ?
Hi, Are there any hardware for ISDN30 ? if yes any problem with this ? is i out-of-box like ISDN2 but with 30 linies ? Do I need more than the cable from my teleprowider and a PCI-card ? /HHA _ Find high-speed net deals comparison-shop your local providers here. https://broadband.msn.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] ISDN30 - HW ?
Yes, ISDN30 ( I am assuming you are in the UK) is basically an ISDN PRI line which is basically an E1 line, so you would need to get an E100P card from Digium to be able to connect your ISDN30 into Asterisk.. I'm from Denmark (else my english would had been better:( ) As for the rest of the questions I can't reallt answer as I have never personally connected an E100P to an ISDN30 line.. many on this list have and will hopefully be able to give you more of the technical details.. later.. I'll wait to see if some one else can help. (Wipeout - nothing about the echo ?) _ Check out the coupons and bargains on MSN Offers! http://shopping.msn.com/softcontent/softcontent.aspx?scmId=1418 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP-Client for Handheld PC
Hi, Yes Telesym, xten and one more I can't remember the name of it, they are all for PPC-only. :( /HHA From: Ray Burkholder [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP-Client for Handheld PC Date: Mon, 12 Jan 2004 14:33:39 -0500 What are the ones you found for PocketPC? I guess you've looked at the Telesym site? They have a SIP flavor coming out shortly for some PDA's. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 644 6999 x2002 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans-Henrik Andresen Sent: January 12, 2004 05:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP-Client for Handheld PC Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _ Let the new MSN Premium Internet Software make the most of your high-speed experience. http://join.msn.com/?pgmarket=en-uspage=byoa/premST=1 ___ 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] Bandwidth ? + Doc + cdr
Hi, How much bandwidth do I need for 1 conversation ? I know it depends on the codecs, in X-lite I can see a codec called gsm, and the grandstream aha analog/ip converter have a codec called 721. Doc. I have found the asterisk handbook, but only a draft from marts 2003 anything newer ? Guides/howtos are welcome as well. anyone have a php interface to accounting ? /HHA _ There are now three new levels of MSN Hotmail Extra Storage! Learn more. http://join.msn.com/?pgmarket=en-uspage=hotmail/es2ST=1 ___ 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-Client for Handheld PC
Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _ Scope out the new MSN Plus Internet Software optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1 ___ 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] Policies - deny some nubers
Hi, I had asterisk installed, ISDN-adapter, some x-lite software-phones and I can call betweens the softphone- and 'normal' phones during the ISDN-card. 2 questions now 1) Is it posible to create policies, so that some SIP-users can dial ALL numbers, and some SIP-users not are allowed to dial eg. 900xxx-numbers, 30xx (mobilphones), 40(long distance) and if possible on time basic- so that from 18.00-0800 it is possible to dial all numbers ? 2) when dialing in to asterisk via ISDN get a new dialtone so it's posibble to dial all sipphones, or get voiceresponce saying press 1 to dial manager, press 2 to dial dad, and 3 to leave a message ? /Hans-Henrik _ Tired of slow downloads? Compare online deals from your local high-speed providers now. https://broadband.msn.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Policies - deny some nubers
Thank you both, I will start reading, and had already get something to work
:)
/Hans-Henrik Andresen
Look at contexts and the include statement. Read the draft handbook
linked from www.asterisk.org, support section. Or look here:
http://www.voip-info.org/wiki-Asterisk+howto+dial+plan
For the dialtone: You can very well use separate context for that with a
Background() announcement and something like
[dial_what_you_want_context]
exten = s,1,Background(enter-an-extension)
exten = _.,1,Dial(Local/${EXTEN})
Apart from that if you wish to dial out again look at/ search for DISA.
http://www.voip-info.org/wiki-Asterisk+cmd+DISA
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[Asterisk-Users] More voicemodem
Hi, I got this setup. analog phone (ext7) --- analog pbx - (ext 6 analog) voicemodem (ext 3 asterisk) ttyS0/asterisk sipphones q1: I got the voicemodem to work, but oneway only. I can talk from my analog phone, to my sipphone, but not the other way ? I know it only suppose to works in half duplex, but nothing come TO the phone. q2: From SIPphone I dial 3+ext on my analog pbx - it works :) From analog phone I dial my voicemodem (ext 6) asterisk answer and it automatic forward to one specific sipphone, how do I get a new 'dialtone' from asterisk so I can dial ANY number in asterisk ? Hope to get some hints. (I'm really new to asterisk so an exsample would be good) /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP -- PSTN via. voicemodem/soundcard.
Hans-Henrik Andresen wrote: How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? The same way you recieve videos through your fax machine.. :) HMM. greate sarcasm. I had read about a driver for asterisk for voicemodems, that why i'm asking. So if anyone had tried this, or can help hith link to documentations I will be happy. No, it can't be done.. /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
