[Asterisk-Users] Rebooting GS phone thru sip_notify
Hello list, Does anybody managed to reboot GrandStream phone with sip notify sip_notify.conf section peer It seems that I need to send a sys-control Event but i suspect that's not enaugh my phone just answer me a CSeq: 102 NOTIFY. Cheers Laurent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] defining range of user in sip.conf
hello any one please tell me if there is a way to define a range of users in sip.conf suppose i want to create 1000 user from 500 to 5000999 with no password from thanks Kamran Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with realtime SIP
Hi Guys,We have just set up Asterisk 1.0.7 with (CVS Head) for a realtime enviorment using MySQL Asterisk Addons.I have populated the sip_buddies table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up.Does anyone have any idea what we have done? Asterisk Console Message when SIP try to login Aug 5 11:52:31 NOTICE[8941]: chan_sip.c:9518 handle_request_register: Registration from 'vinodmalani sip:[EMAIL PROTECTED]' failed for '192.168.0.112 '*CLI dial [EMAIL PROTECTED] *CLI -- Executing Dial(OSS/dsp, SIP/400) Aug 5 12:24:06 WARNING[9008]: chan_sip.c:1780 create_addr: No such host: 400 Destroying call '[EMAIL PROTECTED]'Aug 5 12:24:06 NOTICE[9008]: app_dial.c:1091 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1)-- Executing Answer(OSS/dsp, Ringing) Console call has been answered -- SIP read from 192.168.0.112:5060:--- (0 headers 0 lines) Nat keepalive --- Aug 5 12:24:19 WARNING[9008]: pbx.c:2334 __ast_pbx_run: Timeout, but no rule 't' in context 'mycontext' Hangup on console SIP DEBUG MESSAGE ( for reference )-- SIP read from 192.168.0.112:5060:REGISTER sip:192.168.0.34 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.112:5060;rport;branch=z9hG4bK46E3233E27ED47DABD0B778CE4D37C87From: vinodmalani sip:[EMAIL PROTECTED];tag=1345370993To: vinodmalani sip:[EMAIL PROTECTED] Contact: vinodmalani sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED]CSeq: 55251 REGISTER Expires: 1800Max-Forwards: 70 User-Agent: X-Lite release 1103mContent-Length: 0 --- (11 headers 0 lines)---Using latest request as basis request Sending to 192.168.0.112 : 5060 (NAT)Transmitting (NAT) to 192.168.0.112:5060 :SIP/2.0 403 ForbiddenVia: SIP/2.0/UDP 192.168.0.112:5060;branch=z9hG4bK46E3233E27ED47DABD0B778CE4D37C87;received=192.168.0.112;rport=5060From: vinodmalani sip:[EMAIL PROTECTED];tag=1345370993To: vinodmalani sip:[EMAIL PROTECTED] ;tag=as740959f2Call-ID: [EMAIL PROTECTED] CSeq: 55251 REGISTERUser-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYContact: sip:[EMAIL PROTECTED] Content-Length: 0 ---Aug 5 12:24:39 NOTICE[9008]: chan_sip.c:9518 handle_request_register: Registration from 'vinodmalani sip:[EMAIL PROTECTED]' failed for '192.168.0.112'Scheduling destruction of call ' [EMAIL PROTECTED]' in 15000 msDestroying call '[EMAIL PROTECTED] '-- SIP read from 192.168.0.112:5060: --- (0 headers 0 lines) Nat keepalive ---i am describing entire files that we have used extconfig.conf :- content[settings]sippeers = mysql,cdr,sip_buddiessipusers = mysql,cdr,sip_buddies ;sipfriends = mysql,cdr,sip_buddiesrealextension = mysql,cdr,extensions_table extensions.conf : content[general] static=no / yes ( tried with both) writeprotect=yes / no ( tried with both) [mycontext]switch = Realtime/[EMAIL PROTECTED]res_mysql.conf :- content[general] dbhost = 127.0.0.1 dbname = cdr dbuser = root dbpass = dbport = 3306 dbsock = /tmp/mysql.socksip.conf : content[general]type=friend;rtcachefriends = yes;rtcache=yesnat=yes / no ( tried with both )( tried with both with DB parameters without it, but same result of failure )localnet=192.168.0.0/255.255.255.0dbhost = 127.0.0.1 dbname = cdr dbuser = root dbpass = dbport = 3306 dbsock = /tmp/mysql.sockModules.conf[modules]autoload=yesnoload = pbx_gtkconsole.so;load = pbx_gtkconsole.sonoload = pbx_kdeconsole.so noload = app_intercom.soload = chan_modem.soload = res_musiconhold.sonoload = chan_alsa.sonoload = res_odbc.sonoload = libodbc.sonoload = pbx_wilcalu.sonoload = cdr_odbc.so load = cdr_addon_mysql.soload = chan_oss.so[global]chan_modem.so=yesthese modules1. noload = chan_alsa.so2. noload = res_odbc.so3. noload = libodbc.so4. noload = pbx_wilcalu.so 5. noload = cdr_odbc.so gave us problem when we updated CVS so we decided to block them... but even after that asterisk was wroking fine with sip.conf extensions.conf wtih static entries sip_buddeis table of mysql :- content+---+--++-+--+++--+--+-+-+---++-+-++--++---+-+-++-+++---+---++---+-+-+++---+---+-+---+ | id| name | accountcode| amaflags| callgroup| callerid | canreinvite| context | defaultip| dtmfmode| fromuser| fromdomain| host | insecure| language| mailbox| md5secret| nat| permit| deny| mask| pickupgroup| port| qualify| restrictcid| rtptimeout| rtpholdtimeout| secret | type | username| disallow| allow | musiconhold| regseconds| ipaddr| regexten| cancallforward|
Re: [Asterisk-Users] function declaration isn't a prototype
hi i gues the error is in this line include/asterisk/strings.h:232: parse error before `va_list' can anyone help me please. how can i fix this? much thnks. chris. - Original Message - From: chris To: asterisk-users@lists.digium.com Sent: Tuesday, July 26, 2005 4:01 PM Subject: [Asterisk-Users] function declaration isn't a prototype hello, i got this error when i run make after downloading asteirsk from cvs. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -Iinclude/solaris-compat -I/usr/local/ssl/include -D_REENTRANT -D_GNU_SOURCE -O6 -Wcast-align -DSOLARIS -DBUSYDETECT_MARTIN -fomit-frame-pointer -c -o term.o term.cIn file included from include/asterisk/utils.h:26, from term.c:32:include/asterisk/strings.h:232: parse error before `va_list'include/asterisk/strings.h:232: warning: function declaration isn't a prototypemake: *** [term.o] Error 1 pls advise on how i can fix this, thnks, ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving Calls from FWD Network using IAX2
in iax.conf devi anche mettere questa riga per ogni fwd: register = FWDNumber:[EMAIL PROTECTED] Bruno. kswail wrote: Hello, I am trying to setup my Asterisk box to accept calls from the FWD network. I've followed all the config advice / samples I've found on the web. Making calls to devices on the FWD network from my Asterisk box works flawlessly, but whenever I try to call my Asterisk box from a FWD client I get a busy signal, and a Call Disconnected 486 error. What's odd is that I don't see any debug info from the console (iax2 debug). I've tried forwarding UDP port 4569 to my Asterisk box and no diff. Anyone have any advice? Cheers! kswail === Here are relevant parts of my configs --- iax.conf --- register=x:[EMAIL PROTECTED] [fwd] username=x type=peer secret= qualify=yes host=iax2.fwdnet.net auth=md5 [fwd-in] type=user inkeys=freeworlddialup context=from-pstn auth=rsa === Here is output from the asterisk console as it pertains to IAX2 --- asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 65.39.205.121:4569x 00.00.00.244:4569 60 Registered --- asterisk*CLI iax2 show peers Name/UsernameHost Mask Port Status fwd/x65.39.205.121 (S) 255.255.255.255 4569 OK (15 ms) --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More questions
Hello,I
have few questions about Asterisk.I installed Asterisk from CVS on
FreeBSD and I made cvsup 2 days ago.1.I couldn't find Asterisk version
using "asterisk -V" command.How can I to find version
information?2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV
E/F (FXS)onit.I tried Asterisk CallerID feature, but unable to get
it.I tried callerid signaling V23, Bell202, DTMF, no success. Finally,
Ifound in our country (Mongolia) PSTN/Cellular provider send
FSK/ETSItype of CallerID.Is Asterisk support such type of CallerID
signaling?If no, is there any way to get it?3.I enjoyed Asterisk
most of feature until now. I registered X-Prosoftphone, SIP analog and
analog phone connected to FXS port too.There one problem is I am unable
to make outgoing call from SIP phone,softphone, analog phone through FXO
port.Following is my Asterisk
configuration:--zaptel.confloadzone=usdefaultzone=usfxsks=1fxoks=2zapata.confcontext=bellsignaling=fxs_ksgroup=1channel
= 1context=homegroup=2signalling=fxo_kschannel =
2sip.conf[]type=friendusername=;secret=host=dynamicnat=yesdefaultip=192.168.1.5context=bellreinvite=nocanreinvite=nocallerid=[EMAIL PROTECTED]allow=g729allow=g723allow=all[]type=friendusername=;secret=host=dynamicnat=yesdefaultip=192.168.1.1context=bellreinvite=nocanreinvite=nocallerid=[EMAIL PROTECTED]allow=g729allow=g723extensions.conf[bell]exten
= s,1,Waitexten = s,2,Answerexten =
s,3,Playback(greetings)exten = s,4,WaitExten; used to record
promptsexten = 205,1,Wait(2)exten =
205,2,Record(/tmp/greetings:alaw)exten = 205,3,Wait(2)exten =
205,4,Playback(/tmp/greetings)exten = 205,5,Wait(2)exten =
205,6,Hangupexten = 111,1,Dial(CONSOLE/dsp)exten =
111,2,Hangupexten = 100,1,Answerexten =
100,2,MusicOnHold()exten = 100,4,Hangupexten =
200,1,VoicemailMainexten = 300,1,Dial(Zap/2)exten =
400,1,Voicemail(9)exten = 800,1,MeetMe(100|Mp)exten =
800,2,Hangupexten = 601,1,WaitMusicOnHold(30)exten =
700,1,Dial(SIP/,20,rt)exten =
900,1,Dial(SIP/,20,rt)exten = _ZXXX,1,Answerexten =
_ZXXX,2,Dial(Zap/g1/${EXTEN})exten = _Z,1,Answerexten =
_Z,2,Dial(Zap/g1/${EXTEN})exten = _NX,1,Answerexten =
_NX,2,Dial(Zap/g1/${EXTEN})exten = _NXXX,1,Answerexten =
_NXXX,2,Dial(Zap/g1/${EXTEN})[home]exten =
s,1,Playback(greetings)exten = 100,1,Answerexten =
100,2,MusicOnHold()exten = 100,4,Hangupexten =
111,1,Dial(CONSOLE/dsp)exten = 111,4,Hangupexten =
700,1,Dial(SIP/,20,rt)exten =
900,1,Dial(SIP/,20,rt)exten = _ZXXX,1,Answerexten =
_ZXXX,2,Dial(Zap/g1/${EXTEN})exten = _Z,1,Answerexten =
_Z,2,Dial(Zap/g1/${EXTEN})exten = _NX,1,Answer;exten =
_NX,2,SetVar(TIMEOUT(AbsoluteTimeout)=10)exten =
_NX,3,Dial(Zap/g1/${EXTEN})exten = _NXXX,1,Answerexten =
_NXXX,2,Dial(Zap/g1/${EXTEN})I can to see following in
/var/log/messages when I make outgoing call.Jul 20 00:50:26 boldsoft
kernel: Zapata Telephony Interface Registeredon major 196Jul 20 00:50:26
boldsoft kernel: ZapTel device: vendor=e159 device=1subvendor=8085Jul 20
00:50:26 boldsoft kernel: wcfxo0: Wildcard X101P port0xe800-0xe8ff
mem 0xfaffe000-0xfaffefff irq 18 at device 9.0 onpci2Jul 20 00:50:26
boldsoft kernel: ZapTel Attach for wcfxo0: deviceID :0xe159Jul 20
00:50:26 boldsoft kernel: wcfxo: DAA mode is 'FCC'Jul 20 00:50:26 boldsoft
kernel: Found a Wildcard FXO: Wildcard X101PJul 20 00:50:26 boldsoft kernel:
ZapTel device loaded.Jul 20 00:50:33 boldsoft kernel: FXS device:
vendor=e159 device=1subvendor=b100Jul 20 00:50:33 boldsoft kernel:
wcfxs0: Wildcard TDM400P REV E/Fport 0xec00-0xecff mem
0xfafff000-0xfaff irq 17 at device 8.0 on pci2Jul 20 00:50:33
boldsoft kernel: FXS Attach for wcfxs0: deviceID :0xe159Jul 20 00:50:33
boldsoft kernel: Freshmaker version: 63Jul 20 00:50:33 boldsoft kernel:
Freshmaker passed register testJul 20 00:50:35 boldsoft kernel: Module 0:
Installed -- AUTO FXSJul 20 00:50:35 boldsoft kernel: ProSLIC sanity check
failedJul 20 00:50:35 boldsoft kernel: Module 1: Not installedJul 20
00:50:35 boldsoft kernel: ProSLIC sanity check failedJul 20 00:50:35
boldsoft kernel: Module 2: Not installedJul 20 00:50:35 boldsoft kernel:
ProSLIC sanity check failedJul 20 00:50:35 boldsoft kernel: Module 3: Not
installedJul 20 00:50:35 boldsoft kernel: Found a Wildcard TDM: Wildcard
TDM400PREV E/F (4 modules)Jul 20 00:50:39 boldsoft kernel: Registered
tone zone 0 (United States/ North America)Jul 21 02:36:28 boldsoft
kernel: DIAL: T345598wJul 21 02:39:43 boldsoft kernel: DIAL: T345598wJul
21 02:45:35 boldsoft kernel: DIAL: T345598wJul 21 02:45:56 boldsoft kernel:
DIAL: T99114909wJul 21 02:47:09 boldsoft kernel: DIAL: T345598wJul 21
02:47:56 boldsoft kernel: DIAL: T345595wJul 21 02:48:16 boldsoft kernel:
DIAL: T95158330wJul 21 02:48:57 boldsoft kernel: DIAL: T95158330wJul 21
02:49:20 boldsoft kernel: DIAL: T345598wJul 21
Re: [Asterisk-Users] HELP! X100P IRQ conflict w/ USB
On Fri, Aug 05, 2005 at 01:17:05PM +0800, 163 wrote: Thanks a lot for you help first. I tried to load the drivers, but failed. [EMAIL PROTECTED] voicepet-single-x100p]# /sbin/modprobe zaptel modprobe: Can't locate module zaptel [EMAIL PROTECTED] voicepet-single-x100p]# pwd /home/shengl/voicepet-single-x100p You probably need to build them first for your kernel The source tree has a target for that. Please read the docs. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IP Phone 30 VIP
Jason ha scritto: Could someone assist me in configuring this phone. It is saying in the CLI that its registered and saying its capabilities are recieved but i got no dialtone on the phone. Thanks are you using chan_skinny or chan_sccp? Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIPPeersAction class file not found in theAsterisk-java.jar file
Well thanks Stefan, for the help but when I am executing the AGI script I am
getting the errors as below:
Aug 5, 2005 3:29:44 AM net.sf.asterisk.util.impl.JavaLoggingLog info
INFO: Received connection.
Aug 5, 2005 3:29:45 AM net.sf.asterisk.util.impl.JavaLoggingLog error
SEVERE: Unable to create AGIScript instance of type HelloScript
Aug 5, 2005 3:29:45 AM net.sf.asterisk.util.impl.JavaLoggingLog error
SEVERE: No script configured for agi://65.125.224.207/bharat.agi
What does it mean by No Script configured for agi:// and can you please
tell me how do I come up with this error?
Regards,
Bharat M. Sarvan
Software Engineer - VoIP
EZZI BPO Pvt Ltd.,
PUNE.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter
Sent: Friday, August 05, 2005 6:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIPPeersAction class file not found in
theAsterisk-java.jar file
On Thu, 2005-08-04 at 18:25 +0530, Bharat M. Sarvan wrote:
I am working on Fastagi and I am making use of
Asterisk-java. But I don't find the class file for SIPPeersAction.
The SIPPeersAction is not part of Asterisk-Java 0.1, it is available in
CVS-HEAD only.
Besides that the action classes in net.sf.asterisk.manager.action can
only be used with the Manager API and not with FastAGI.
So if you want to retrieve a list of sip peers you need to do that via
the Manager API. With Asterisk 1.0.x and Asterisk-Java 0.1 you can do
this via the CommandAction. Only if you are already using Asterisk
CVS-HEAD and Asterisk-Java CVS-HEAD you can use the new SipPeerAction.
Example with CommandAction:
import java.util.Iterator;
import net.sf.asterisk.manager.ManagerConnection;
import net.sf.asterisk.manager.ManagerConnectionFactory;
import net.sf.asterisk.manager.action.CommandAction;
import net.sf.asterisk.manager.response.CommandResponse;
public class Manager
{
private ManagerConnection c;
public Manager() throws Exception
{
c = new ManagerConnectionFactory().getManagerConnection(host,
user, pass);
}
public void run() throws Exception
{
c.login();
CommandAction action;
CommandResponse response;
Iterator lineIterator;
action = new CommandAction();
action.setCommand(sip show peers);
response = (CommandResponse) c.sendAction(action);
lineIterator = response.getResult().iterator();
while (lineIterator.hasNext())
{
System.out.println(lineIterator.next());
}
c.logoff();
}
public static void main(String[] args) throws Exception
{
new Manager().run();
}
}
This produces something like:
Name/usernameHostDyn Nat ACL Mask Port
Status
1313/131310.13.0.61 D A 255.255.255.255 5061
Unmonitored
1312/131210.13.0.61 D A 255.255.255.255 5061
Unmonitored
1311/131110.13.0.61 D A 255.255.255.255 5061
Unmonitored
1310/1310(Unspecified)D A 255.255.255.255 0
Unmonitored
1303/1303(Unspecified)D N 255.255.255.255 0
Unmonitored
1302/1302(Unspecified)D A 255.255.255.255 0
Unmonitored
1301/1301(Unspecified)D A 255.255.255.255 0
Unmonitored
=Stefan
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[Asterisk-Users] PLEASE HELP: X100P/Caller ID: clidtest works, * complains [banging head]
Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 041222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a failed checksum like this:- -- Starting simple switch on 'Zap/1-1' Jul 30 16:06:14 NOTICE[9597]: callerid.c:306 callerid_feed: Caller*ID failed checksum Jul 30 16:06:15 NOTICE[9597]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:06:16 NOTICE[9597]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:06:18 NOTICE[9597]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait(Zap/1-1, 2) in new stack snip and sometimes I get an error that I _really_ don't understand:- -- Starting simple switch on 'Zap/1-1' Jul 30 16:25:02 NOTICE[9616]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:25:02 NOTICE[9616]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:25:04 ERROR[9616]: callerid.c:260 callerid_feed: fsk_serie made mylen 0 (-62) Jul 30 16:25:04 WARNING[9616]: chan_zap.c:5434 ss_thread: CallerID feed failed: Success Jul 30 16:25:04 WARNING[9616]: chan_zap.c:5476 ss_thread: CallerID returned with error on channel 'Zap/1-1' snip This seems to be a common topic in the archives! I have tried adjusting the gain to no avail. This is a Telstra (Australia) CLID service, and I have ADSL on the same line (a line filter is installed.) The fact that clidtest works suggests that the card's getting the CLID fine, but there's a problem after that. Sorry for the repeat post - I managed to post the original during the recent list 'blackout', so I guess it didn't get to many people. Any ideas would be greatly appreciated. Cheers, Jon I still haven't found a solution to this - is it possible to disable the checksum code and see what comes through? I've had a look at that piece of code, but I'm no coder, so I don't know how I'd do it. As I said before, this seems to be a common topic on this list, but there are rarely any answers to the problems. Cheers, Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I don't have option 5 in my voicemail
What do I put in voicemail.conf to let me send another user a voicemail from inside Comedian? I've CVS-HEAD, and the instructions are a bit ambiguous on the voicemaill.conf.sample. Advanced option 5 is the only on I don't have, and a very important one to have, indeed. Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco IP Phones on Asterisk: chan_sip or chan_sccp
Hello ! I 'd like to connect Cisco IP phones to *. (7940 7960) Shall I use SIP or SCCP. Which approach provides better support of features of the Cisco IP phones ? Thanks ! Johann ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PolyCom SoundPoint 300 and distinctive ring
Yes it does apply. Near the top of your sip.cfg file, you should have lines
like this:
alertInfo voIpProt.SIP.alertInfo.1.value=ring-answer
voIpProt.SIP.alertInfo.1.class=4/
alertInfo voIpProt.SIP.alertInfo.2.value=internal
voIpProt.SIP.alertInfo.2.class=5/
alertInfo voIpProt.SIP.alertInfo.3.value=doorphone
voIpProt.SIP.alertInfo.3.class=6/
(I have a few here for auto-answer, internal extension ring cadence, and a
Zap doorphone alert)
You will also have something like these toward the bottom of sip.cfg under
ringType
RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer
se.rt.4.timeout=2000 se.rt.4.ringer=7/
INTERNAL se.rt.5.name=Internal se.rt.5.type=ring se.rt.5.ringer=3/
DOORPHONE se.rt.6.name=Doorphone se.rt.6.type=ring
se.rt.6.ringer=11/
Notice the connection between the class=4 on ring-answer above and below.
Duplicate these lines if you don't have them. Place them within the
SIP/SIP section and the ringType sections respectively. Pick your
ringer values based on the ones on the IP300 menu, which gives you chirps,
stutters, and trills etc.
Whatever value you have assigned (i.e. doorphone) is the value you must have
set in the _ALERT_INFO variable when you make the Dial(SIP) command:
[doorphone]
exten = s,1,Answer ;DOORPHONE IS CALLING
exten = s,2,SetCIDName(Doorphone 1)
exten = s,3,SetCIDNum(400)
exten = s,4,SetVar(_ALERT_INFO=doorphone) ;SET ALERT-INFO TO POLYCOMS
exten = s,5,Monitor(gsm,doorphone-${TIMESTAMP},m) ;RECORD THE DOORPHONE
CALL
exten = s,6,Dial(SIP/101SIP/102SIP/104SIP/201SIP/203Zap/2r3Zap/3,22)
;RING SOME PHONES
exten = s,7,Playback(nobody-but-chickens) ; NOBODY'S HOME
exten = s,8,Hangup
Note that you need the first underscore for ALERT_INFO if you are using
CVS-HEAD.
Hope that helps!
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: David Koski [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, August 04, 2005 10:00 PM
Subject: [Asterisk-Users] PolyCom SoundPoint 300 and distinctive ring
I am looking for clues on how to configure distinctive ring for a PolyCom
SoundPoint
300. Does ALERT_INFO apply? If so, how?
Thanks,
David Koski
[EMAIL PROTECTED]
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RE: [Asterisk-Users] SIPPeersAction class file not found in theAsterisk-java.jar file
Well thanks Stefan, for the help but when I am executing the AGI script I am getting the errors as below: If you want to retrieve sip peers from Asterisk you won't do this via an AGI as I explained. You will just run the main() method of the Manager class I sent you in my last mail as an example, like: $ java -cp asterisk-java-0.1.jar:. Manager SEVERE: No script configured for agi://65.125.224.207/bharat.agi What does it mean by No Script configured for agi:// and can you please tell me how do I come up with this error? That means you when you use FastAGI (which you should NOT in this case) you failed to provide a correct fastagi-mapping.properties file on the CLASSPATH. You find more information on how to set it up correctly at http://asterisk-java.sourceforge.net/tutorial.html =Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with realtime SIP
Hi Guys, We have just set up Asterisk 1.0.7 with (CVS Head) for a realtime enviorment using MySQL Asterisk Addons. I have populated the sip_buddies table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up. Does anyone have any idea what we have done? Asterisk Console Message when SIP try to login Aug 5 11:52:31 NOTICE[8941]: chan_sip.c:9518 handle_request_register: Registration from 'vinodmalani sip:[EMAIL PROTECTED]' failed for '192.168.0.112 http://192.168.0.112' You need to have a switch in extensions.conf: switch = Realtime/[EMAIL PROTECTED] to tell asterisk to go to the database to look for the users rofile and extensions yusuf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More questions
Please stop asking the same questions over and over. On Monday 25 Jul 2005 02:46, Balgansuren.B wrote: Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using asterisk -V command. How can I to find version information? In the CLI, show version 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. Forget it. It is hideously broken. It may work, it may not work. [... I'll leave the rest to others ...] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IP Phones on Asterisk: chan_sip or chan_sccp
Johann Steinwendtner wrote: Hello ! I 'd like to connect Cisco IP phones to *. (7940 7960) Shall I use SIP or SCCP. Which approach provides better support of features of the Cisco IP phones ? SIP will cost you an extra $100 per phone to license the SIP software. But the SIP has been working for a long time with * and is gernerally quite stable. On the other hand, SCCP comes with the phone, and the phone has many more features. However chan_sccp has not been tested heavily and is likely to have a few bugs in it. I would recommend that you set it up both ways and see for yourself. The phone definitely feels nicer in sccp. -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem running Astrerisk after defined card in /etc/asterisk/zapatal.conf
I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15' Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Aug 5 10:47:29 WARNING[1076842624]: loader.c:423 load_modules: Loading module chan_zap.so failed! Here is the /proc/zaptel/1 file(seems to be correct, and log Messages too indicates the initialization is correct.): Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4 RED 1 WCT1/0/1 Clear 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear 9 WCT1/0/9 Clear 10 WCT1/0/10 Clear 11 WCT1/0/11 Clear 12 WCT1/0/12 Clear 13 WCT1/0/13 Clear 14 WCT1/0/14 Clear 15 WCT1/0/15 Clear 16 WCT1/0/16 HDLCFCS 17 WCT1/0/17 Clear 18 WCT1/0/18 Clear 19 WCT1/0/19 Clear 20 WCT1/0/20 Clear 21 WCT1/0/21 Clear 22 WCT1/0/22 Clear 23 WCT1/0/23 Clear 24 WCT1/0/24 Clear 25 WCT1/0/25 Clear 26 WCT1/0/26 Clear 27 WCT1/0/27 Clear 28 WCT1/0/28 Clear 29 WCT1/0/29 Clear 30 WCT1/0/30 Clear 31 WCT1/0/31 Clear In BERONET instructions for install the device was indicated as Span 1: TE1/0/1 "TE110P (PCI) Card 0 Span 1" HDB3/CCS/CRC4 ClockSource IRQ misses: 0 1 TE1/0/1/1 Clear (In use) ... .. /etc/zaptel file: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 /etc/asterisk/zapata.conf [channels]switchtype = euroisdnsignalling = bri_cpepridialplan = local language=itcontext=homeoverlapdial=yesusecallerid=yeshidecallerid=yescallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=noechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0callgroup=1pickupgroup=1immediate=nomusiconhold=defaultgroup=1channel = 1-15channel = 17-31 Any idea? Regards and thank you... Mauro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem running Astrerisk after defined card in /etc/asterisk/zapatal.conf
I think that you are wrong: Here is the /proc/zaptel/1 file(seems to be correct, and log Messages too indicates the initialization is correct.): Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4 RED 1 WCT1/0/1 Clear If you see your /proc/zaptel/1 you will see a RED alarm, it mean: - Failure on zaptel.conf configuration. - The cable is not connected to the Digium Card. - others issues. Mauro Zanin wrote: I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15' Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Aug 5 10:47:29 WARNING[1076842624]: loader.c:423 load_modules: Loading module chan_zap.so failed! Here is the /proc/zaptel/1 file(seems to be correct, and log Messages too indicates the initialization is correct.): Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4 RED 1 WCT1/0/1 Clear 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear 9 WCT1/0/9 Clear 10 WCT1/0/10 Clear 11 WCT1/0/11 Clear 12 WCT1/0/12 Clear 13 WCT1/0/13 Clear 14 WCT1/0/14 Clear 15 WCT1/0/15 Clear 16 WCT1/0/16 HDLCFCS 17 WCT1/0/17 Clear 18 WCT1/0/18 Clear 19 WCT1/0/19 Clear 20 WCT1/0/20 Clear 21 WCT1/0/21 Clear 22 WCT1/0/22 Clear 23 WCT1/0/23 Clear 24 WCT1/0/24 Clear 25 WCT1/0/25 Clear 26 WCT1/0/26 Clear 27 WCT1/0/27 Clear 28 WCT1/0/28 Clear 29 WCT1/0/29 Clear 30 WCT1/0/30 Clear 31 WCT1/0/31 Clear In BERONET instructions for install the device was indicated as Span 1: TE1/0/1 "TE110P (PCI) Card 0 Span 1" HDB3/CCS/CRC4 ClockSource IRQ misses: 0 1 TE1/0/1/1 Clear (In use) ... .. /etc/zaptel file: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 /etc/asterisk/zapata.conf [channels] switchtype = euroisdn signalling = bri_cpe pridialplan = local language=it context=home overlapdial=yes usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default group=1 channel = 1-15 channel = 17-31 Any idea? Regards and thank you... Mauro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with realtime SIP
Hi I already have an swtich stmt in my extensions.conf switch=Realtime/[EMAIL PROTECTED] Even i tried with one you send, but same error. please if you are using realtime do me a favour by sending all configuration you are using. Thanks vinod malani On 8/5/05, yusuf [EMAIL PROTECTED] wrote: Hi Guys,We have just set up Asterisk 1.0.7 with (CVS Head) for a realtimeenviorment using MySQL Asterisk Addons.I have populated the sip_buddies table with the same information that is came from our sip.conf, however registration seems to fail forthesoftphone we have set up.Does anyone have any idea what we have done? Asterisk Console Messagewhen SIP try to login Aug5 11:52:31 NOTICE[8941]: chan_sip.c:9518 handle_request_register:Registration from 'vinodmalani sip:[EMAIL PROTECTED]' failed for' 192.168.0.112 http://192.168.0.112'You need to have a switch in extensions.conf:switch = Realtime/[EMAIL PROTECTED]to tell asterisk to go to the database to look for the users rofile and extensionsyusuf___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone
what is the upload speed on B? Looks to me as you have bandwidth problem! Martin Kronstad wrote: Hi! Problem: I can’t hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) - Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone(Location B) I am having problems with sound, I have opened the following ports: Location A: 10 000 - 20 000 (TCP and UDP) 5060 (TCP and UDP) 8000 (TCP and UDP) Location B: 8000 (TCP and UDP) 5060 (TCP and UDP) I am using [EMAIL PROTECTED] 1.3 , and xlite as softphone. I have tried to set the softphone I have set the extention parameters(in sip.conf) to: ;; Location A [200] username=200 type=friend secret=1234 record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Location A 200 ;; Location B [201] username=201 type=friend secret=1234 record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Location B 201 My sip.conf : port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) externip=80.202.50.16 disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown language=no Best Regard Martin Kronstad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FastAGI problems
Hello! I use FastAGI very frequently [meaning 30 channels IVR is made in it] and sometimes I find, that there is a message like: Jul 29 09:38:55 VERBOSE[896] logger.c: == Auto fallthrough, channel 'Local/[EMAIL PROTECTED],2' status is 'CHANUNAVAIL' Jul 29 09:38:55 VERBOSE[893] logger.c: Channel Local/[EMAIL PROTECTED],1 was never answered. Jul 29 09:38:55 VERBOSE[896] logger.c: -- Executing DeadAGI(Local/[EMAIL PROTECTED],2, agi://127.0.0.1/callhangup ) in new stack Jul 29 09:38:55 VERBOSE[590] logger.c: -- AGI Script agi://127.0.0.1/callhangup completed, returning 0 Jul 29 09:38:55 WARNING[896] res_agi.c: Connect to 'agi://127.0.0.1/callhangup' failed: Bad file descriptor Jul 29 09:38:55 VERBOSE[896] logger.c: == Spawn extension (route-out, h, 1) exited non-zero on 'Local/[EMAIL PROTECTED] t-eeae,2' Jul 29 09:38:55 DEBUG[893] channel.c: Avoiding deadlock for 'Local/[EMAIL PROTECTED],2' Jul 29 09:38:55 DEBUG[893] channel.c: Avoiding deadlock for 'Local/[EMAIL PROTECTED],2' Jul 29 09:38:55 DEBUG[893] channel.c: Avoiding deadlock for 'Local/[EMAIL PROTECTED],2' Jul 29 09:38:55 DEBUG[893] channel.c: Avoiding deadlock for 'Local/[EMAIL PROTECTED],2' Jul 29 09:38:55 DEBUG[893] channel.c: Avoiding deadlock for 'Local/[EMAIL PROTECTED],2' Jul 29 09:38:55 DEBUG[893] channel.c: Avoiding deadlock for 'Local/[EMAIL PROTECTED],2' Jul 29 09:38:55 DEBUG[893] channel.c: Avoiding deadlock for 'Local/[EMAIL PROTECTED],2' Jul 29 09:38:55 DEBUG[893] channel.c: Avoiding deadlock for 'Local/[EMAIL PROTECTED],2' Jul 29 09:38:55 DEBUG[893] channel.c: Avoiding deadlock for 'Local/[EMAIL PROTECTED],2' Could anybody tell me what causes this 'Bad file descriptor' message? From the code I see, that it comes after the connection has been established with FastAGI server, however I don't see anything on that. This problem happens only very rarely [once/2days with continuous 30channels/8hours load]. What can cause that issue? Did anybody think about using a unix socket for communicating asterisk and the fastagi server? I know, we would lose the remote processing feature, however we can save on IP stack when AGI requested handled locally. Any idea how can I stabilize the FastAGI running? On the other side is a python threading socketserver. Thanks in advance, Tamas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Roundrobin queue strategy broken ?
Hi there, this is my queues.conf, I'm using todays CVS: [599] joinempty = yes musiconhold = default strategy = roundrobin servicelevel = 60 wrapuptime = 0 maxlen = 0 timeout=15 announce-frequency = 15 member = SIP/381 member = SIP/300 At first call 381 rings, if you hang up and call again you get the 300 ringing ... this looks more rrmemory than roundrobin, there is something wrong in my setup maybe ? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP: X100P/Caller ID: clidtest works, * complains [banging head]
You might have better luck posting this question on Asterisk-Dev (on how to disable checksum etc). On 8/5/05, Jon Whitear [EMAIL PROTECTED] wrote: Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 041222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a failed checksum like this:- -- Starting simple switch on 'Zap/1-1' Jul 30 16:06:14 NOTICE[9597]: callerid.c:306 callerid_feed: Caller*ID failed checksum Jul 30 16:06:15 NOTICE[9597]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:06:16 NOTICE[9597]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:06:18 NOTICE[9597]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait(Zap/1-1, 2) in new stack snip and sometimes I get an error that I _really_ don't understand:- -- Starting simple switch on 'Zap/1-1' Jul 30 16:25:02 NOTICE[9616]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:25:02 NOTICE[9616]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:25:04 ERROR[9616]: callerid.c:260 callerid_feed: fsk_serie made mylen 0 (-62) Jul 30 16:25:04 WARNING[9616]: chan_zap.c:5434 ss_thread: CallerID feed failed: Success Jul 30 16:25:04 WARNING[9616]: chan_zap.c:5476 ss_thread: CallerID returned with error on channel 'Zap/1-1' snip This seems to be a common topic in the archives! I have tried adjusting the gain to no avail. This is a Telstra (Australia) CLID service, and I have ADSL on the same line (a line filter is installed.) The fact that clidtest works suggests that the card's getting the CLID fine, but there's a problem after that. Sorry for the repeat post - I managed to post the original during the recent list 'blackout', so I guess it didn't get to many people. Any ideas would be greatly appreciated. Cheers, Jon I still haven't found a solution to this - is it possible to disable the checksum code and see what comes through? I've had a look at that piece of code, but I'm no coder, so I don't know how I'd do it. As I said before, this seems to be a common topic on this list, but there are rarely any answers to the problems. Cheers, Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there a right place for a include_once statement in a PHP AGI script?
Hello there, I'm new to PHP AGIs and I'm having problems with a particular script that has a include_once statement on it. If I remove that stament, the script runs until the section of the code that depends on the include and then returns. If I include that statement, the script does not seem to run at all. What shall I do? Thanks in advance, Leo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TFTP - Good or Bad?
I don't post here often but I read with interest all the postings. - I've been on a lot of mailing lists, but this one is by far the most interesting. I've been doing a lot of work with 'tftp' loading Cisco 79xx phones with firmware, configs. for asterisk, etc. And I see where a lot of folks have trouble with 'tftp', use alternate port numbers (probably to get around firewall issues), etc. - And I've even seen where some folks complain that 'tftp' is one of the 'worst' protocols on the Internet. At the end of this posting, I've included a little tid-bit on 'primary/alternate' 'tftp' servers for the Cisco 79xx phone setup. This next part is mainly for 'newbies' who are new to asterisk haven't got a clue as to what 'tftp' is. - Advanced users, geeks, etc., please disregard the next part if you want. Apologize in advance if this is boring. Going back to 'Networking 101', just exactly what is 'tftp'? - Is there any reason WHY it came into being in the first place? 'tftp' stands for 'Trivial File Transfer Protocol'. - Unlike the more popular 'ftp' protocol, 'tftp' is considered 'non-secure'. - Meaning that no login name/password challenge is require. - The 'device' (computer, phone, whatever) sends a request to the 'tftp' server for the file the server sends it. - Plain and simple. 'tftp' also uses the 'UDP' (User Datagram Protocol). - The main difference between 'UDP' and 'TCP' is that 'UDP' uses NO ERROR CORRECTION. - No 'acks' 'naks' to make sure all the packets arrive okay at the receiving end. - It's up the receiving end to make sure everything was received okay. It also makes it relatively simpler for someone on the same LAN (mostly) to fake being a tftp server for that client (or vice versa). A UDP packet is generally more predictable, so if I wanted to send the phone bogus firmware or bogus config, it would generally be easier for me than if the server has read the files using, e.g. HTTP. HTTP is simple, well-supported and supports all the file transfers operations TFTP supports. And, FWIW, there are a large number of tftp implementations (mostly in the non-linux pc arena) that have issues dealing with the last packet in a tftp transfer causing failures. (Based on about 15 years of using various tftp products as a mechanism to upgrade cisco ios's.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPManager has been released - the ultimate configuration tool for Asterisk
IPManager - Asterisk Configuration Tool has been released with IPSwitchBoard Overview IPManager is a configuration tool for Asterisk. It gives you an easy way of configuring Asterisk to perform maintenance and creation of the following: . SIP Extensions can be configured very easy with Caller ID and Voicemail . Virtual User - A user can login at any phone with an Virtual user extension, and (s)he will receive all calls at that extension, the voicemail and Called ID will be moved to that extension as well. This would be very useful if you have people sharing a phone or a person travelling between departments who need to be reached at his own number everywhere. . Queues - configure Queues and ACD groups very easily. . Extension Opening Hours - Any extension or Queue can have its own opening hours, say you want to receive calls on your office phone during office hours and then calls will be transferred to your mobile after office hours. You can always force an extension to be open or closed by dialing a code on the phone. . IVR Menus can be set up very easily, you can even attach a wav file, which will be uploaded to Asterisk and converted to gsm format automatically. . Direct Dial In - Map DDI to local extensions . Least Cost Routing - Configure which calls should use which trunks . Conferencing - setup a conference room that even outside users can join . Virtual Faxes - receives faxes and forwards them to an email account . DISA - Call this number and get a new dial tone where you can call any local extension . SIP Channels . IAX Channels All you need is a PC with Linux and Asterisk installed, and IPManager can do the rest for you in a very simple and intuitive way. You can also maintain many different configurations for different servers in IPManager and connect to them by the click of a button. You can even have IPManager configure and connect IPSwitchBoard to the server you are working on; this makes it very easy for you to support multiple servers. FREE Download: http://ipswitchboard.thorben.dk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a right place for a include_once statement in a PHP AGI script?
On Friday 05 August 2005 14:04, Leo Burd wrote: Hello there, I'm new to PHP AGIs and I'm having problems with a particular script that has a include_once statement on it. If I remove that stament, the script runs until the section of the code that depends on the include and then returns. If I include that statement, the script does not seem to run at all. What shall I do? Leo, wrap a function around whatever is in the included script, make your include_once() statement at the top of the AGI and then simply call the function at the place where it's necessary for that code to be executed. Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] USB ISDN devices
Has anyone had any luck getting USB ISDN devices to work with Asterisk? I have bought a DayTrek miniVigor 128 and would like to get it to work with CAPI or mISDN. Has anyone every successfully done something like this? Thanks, Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk registered in ser proxy
is it possible to register asterisk in a sip proxy as if it were a terminal (like a cisco ATA)? how? Thanx Jenna ;) ___ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA and a PayPhone
I've done this using SPA-2000, SPA-2000 can generate polarity reversal signal, The pay-phone detects call answer and hangup by revesal signal. also the pay-phone must be supported polarity reversal detection. Anyone got any suggestions? I need to know what piece of hardware I need (ATA preferably) that allows me to pick up an analog phone, sit idle and not get the reorder tones for at least 1 minute. I am currently using a Cisco ATA-188 and I get them at 10 seconds. I've monkeyed with every single bit of the config file and can't seem to extend or disable it. HELP!! Help is on the way:) This is quite simple to achieve on Sipura units. There is a parameter called Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2) It defines the frequencies and duration of the tone. The 10 you see near the end is the duration. Simply change it to 60 like this and you're done: Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];60(*/0/1+2) I just tried it and it works like you want it. I'm not the OP and do plan on deploying several spa3k's, is there somewhere this kind of stuff is documented for the spa's? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
ARTCP Project (was RE: [Asterisk-Users] Features you'd like to see in a GUI?)
I'd like to officially reclaim the Features in a GUI thread ;) Asterisk Hackers, Admins, and general digital phreakers of old, After careful consideration, the ARTCP project will probably have to be split into two major sections, both distros or at least maps for a system to be designed as well as the software we develop. Section 1. ARTCP Provider This version will be a distro including the designed management/enduser/billing software hooking into an asterisk RT installation using pre-set mysql schemas. The reasoning for this is that it's much easier to design a database driven php package when you know what the schema will be. Section 2. ARTCP PBX Intended for medium to large pbx's for endusers that want RealTime performance, this project will be a distro with less overall features, but more than enough to handle PBX functions and more. Possible Section 3? AI-PBX Cpanel? (name?) This version will be the same as ARTCP PBX, but not running the RT version of Asterisk. All the above sections should have: A complete branded distribution of linux, as small as possible. Bacula backup system Zabbix monitoring system Asterisk (STABLE) All Asterisk apps/modules that are required for final product PHP MySQL Apache Samba (for end user uploading music on hold files) Webmin (for end user control of system) I'd like to try and get anyone interested in contributing code work to join me in an online IRC chatroom sometime around August 17, 2005. Please reply (just please erase the [Asterisk-Users] section of your subject, otherwise it'll get trapped in the general list folder due to message moving rules) to me directly, and we'll get everyone's availability worked out. I'll take contributions of funds as well to be split across the developers, but won't look for donations before at least some cursory info has been released to show that the project is at least happening ;) Cheers all, and I hope to see interest in getting this going. Sherwood McGowan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TFTP - Good or Bad?
There is error correction in TFTP. Its done at the application layer and not the transport layer. TFTP uses two UDP ports for control and data transfer, this is probably where there are problems with NAT devices. The control connection is ; client - sport dynamic(x) - server dport 69 client asks for a file server then sends data to the client server - sport dynmaic - client dport (x) Each data packet includes a block number. When the client receives a good block it then ACKs the block. The server will then send the next block, If the server does not get an ACK for a block it will re-transmitt the block. I have seen issues with certain implementations (including busybox) where the server/client does not properly re-send blocks. To test specific TFTP implementations something like 'dummynet' (included in FreeBSD kernel) can be used to simulate poor network conditions. TFTP does have some limitations; Max file size is 32MB (due to the size of the block counter in the standard) default payload is 512 bytes (RFC 1783 introduced block size negotiation) Its can be very slow over wide area networks due to the server not sending data until previous ACK has been received. Giles - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 05, 2005 2:02 PM Subject: Re: [Asterisk-Users] TFTP - Good or Bad? I don't post here often but I read with interest all the postings. - I've been on a lot of mailing lists, but this one is by far the most interesting. I've been doing a lot of work with 'tftp' loading Cisco 79xx phones with firmware, configs. for asterisk, etc. And I see where a lot of folks have trouble with 'tftp', use alternate port numbers (probably to get around firewall issues), etc. - And I've even seen where some folks complain that 'tftp' is one of the 'worst' protocols on the Internet. At the end of this posting, I've included a little tid-bit on 'primary/alternate' 'tftp' servers for the Cisco 79xx phone setup. This next part is mainly for 'newbies' who are new to asterisk haven't got a clue as to what 'tftp' is. - Advanced users, geeks, etc., please disregard the next part if you want. Apologize in advance if this is boring. Going back to 'Networking 101', just exactly what is 'tftp'? - Is there any reason WHY it came into being in the first place? 'tftp' stands for 'Trivial File Transfer Protocol'. - Unlike the more popular 'ftp' protocol, 'tftp' is considered 'non-secure'. - Meaning that no login name/password challenge is require. - The 'device' (computer, phone, whatever) sends a request to the 'tftp' server for the file the server sends it. - Plain and simple. 'tftp' also uses the 'UDP' (User Datagram Protocol). - The main difference between 'UDP' and 'TCP' is that 'UDP' uses NO ERROR CORRECTION. - No 'acks' 'naks' to make sure all the packets arrive okay at the receiving end. - It's up the receiving end to make sure everything was received okay. It also makes it relatively simpler for someone on the same LAN (mostly) to fake being a tftp server for that client (or vice versa). A UDP packet is generally more predictable, so if I wanted to send the phone bogus firmware or bogus config, it would generally be easier for me than if the server has read the files using, e.g. HTTP. HTTP is simple, well-supported and supports all the file transfers operations TFTP supports. And, FWIW, there are a large number of tftp implementations (mostly in the non-linux pc arena) that have issues dealing with the last packet in a tftp transfer causing failures. (Based on about 15 years of using various tftp products as a mechanism to upgrade cisco ios's.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by Swift Internet, and is believed to be clean. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: ARTCP Project (was RE: [Asterisk-Users] Features you'd like to see ina GUI?)
Outta intersted - why mysql? If postgresql not a better option? I would happily contribute to postgres work (and am indeed starting to work on something similar atm, schemas written, etc) - but at the end of the day mysql still does not cut it inmo. No offence to mysql developers, etc. Cheers Chris - Original Message - From: Sherwood McGowan [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, August 05, 2005 2:06 PM Subject: ARTCP Project (was RE: [Asterisk-Users] Features you'd like to see ina GUI?) I'd like to officially reclaim the Features in a GUI thread ;) Asterisk Hackers, Admins, and general digital phreakers of old, After careful consideration, the ARTCP project will probably have to be split into two major sections, both distros or at least maps for a system to be designed as well as the software we develop. Section 1. ARTCP Provider This version will be a distro including the designed management/enduser/billing software hooking into an asterisk RT installation using pre-set mysql schemas. The reasoning for this is that it's much easier to design a database driven php package when you know what the schema will be. Section 2. ARTCP PBX Intended for medium to large pbx's for endusers that want RealTime performance, this project will be a distro with less overall features, but more than enough to handle PBX functions and more. Possible Section 3? AI-PBX Cpanel? (name?) This version will be the same as ARTCP PBX, but not running the RT version of Asterisk. All the above sections should have: A complete branded distribution of linux, as small as possible. Bacula backup system Zabbix monitoring system Asterisk (STABLE) All Asterisk apps/modules that are required for final product PHP MySQL Apache Samba (for end user uploading music on hold files) Webmin (for end user control of system) I'd like to try and get anyone interested in contributing code work to join me in an online IRC chatroom sometime around August 17, 2005. Please reply (just please erase the [Asterisk-Users] section of your subject, otherwise it'll get trapped in the general list folder due to message moving rules) to me directly, and we'll get everyone's availability worked out. I'll take contributions of funds as well to be split across the developers, but won't look for donations before at least some cursory info has been released to show that the project is at least happening ;) Cheers all, and I hope to see interest in getting this going. Sherwood McGowan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: ARTCP Project (was RE: [Asterisk-Users] Features you'd like to see ina GUI?)
Side note, I've jumped to a different name, as ARTCP defines more the control program portion than an entire distro. ARTP -Now- AstCD (Asterisk Complete Distribution) Obviously the name would change to something a little more memorable once the project is in a release phase Sherwood Mcgowan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: ARTCP Project (was RE: [Asterisk-Users] Features you'd like tosee ina GUI?)
I personally prefer MySQL-MAX. I curently run *RT in a large production environment comprised of more than 1K users, with MySQL-MAX as my backend. Also, it's a point of I've spent so much time working with MySQL that I don't want to have to jump systems. It's fit the needs of the VOIP provider I work for and causes no problems that I see, so if it ain't broke, don't fix it is the rule here ;) Thanks for your suggestion though :) --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Chris Thompson -Sent: Friday, August 05, 2005 9:11 AM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: ARTCP Project (was RE: [Asterisk-Users] Features -you'd like tosee ina GUI?) - -Outta intersted - why mysql? - -If postgresql not a better option? -I would happily contribute to postgres work (and am indeed -starting to work on something similar atm, schemas written, -etc) - but at the end of the day mysql still does not cut it inmo. - -No offence to mysql developers, etc. -Cheers -Chris - -- Original Message - -From: Sherwood McGowan [EMAIL PROTECTED] -To: 'Asterisk Users Mailing List - Non-Commercial Discussion' -asterisk-users@lists.digium.com -Sent: Friday, August 05, 2005 2:06 PM -Subject: ARTCP Project (was RE: [Asterisk-Users] Features -you'd like to see ina GUI?) - - - I'd like to officially reclaim the Features in a GUI thread ;) - - Asterisk Hackers, Admins, and general digital phreakers of -old, After - careful consideration, the ARTCP project will probably have to be - split into two major sections, both distros or at least maps for a - system to be designed as well as the software we develop. - - Section 1. ARTCP Provider - This version will be a distro including the designed - management/enduser/billing software hooking into an asterisk RT - installation using pre-set mysql schemas. The reasoning for this is - that it's much easier to design a database driven php -package when you - know what the schema will be. - - Section 2. ARTCP PBX - Intended for medium to large pbx's for endusers that want RealTime - performance, this project will be a distro with less -overall features, - but more than enough to handle PBX functions and more. - - Possible Section 3? AI-PBX Cpanel? (name?) This version -will be the - same as ARTCP PBX, but not running the RT version of Asterisk. - - All the above sections should have: - A complete branded distribution of linux, as small as possible. - Bacula backup system - Zabbix monitoring system - Asterisk (STABLE) - All Asterisk apps/modules that are required for final product PHP - MySQL Apache Samba (for end user uploading music on hold -files) Webmin - (for end user control of system) - - I'd like to try and get anyone interested in contributing -code work to - join me in an online IRC chatroom sometime around August 17, 2005. - Please reply (just please erase the [Asterisk-Users] -section of your - subject, otherwise it'll get trapped in the general list -folder due to - message moving rules) to me directly, and we'll get everyone's - availability worked out. - - I'll take contributions of funds as well to be split across the - developers, but won't look for donations before at least -some cursory - info has been released to show that the project is at least -happening - ;) - - Cheers all, and I hope to see interest in getting this going. - - Sherwood McGowan - - - ___ - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - - -___ -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: ARTCP Project (was RE: [Asterisk-Users] Features you'd like to see ina GUI?)
On Friday 05 August 2005 09:11, Chris Thompson wrote: Outta intersted - why mysql? If postgresql not a better option? This is an old argument which works both ways just fine. -- List Manager Network Voice Communications, Inc. netwvcom.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk (Comedian Mail) and AUDIX
Has anyone been able to successfully integrate the Avaya AUDIX voicemail system with Asterisk? I have been diligently investigating converting our small (Ontario, Canada) office to Asterisk, and ditching our Avaya PBX. However, our head office (New Jersey, USA) maintains our AUDIX system, and a) have no intentions of leaving it and b) some users rely upon AUDIX's ability to transfer messages between voicemail accounts. At worst case, I would like our Asterisk users to be able to bounce to an AUDIX mailboxfor voicemail storage. At best, I would like the users to use Comedian mail, with AUDIX messages from our head office forwarded automagically to Comedian. Please, help. Mark McQuiggan This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP signaling vs Media (Voice) Traffic
I have an Asterisk serving 15 people using the X-Lite soft-phone. Currently they all register to the internal IP address of Asterisk (192.168.1.110). I only use VoIP internally. External calls go PSTN. I'd like to arrange it so that they register to our external WAN address (port forwarded to Asterisk) so that they can go mobile and still have Asterisk service. Is it possible to arrange it so that when in the office, the SIP signaling goes through the external WAN, but the Media (Voice) traffic stays local? In other words when a user is on the local LAN, I don't want their voice traffic going out on the net and then back in. Thanks, Hugh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phone hangups after a TEI check request
Hello, I am using asterisk with a HFC-Card which is connected to the internal S0 of a Siemens Hi Path 3000. When asterisk receives a TEI check request an active call to the PSTN ends. Does someone know this problem? I tried bri-stuff.0.1.0-RC4a, bristuff-0.2.0-RC8h and bristuff-0.2.0-RC8m. There is always the same error: == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up received TEI check request for TEI = 64 received TEI check request for TEI = 64 == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up Thanks for your help. Achim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with realtime SIP
vinod malani wrote: Hi Guys, We have just set up Asterisk 1.0.7 with (CVS Head) for a realtime enviorment using MySQL Asterisk Addons. 1.0.7 is NOT CVS HEAD! 1.0.7 is STABLE and RealTime doesn't work on STABLE! -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk registered in ser proxy
Yes you can.
In sip.conf you must edit:
register = user in SIP proxy:password in SIP proxy:AUTH-ID in SIP
proxy@IP of SIP proxy/local peer in asterisk where you answer the call
and you must define a peer for the SIP proxy:
[SIP-proxy]
type=peer
context=where you have the peer for answer
secret=password in SIP proxy
username=AUTH-ID in SIP proxy
fromdomain=IP of SIP proxy
canreinvite=yes
dtmfmode=RFC2833
canreinvite=yes
qualify=yes
host=IP of SIP proxy
insecure=very
fromuser=user in SIP proxy
disallow=all
allow=g729
Finally, to make a call from asterisk yo need in the extension.conf
something like this:
exten = _X.,1,Dial(SIP/SIP-proxy/${EXTEN})
This should work!
Regards.
Jsalas
-Mensaje original-
De: Jenna Cole [mailto:[EMAIL PROTECTED]
Enviado el: Friday, August 05, 2005 8:25 AM
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] asterisk registered in ser proxy
is it possible to register asterisk in a sip proxy as
if it were a terminal (like a cisco ATA)? how?
Thanx
Jenna ;)
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Re: [Asterisk-Users] Cisco ATA and a PayPhone
Help is on the way:) This is quite simple to achieve on Sipura units. There is a parameter called Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2) It defines the frequencies and duration of the tone. The 10 you see near the end is the duration. Simply change it to 60 like this and you're done: Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];60(*/0/1+2) I just tried it and it works like you want it. I'm not the OP and do plan on deploying several spa3k's, is there somewhere this kind of stuff is documented for the spa's? The Sipura Admin guide covers also the spa3k. The Dial Tone parameter is the same for all SPAs. You can ask your reseller for the Admin guide if you don't have it. Cheers. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, Tenovis, Fritz, capi problem
Background: We are currently implementing an Asterisk based solution for a customer to enable teleworker phone access. We have connected an Asterisk box running SUSE 9.3 with an AVM Fritz PCI ISDN card installed running CAPI to a Tenovis box. Softphones using SIP (referred to as SIP user) have been configured and can register no problem with Asterisk. The SIP users can call each other with no problem. Problem: Incoming calls to the SIP users work fine, but outgoing calls do not. Outgoing calls ring the called number no problem (dialing using chan_capi works fine), but when the called number answers, Asterisk does not receive any notification that the call has been answered, and hence the softphone keeps ringing. If the hash (#) is pressed on the called phone, the call is then shown as answered, Asterisk sees it as answered, but there is only oneway voice. The called party can hear the SIP user, but the SIP user cannot hear the called party. Asterisk also does not get notification that the call was terminated if the called party disconnects the call. Attempted Solutions: We believe this to be a DTMF problem, but are not sure. We have tried changing the DTMF in the sip.conf and capi.conf files, but nothing seems to solve the problem. If anyone has solved this problem, or had any experience with a similar setup, we would greatly appreciate any assistance. Regards to all, Joe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIPPeersAction class file not found intheAsterisk-java.jar file
Hi Stefan,
I have all the necessary files for the code to be executed. The
fastagi-mapping.properties file is also correct. But still I am getting the
error for
No script configured for agi://
The IP address is correct and as well as the agi file name. Does it make a
difference giving a Tab or a space when giving the mapping of agi file name
and class file name in the fastagi-mapping.properties file.
Is there any other reason for getting this error
Please reply
Regards,
Bharat M. Sarvan
Software Engineer - VoIP
EZZI BPO Pvt Ltd.,
PUNE.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bharat M.
Sarvan
Sent: Friday, August 05, 2005 2:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SIPPeersAction class file not found
intheAsterisk-java.jar file
Well thanks Stefan, for the help but when I am executing the AGI script I am
getting the errors as below:
Aug 5, 2005 3:29:44 AM net.sf.asterisk.util.impl.JavaLoggingLog info
INFO: Received connection.
Aug 5, 2005 3:29:45 AM net.sf.asterisk.util.impl.JavaLoggingLog error
SEVERE: Unable to create AGIScript instance of type HelloScript
Aug 5, 2005 3:29:45 AM net.sf.asterisk.util.impl.JavaLoggingLog error
SEVERE: No script configured for agi://65.125.224.207/bharat.agi
What does it mean by No Script configured for agi:// and can you please
tell me how do I come up with this error?
Regards,
Bharat M. Sarvan
Software Engineer - VoIP
EZZI BPO Pvt Ltd.,
PUNE.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter
Sent: Friday, August 05, 2005 6:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIPPeersAction class file not found in
theAsterisk-java.jar file
On Thu, 2005-08-04 at 18:25 +0530, Bharat M. Sarvan wrote:
I am working on Fastagi and I am making use of
Asterisk-java. But I don't find the class file for SIPPeersAction.
The SIPPeersAction is not part of Asterisk-Java 0.1, it is available in
CVS-HEAD only.
Besides that the action classes in net.sf.asterisk.manager.action can
only be used with the Manager API and not with FastAGI.
So if you want to retrieve a list of sip peers you need to do that via
the Manager API. With Asterisk 1.0.x and Asterisk-Java 0.1 you can do
this via the CommandAction. Only if you are already using Asterisk
CVS-HEAD and Asterisk-Java CVS-HEAD you can use the new SipPeerAction.
Example with CommandAction:
import java.util.Iterator;
import net.sf.asterisk.manager.ManagerConnection;
import net.sf.asterisk.manager.ManagerConnectionFactory;
import net.sf.asterisk.manager.action.CommandAction;
import net.sf.asterisk.manager.response.CommandResponse;
public class Manager
{
private ManagerConnection c;
public Manager() throws Exception
{
c = new ManagerConnectionFactory().getManagerConnection(host,
user, pass);
}
public void run() throws Exception
{
c.login();
CommandAction action;
CommandResponse response;
Iterator lineIterator;
action = new CommandAction();
action.setCommand(sip show peers);
response = (CommandResponse) c.sendAction(action);
lineIterator = response.getResult().iterator();
while (lineIterator.hasNext())
{
System.out.println(lineIterator.next());
}
c.logoff();
}
public static void main(String[] args) throws Exception
{
new Manager().run();
}
}
This produces something like:
Name/usernameHostDyn Nat ACL Mask Port
Status
1313/131310.13.0.61 D A 255.255.255.255 5061
Unmonitored
1312/131210.13.0.61 D A 255.255.255.255 5061
Unmonitored
1311/131110.13.0.61 D A 255.255.255.255 5061
Unmonitored
1310/1310(Unspecified)D A 255.255.255.255 0
Unmonitored
1303/1303(Unspecified)D N 255.255.255.255 0
Unmonitored
1302/1302(Unspecified)D A 255.255.255.255 0
Unmonitored
1301/1301(Unspecified)D A 255.255.255.255 0
Unmonitored
=Stefan
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[Asterisk-Users] Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone
Hi! The bandwith is not the problem, uploadspeed is about 400 kbits. I think I found the solution, I need to have a Proxy in the middle, or set up a IAX2 client and server at each end I will be testng this next week. BR Martin Kronstad What is the upload speed on B? Looks to me as you have bandwidth problem! Martin Kronstad wrote: Hi! Problem: I can_t hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) - Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone(Location B) I am having problems with sound, I have opened the following ports: Location A: 10 000 - 20 000 (TCP and UDP) 5060 (TCP and UDP) 8000 (TCP and UDP) Location B: 8000 (TCP and UDP) 5060 (TCP and UDP) I am using [EMAIL PROTECTED] 1.3 , and xlite as softphone. I have tried to set the softphone I have set the extention parameters(in sip.conf) to: ;; Location A [200] username=200 type=friend secret=1234 record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Location A 200 ;; Location B [201] username=201 type=friend secret=1234 record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Location B 201 My sip.conf : port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) externip=80.202.50.16 disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown language=no Best Regard Martin Kronstad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] ip phones
Thanks Greg, Not too much skill yet. I will be doing first time. What is cheapest available in cisco or polycon. Any other company that is a little cheaper. Varun - Original Message - From: [EMAIL PROTECTED] Date: Friday, August 5, 2005 10:34 am Subject: RE: [Asterisk-Users] ip phones Well this depends on your skill and budget. I have tried a number of phones, and the cisco 7960 and polycom ip600 are the best ones I ever used. I only with there was a cisco with a hold button :) When it comes down to it, although these phones are expensive, to me, they are worth every penny versus the cheaper phones. Even softphones, while they are great for traveling, do not closely parallel the cisco and polycoms. They are a trick to setup, but after spending a lot of time and money on less expensive units, the only way to go for me. Hope this is of some help. Greg -Original Message- From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, August 05, 2005 12:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ip phones Hello, I want to setup asterisk and do VOIP. Somebody from US has offered to get me ip phones. Can anybody suggest a few good and resonably priced phones models. Thanks Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] ip phones
Hard phones. Varun - Original Message - From: Jason Walker [EMAIL PROTECTED] Date: Friday, August 5, 2005 10:35 am Subject: RE: [Asterisk-Users] ip phones Soft phones or hard phones? There are many free VOIP soft phones out there. -Original Message- From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, August 04, 2005 9:57 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ip phones Hello, I want to setup asterisk and do VOIP. Somebody from US has offered to get me ip phones. Can anybody suggest a few good and resonably priced phones models. Thanks Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk (Comedian Mail) and AUDIX
McQuiggan, Mark xt46480 wrote: Has anyone been able to successfully integrate the Avaya AUDIX voicemail system with Asterisk? Haven't tried it, but sounds doable. At worst case, I would like our Asterisk users to be able to bounce to an AUDIX mailbox for voicemail storage. At best, I would like the users to use Comedian mail, with AUDIX messages from our head office forwarded automagically to Comedian. Our Definity admin says he could make fantom exenstions with mailboxes. Your dial plan then would, instead of calling the asterisk voicemail system on unavable or busy, send them to the Definity fantom extension. The problem being, vm indicators would not be present. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a right place for a include_once statement in a PHP AGI script?
its kind of difficult to say if we dont know what the included php script has. i think that the wrap function that Christoph propouse it may work for debuggin purposes, but i dont think it will solve the problem. Until you tell us, or show us, the content of the scripts we will be doing our best to guess the problem. I think you have parse error in the included script, try turning on the log errors directives in php.ini, turn off the output errors stuff, so Asterisk will not get confused with php warnings and other stuff. Let us know what happen... best regards On 8/5/05, Christoph Eicke [EMAIL PROTECTED] wrote: On Friday 05 August 2005 14:04, Leo Burd wrote: Hello there, I'm new to PHP AGIs and I'm having problems with a particular script that has a include_once statement on it. If I remove that stament, the script runs until the section of the code that depends on the include and then returns. If I include that statement, the script does not seem to run at all. What shall I do? Leo, wrap a function around whatever is in the included script, make your include_once() statement at the top of the AGI and then simply call the function at the place where it's necessary for that code to be executed. Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a right place for a include_oncestatement in a PHP AGI script?
agree with all written below - additionally use php -l to lint/check the syntax of the file (and the include) if needed - do a include_once 'bleh.php || die some message; to see if thats an issue. my $0.02 - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 05, 2005 3:49 PM Subject: Re: [Asterisk-Users] Is there a right place for a include_oncestatement in a PHP AGI script? its kind of difficult to say if we dont know what the included php script has. i think that the wrap function that Christoph propouse it may work for debuggin purposes, but i dont think it will solve the problem. Until you tell us, or show us, the content of the scripts we will be doing our best to guess the problem. I think you have parse error in the included script, try turning on the log errors directives in php.ini, turn off the output errors stuff, so Asterisk will not get confused with php warnings and other stuff. Let us know what happen... best regards On 8/5/05, Christoph Eicke [EMAIL PROTECTED] wrote: On Friday 05 August 2005 14:04, Leo Burd wrote: Hello there, I'm new to PHP AGIs and I'm having problems with a particular script that has a include_once statement on it. If I remove that stament, the script runs until the section of the code that depends on the include and then returns. If I include that statement, the script does not seem to run at all. What shall I do? Leo, wrap a function around whatever is in the included script, make your include_once() statement at the top of the AGI and then simply call the function at the place where it's necessary for that code to be executed. Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another problem on queues
Hello
all,
I have
been posting some questions about this problems that I cannot yet solve, but I
think I have a better diagostic, so maybe someone can give me a clue why it is
happenning.
I have
Asterisk + AMPconfigured as a PBX with a Customer Center Queue with 4
agents that login/logout dinamically.
If
there are no agents, queue timesout and gets derived to another queue that
somebody answers as last resort or waits there.
If
there is at least one agent logged in, but it is busy, dialparties.agi detects
that that extension has no callwaiting, no callforward, no voicemail, and hangs
up the call inmediately with a "nobody is available to take your call right now"
message, making the queue useless.
My
PSTN connection is an AS5300 in SIP, my extensions are analog phones connected
to an Audiocodes MP108-FXS with SIP.
This
is the output from CLI with High Verbosity:
XXX.XXX.XXX.XXX is the IP of the AS5300, 8521 and 8522 are the only two
agents in the queue that have inbound calls in progress when a third call
arrives and this happens. 8500 is the queue number
-- Executing SetVar("SIP/XXX.XXX.XXX.XXX-43921110", "FROM_DID=1154538500") in
new stack -- Executing
Goto("SIP/XXX.XXX.XXX.XXX-43921110", "ext-did|1154538500|1") in new
stack -- Goto (ext-did,1154538500,1)
-- Executing Goto("SIP/XXX.XXX.XXX.XXX-43921110", "ext-queues|8500|1") in new
stack -- Goto (ext-queues,8500,1) --
Executing Answer("SIP/XXX.XXX.XXX.XXX-43921110", "") in new
stack -- Executing
SetCIDName("SIP/XXX.XXX.XXX.XXX-43921110", "XXX.XXX.XXX.XXX") in new
stack -- Executing SetVar("SIP/XXX.XXX.XXX.XXX-43921110",
"MONITOR_FILENAME=/var/spool/asterisk/monitor/q") in new
stack -- Executing Queue("SIP/XXX.XXX.XXX.XXX-43921110",
"8500|t|||300") in new stack -- Started music on hold,
class 'operadores', on SIP/XXX.XXX.XXX.XXX-43921110 --
Executing Macro("Local/[EMAIL PROTECTED],2",
"exten-vm|[EMAIL PROTECTED]|8521") in
new stack -- Executing SetVar("Local/[EMAIL PROTECTED],2",
"FROMCONTEXT=exten-vm") in new stack -- Executing
Macro("Local/[EMAIL PROTECTED],2",
"record-enable|8521|IN") in new stack -- Executing
GotoIf("Local/[EMAIL PROTECTED],2",
"0 0?2:4") in new stack -- Goto
(macro-record-enable,s,4) -- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?5:8") in new stack -- Goto
(macro-record-enable,s,8) -- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?9:12") in new stack -- Goto
(macro-record-enable,s,12) -- Executing DBget("Local/[EMAIL PROTECTED],2",
"RecEnable=RECORD-IN/8521") in new stack -- DBget:
varname=RecEnable, family=RECORD-IN, key=8521 -- DBget:
Value not found in database. -- Executing SetVar("Local/[EMAIL PROTECTED],2",
"CALLFILENAME=20050805-43-1123251103.2060") in new
stack -- Called Local/[EMAIL PROTECTED]
-- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?15:99") in new stack -- Goto
(macro-record-enable,s,99) -- Executing NoOp("Local/[EMAIL PROTECTED],2",
"NO RECORDING NEEDED") in new stack -- Executing
GotoIf("Local/[EMAIL PROTECTED],2",
"1?novm|1:4") in new stack -- Goto
(macro-exten-vm,novm,1) -- Executing Macro("Local/[EMAIL PROTECTED],2",
"dial|120|tr|8521") in new stack -- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?4:2") in new stack -- Goto
(macro-dial,s,2) -- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?4:3") in new stack -- Goto
(macro-dial,s,3) -- Executing SetCIDName("Local/[EMAIL PROTECTED],2",
"XXX.XXX.XXX.XXX") in new stack -- Executing AGI("Local/[EMAIL PROTECTED],2",
"dialparties.agi") in new stack -- Launched AGI Script
/var/lib/asterisk/agi-bin/dialparties.agi --
dialparties.agi: request = dialparties.agi --
dialparties.agi: priority = 4 -- dialparties.agi:
extension = s -- dialparties.agi: language =
en -- dialparties.agi: accountcode
= -- dialparties.agi: uniqueid =
1123251103.2060 -- dialparties.agi: channel = Local/[EMAIL PROTECTED],2
-- dialparties.agi: callerid =
XXX.XXX.XXX.XX.XXX.XXX.XXX --
dialparties.agi: context = macro-dial --
dialparties.agi: type = Local -- dialparties.agi:
rdnis = unknown -- dialparties.agi: enhanced =
0.0 -- dialparties.agi: dnid = unknown
dialparties.agi: Caller ID is not set --
dialparties.agi: Added extension 8521 to extension map
-- dialparties.agi: Extension 8521 cf is disabled
-- dialparties.agi: Extension 8521 do not disturb is disabled ==
Parsing '/etc/asterisk/manager.conf': Found == Parsing
'/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged
on from 127.0.0.1 == Manager 'admi
[Asterisk-Users] Audio files problem - as usual
Hello List! I have a problem that has been posted to the list more than once, but so far I have not been able to find a solution searching the archives and Google. The problem is with Asterisk audio files not being played to the x-lite client. I have an out-of-the-box [EMAIL PROTECTED] configuration with no additional hardware. I have created extensions, clients over the LAN are able to talk to each other and I can even listen to the MP3 files that come out of the box with the mp3play command (both on the Linux box and on the phone). But all the standard audio files (voicemail, text-to-speech and even music-on-hold) can't be heard. It looks like Asterisk just hangs until I hang up. Sounds familiar? Can you please help me out with this? Thanks, Luca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime IAX
I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial(SIP/2001-3761, IAX2/[EMAIL PROTECTED]/19566680301) in new stack -- SIP Seeding peer from astdb: '2001' at [EMAIL PROTECTED]:5060 for 3600 -- Called [EMAIL PROTECTED]/19566680301 Aug 5 10:25:50 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congesting call due to slow response -- IAX2/voicepulse-11 is circuit-busy -- Hungup 'IAX2/voicepulse-11' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Dial(SIP/2001-3761, IAX2/[EMAIL PROTECTED]/19566680301) in new stack -- Called [EMAIL PROTECTED]/19566680301 Aug 5 10:25:54 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congesting call due to slow response -- IAX2/NuFone-2 is circuit-busy -- Hungup 'IAX2/NuFone-2' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Dial(SIP/2001-3761, IAX2/[EMAIL PROTECTED]/19566680301) in new stack -- Called [EMAIL PROTECTED]/19566680301 -- Seeding 'pbxserver' at 66.135.38.93:4569 for 60 Aug 5 10:25:58 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congesting call due to slow response -- IAX2/sixTel-13 is circuit-busy -- Hungup 'IAX2/sixTel-13' == Everyone is busy/congested at this time (1:0/1/0) As you can see none of them go through. I have another Asterisk server connected with IAX2 that does work. To that server I can dial any extension without problems. I used http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20IAX to configure my * server. Any ideas? All three providers were working before I changed to Realtime IAX and I made sure to put all the necessary information into the Database. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] include behavior (word puzzle of the day)
The key seems to be listing the 10 digit extensions dialplan in a context other than the context they are defined in in sip.conf, correct? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dbruce Sent: Thursday, August 04, 2005 6:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] include behavior (word puzzle of the day) Try something like this: [context1] Include = internal-extensions include = egress [context2] include = egress [context3] include = pri-ingress include = internal-extensions [internal-extensions] ;sip users with 10 digit extensions [egress] ;media gateway terminating local 10 digit calls [pri-ingress] ;inbound PRI via media gateway Regards, Derek - Original Message - From: Damon Estep To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, August 04, 2005 6:26 PM Subject: [Asterisk-Users] include behavior (word puzzle of the day) In the example below context2 is included in context3 because it is included in context1. Is there a way to include context2 in context1, and context1 in context3, but not context2 in context3 as a result. [Context1] ;sip users with 10 digit extensions Include = context2 [context2] ;media gateway terminating local 10 digit calls [context3] ;inbound PRI via media gateway Include = context1 I have a case where a dialplan is insecure because inbound calls in context3 can be re-routed back out in context2. Actually, what occurs is a loop, where the call comes in context3, finds no match in context1, egresses in context2, and repeats the loop, setting up a lot of calls in a short period of time! Extensions in context1 need to be able to reach extensions in context2 Inbound calls into context3 need to be able to reach extensions in context1 Inbound calls in context3 MUST be restricted from reaching extensions in context2 which are outside extensions sent out to a SIP provider. It would seem more logical and secure if includes did not cascade, or would not make 2 hops Perhaps I have failed to understand some simple concept that would resolve this issue? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Masters changes / Line looses
Hi, We have just done an upgrade now when ever the console displays a single line as below Zaptel: Master Changed to TE4/0/1 Zaptel: Master Changed to TE4/0/2 Zaptel: Master Changed to TE4/0/1 Zaptel: Master Changed to TE4/0/1 The asterisk r Show alls the lines been Hung Up and everyone is disconnected from the PRI /T1 Have been chasing this down all day and now just found this was the cause of the frustration of the 200+ people in the organisation. It is now 1.48am in Brisbane, Australia and heaps of angry people will be here in a few hours. Any knights out there??? Thanks James ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Phone Pro Beta - New Version Available
[ New Beta Version - Beta Extended ] A new version of IAX Phone Pro Beta is available. A few bugs have been fixed and the beta has been extended until October 12, 2005 (the date of AstriCon 2005). You can download either a new install (be sure to un-install the old version) or just a new binary. [ IAX Phone Pro Features ] * Dial/answer/hold/recall/reject * Multi-number advanced speed dial. * Standard and innovative Tool Bar skins. * Handles iax:, sip: and tel: URLs * Integrated web browser for co-browsing * Integrated call recording and playback. * Advanced phone book with CSV import. * Advanced call log with CSV export. * Speaker Phone * Audio mute. * Auto answer. * Intercom calling with password. * Multi-server registration. * Audio Codecs: uLaw, aLaw, GSM, iLBC, Speex. * Server-by-server codec setting. * Call statistics. * Local or server-side call forwarding. * Local or server-side do-not-disturb. * TAPI integration for Outlook, ACT, Goldmine, etc. * Direct IP to IP calling * Dial by IAX or SIP URI (URL) [ Try Out Phone URI/URL Dialing ] IAX Phone Pro supports the ability to handle telephony URIs (links). This feature is great for call centers or web-based contact management solutions. When you install the phone, it configures your copy of Windows to pass all links marked as iax:, sip:, or tel: to IAX Phone Pro. IAX Phone then does its best to place a call to the destination number. You can create these links by using the iax, sip and tel URI schemes. Simply use the following examples as a guide: a href=iax:[EMAIL PROTECTED]Call Ipsando HQ/a a href=tel:1000Call Extension 1000/a a href=tel:18005551212800 Directory Information (US Only)/a a href=sip:[EMAIL PROTECTED]Olle Johansson over SIP (requires the SIP-Over-IAX)/a IAX and SIP accept IAX or SIP URIs respectively. TEL allows you to enter any extension or dialable number. Note that your browser /may/ ask you to authorize each of the URI types (iax, sip, and tel) the first time you click on them. You must select OK in order for the calls to go through. [ Download IAX Phone Pro ] https://www.astricon.net/phone/ipbeta.php Steven Sokol CEO/Manager Sokol Associates, LLC Ask Me About AstriCon 2005! http://www.astricon.net/ begin:vcard fn:Steven Sokol n:Sokol;Steven email;internet:[EMAIL PROTECTED] tel;work:816.822.1807 x-mozilla-html:FALSE url:http://www.sokol-associates.com version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] function declaration isn't a prototype
In file included from include/asterisk/utils.h:26, from term.c:32: include/asterisk/strings.h:232: parse error before `va_list' include/asterisk/strings.h:232: warning: function declaration isn't a prototype make: *** [term.o] Error 1 pls advise on how i can fix this, It's fixed in CVS head now, Do an update. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No dial tone on BT100
I'm seeing all sorts of problems and it's probably more of my lack of experience than anything else. I have a BT100 running 1.0.6.7 code. When I go to the status page it says it's not registered (hmm, that's not good). I also can't get dial tone but I can dial! I'm afraid I'm lost any good pointers? I've done a sip debug and all I'm seeing for the BT100 - Asterisk is Asterisk asking the BT100 for it's option (102 Options) and the BT100 not replying. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: ARTCP Project (was RE: [Asterisk-Users] Features you'd like tosee ina GUI?)
Hi, Sherwood McGowan wrote: I personally prefer MySQL-MAX. I curently run *RT in a large production environment comprised of more than 1K users, with MySQL-MAX as my backend. Also, it's a point of I've spent so much time working with MySQL that I don't want to have to jump systems. It's fit the needs of the VOIP provider I work for and causes no problems that I see, so if it ain't broke, don't fix it is the rule here ;) Many people like many DB's for many different reasons. I for one would appreciate any design where the database functionality either: - is using an abstraction layer so many DB's can be used, or: - is designed so all direct DB interaction is in one centralised place so rewriting for a different DB becomes a manageable task. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need Help Troubleshooting Broadvoice Connection
Ok I can register with BV fine (as far as I can tell from asterisk - see below). I am able to make outgoing calls but all incoming calls get a fast busy. I have opened and forwarded the following ports to my pbx: 5060-5063 UDP + TCP 69 UDP (BV claims they need this) 1-2 UDP I tried switching proxies as well, tried both LAX and CHI with the same problem. Called BV they said they can conenct andd call it with a softphone so it must be a configuration issue. Here are some outputs that might be helpful: Asterisk -r sip show registry asterisk1*CLI HostUsername Refresh State sip.broadvoice.com:5060 [EMAIL PROTECTED]23 Registered asterisk1*CLI sip show peers asterisk1*CLI Name/usernameHostDyn Nat ACL Mask Port Status bv/2068660133147.135.12.128 N 255.255.255.255 5060 Unmonitored /(Unspecified)D 255.255.255.255 0 Unmonitored 1005/1005(Unspecified)D 255.255.255.255 0 Unmonitored 1004/1004(Unspecified)D 255.255.255.255 0 Unmonitored 1003/1003(Unspecified)D 255.255.255.255 0 Unmonitored 1002/1002(Unspecified)D 255.255.255.255 0 Unmonitored [Kasterisk1*CLI sip show peer bv asterisk1*CLI * Name : bv Secret : Set MD5Secret: Not set Context : from-pstn Language : FromUser : 2068660133 FromDomain : sip.broadvoice.com Callgroup: (0) Pickupgroup : (0) Mailbox : LastMsgsSent : -1 Dynamic : No Expire : -1 seconds Expiry : 900 Insecure : Very Nat : Always ACL : No CanReinvite : No PromiscRedir : No DTMFmode : inband LastMsg : 0 ToHost : sip.broadvoice.com Addr-IP : 147.135.12.128 Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Username : 2068660133 Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw|alaw) Status : UNKNOWN Useragent: Full Contact : (not sure about that Status = UNKNOWN, is that a problem?) Get full output on outgoing calls and they connect sucessfully Get zero output on incoming calls, pbx never seem to get them Here is my sip.conf [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED] [sip.broadvoice.com] username=2068660133 user=2068660133 type=user secret=mypass nat=yes insecure=very host=sip.broadvoice.com fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband context=from-pstn authname=2068660133 Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbiew extensions.conf question
We do extension by extension is our dialing plan because we have a wildcard at
the end trapping all unused extensions and playing a this extension is not in
use message and forwarding users into our IVR. It depends on individual
circumstances which works better. We have 300 DIDs for our sip phones, and
only 50 in use. Those 50 are also not sequential extensions. So it's less
painful to approach this way for our circumstance. If you had all of your
extensions in use, the wildcard would be easier and cleaner. Then if you
needed to remove one, include a [not-in-service] context above the in use
extensions.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 9:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question
But why do it that way?
Wouldn't:
exten = _72X,1,Dial(SIP/${EXTEN},50)
Be ideal? Or at least an easier way to expand the dialplan without mucho
administration?
Just a question...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tarpo, Louie
Sent: Thursday, August 04, 2005 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] newbiew extensions.conf question
He means that exten = 720,1,macro(sipexten,${EXTEN}) is the template.
You'd write out the rest of the config file like so
exten = 720,1,macro(sipexten,${EXTEN})
exten = 721,1,macro(sipexten,${EXTEN})
exten = 722,1,macro(sipexten,${EXTEN})
and so forth.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 6:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question
That would make all callers have to call 720 as there is not other extension
defined. As a result, all calls would go to 720. ${EXTEN} would always be
720.
I don't follow your logic.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question
Right track, but it can be simplified even more
exten = 720,1,macro(sipexten,${EXTEN})
On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:
We handled it by creating a macro which dials the exten, then sends
the call to voicemail.
You could create it where each extension is handled seperately
exten = 720,1,Macro(sipexten,720)
exten = 721,1,Macro(sipexten,720)
etc
or you could handle them all in a group with wildcards
exten = _72x,1,Macro(sipexten,${EXTEN})
then the macro would look something like
[macro-sipexten]
exten = s,1,NoOp(${CallerIDNum})
exten = s,2,Dial(SIP/${ARG1},24)
exten = s,3,Goto(s-${DIALSTATUS}, 1)
exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to
voicemail, play unavailable message
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,VoiceMail(b${ARG1});Send to
voicemail, play busy message
exten = s-BUSY,2,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
Depends on your needs which way would work better. We define
extension by extension individually, then have a wildcard at the
end that plays a message that says the extension is not in use and
then puts them in our main menu. In case we have to remove or
change an extension individually.
Louie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kenny
Kant
Sent: Thursday, August 04, 2005 4:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbiew extensions.conf question
I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server. I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc
in my from-sip context I began doing entries such as:
exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)
exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)
..etc ..etc
This is not a big deal for such a small number of
extensions but I was thinking about larger installs..
this would begin to suck. Is there anyway around
this?
Thanks!
Kenny
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
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[Asterisk-Users] Zaptel warning
Hi all, When I was making calls from an IP phone, through a X100P, to PSTN, the following error was encountered. -- Executing Dial(SIP/25086937-aa6c, Zap/1/91713545) in new stack -- Called 1/91713545 -- Zap/1-1 answered SIP/25086937-aa6c Aug 6 00:12:53 WARNING[3983]: chan_zap.c:4717 zt_indicate: Don't know how to set condition 16 on channel Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Hungup 'Zap/1-1' Any idea for this problem? Many thanks. Newbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: ARTCP Project (was RE: [Asterisk-Users] Features you'd like toseeina GUI?)
I'll do what I can. This is all I can say about it. We haven't even had the first meeting of contributors yet, but I'm sure we will do what we can. The idea is that this is STABLE, and since I don't use PostgreSQL at all, I'm sticking with what I know to be sure of stability. I'll take the centralization into consideration though, so that other users can install the distro and then download postgreSQL and modify the configuration to use it. Sherwood --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Florian Overkamp -Sent: Thursday, September 23, 2004 11:19 PM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: ARTCP Project (was RE: [Asterisk-Users] Features -you'd like toseeina GUI?) - -Hi, - -Sherwood McGowan wrote: - I personally prefer MySQL-MAX. I curently run *RT in a large - production environment comprised of more than 1K users, -with MySQL-MAX as my backend. - Also, it's a point of I've spent so much time working with -MySQL that - I don't want to have to jump systems. It's fit the needs of -the VOIP - provider I work for and causes no problems that I see, so -if it ain't - broke, don't fix it is the rule here ;) - -Many people like many DB's for many different reasons. I for -one would appreciate any design where the database -functionality either: - -- is using an abstraction layer so many DB's can be used, or: -- is designed so all direct DB interaction is in one -centralised place so rewriting for a different DB becomes a -manageable task. - -Florian -___ -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk registered in ser proxy
thanx for the reply.
i tried it, and now asterisk is doing something.
but the problem is that instead of sendind a
REGISTER message to the SIP PROXY, it is sendind an
OPTIONS
message, and the PROXY responds with 404 NOT FOUND
ihave in my sip.conf file:
register = 7771::[EMAIL PROTECTED]/7771
[10.0.0.115]
type=peer
context=default
secret=
username=7771
fromdomain=10.0.0.115
canreinvite=yes
dtmfmode=RFC2833
qualify=yes
host=10.0.0.115
insecure=very
fromuser=7771
and in the extentions.conf:
[default]
exten = 7771,1,SetLanguage(en)
exten = 7771,2,Wait(1)
exten = 7771,3,Answer
exten = 7771,4,Playback(privacy-thankyou) ; plays the
demo after answering
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
--- Juan Salas [EMAIL PROTECTED] escribió:
Yes you can.
In sip.conf you must edit:
register = user in SIP proxy:password in SIP
proxy:AUTH-ID in SIP
proxy@IP of SIP proxy/local peer in asterisk
where you answer the call
and you must define a peer for the SIP proxy:
[SIP-proxy]
type=peer
context=where you have the peer for answer
secret=password in SIP proxy
username=AUTH-ID in SIP proxy
fromdomain=IP of SIP proxy
canreinvite=yes
dtmfmode=RFC2833
canreinvite=yes
qualify=yes
host=IP of SIP proxy
insecure=very
fromuser=user in SIP proxy
disallow=all
allow=g729
Finally, to make a call from asterisk yo need in the
extension.conf
something like this:
exten = _X.,1,Dial(SIP/SIP-proxy/${EXTEN})
This should work!
Regards.
Jsalas
-Mensaje original-
De: Jenna Cole [mailto:[EMAIL PROTECTED]
Enviado el: Friday, August 05, 2005 8:25 AM
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] asterisk registered in ser
proxy
is it possible to register asterisk in a sip proxy
as
if it were a terminal (like a cisco ATA)? how?
Thanx
Jenna ;)
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[Asterisk-Users] Looking for IBM or HP Server Recommendation
I am looking for a recommendation on either a Compaq/HP or IBM server for a 100 user Asterisk Server. Unfortunately because of customer constraints I cannot go with Supermicro, etc. Syed Akbar Alico Systems Inc www.alicosystems.com Tel: 562-436-1510 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel warning
VoIP Newbie wrote: Hi all, When I was making calls from an IP phone, through a X100P, to PSTN, the following error was encountered. -- Executing Dial(SIP/25086937-aa6c, Zap/1/91713545) in new stack -- Called 1/91713545 -- Zap/1-1 answered SIP/25086937-aa6c Aug 6 00:12:53 WARNING[3983]: chan_zap.c:4717 zt_indicate: Don't know how to set condition 16 on channel Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Hungup 'Zap/1-1' It's a warning, not an error. You don't have /etc/asterisk/indications.conf -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 Only terrorists use the r option to Dial. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbiew extensions.conf question
On Aug 5, 2005, at 11:20 AM, Tarpo, Louie wrote:
We do extension by extension is our dialing plan because we have a
wildcard at the end trapping all unused extensions and playing a
this extension is not in use message and forwarding users into
our IVR. It depends on individual circumstances which works
better. We have 300 DIDs for our sip phones, and only 50 in use.
Those 50 are also not sequential extensions. So it's less painful
to approach this way for our circumstance. If you had all of your
extensions in use, the wildcard would be easier and cleaner. Then
if you needed to remove one, include a [not-in-service] context
above the in use extensions.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 9:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question
But why do it that way?
Wouldn't:
exten = _72X,1,Dial(SIP/${EXTEN},50)
Be ideal? Or at least an easier way to expand the dialplan without
mucho
administration?
Just a question...
If all calls are handled exactly the same way then yes. But in my
world all extensions are not the same.
As an example some have voicemail, others do not. some are sip some
are zap.
By creating macros you have a macro for each class of extension and
your dialplan calls appropriately, but you need a specified line for
each. although a _match might catch all unspecified extensions would
have to try it. I find it much easier to troubleshoot/read/support by
more people to have each step explicitly spelled out. Although we do
use exten matching on outdials - don't really want to enter every
possible telephone number:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tarpo, Louie
Sent: Thursday, August 04, 2005 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] newbiew extensions.conf question
He means that exten = 720,1,macro(sipexten,${EXTEN}) is the template.
You'd write out the rest of the config file like so
exten = 720,1,macro(sipexten,${EXTEN})
exten = 721,1,macro(sipexten,${EXTEN})
exten = 722,1,macro(sipexten,${EXTEN})
and so forth.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
Walker
Sent: Thursday, August 04, 2005 6:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] newbiew extensions.conf question
That would make all callers have to call 720 as there is not other
extension
defined. As a result, all calls would go to 720. ${EXTEN} would
always be
720.
I don't follow your logic.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbiew extensions.conf question
Right track, but it can be simplified even more
exten = 720,1,macro(sipexten,${EXTEN})
On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:
We handled it by creating a macro which dials the exten, then sends
the call to voicemail.
You could create it where each extension is handled seperately
exten = 720,1,Macro(sipexten,720)
exten = 721,1,Macro(sipexten,720)
etc
or you could handle them all in a group with wildcards
exten = _72x,1,Macro(sipexten,${EXTEN})
then the macro would look something like
[macro-sipexten]
exten = s,1,NoOp(${CallerIDNum})
exten = s,2,Dial(SIP/${ARG1},24)
exten = s,3,Goto(s-${DIALSTATUS}, 1)
exten = s-NOANSWER,1,VoiceMail(u${ARG1});Send to
voicemail, play unavailable message
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,VoiceMail(b${ARG1});Send to
voicemail, play busy message
exten = s-BUSY,2,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
Depends on your needs which way would work better. We define
extension by extension individually, then have a wildcard at the
end that plays a message that says the extension is not in use and
then puts them in our main menu. In case we have to remove or
change an extension individually.
Louie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kenny
Kant
Sent: Thursday, August 04, 2005 4:07 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbiew extensions.conf question
I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server. I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc
in my from-sip context I began doing entries such as:
exten = 720,1,Dial(SIP/720,20)
exten = 720,2,Voicemail(u720)
exten = 721,1,Dial(SIP/721,20)
exten = 721,2,Voicemail(u721)
..etc ..etc
This is not a big deal for such a small number of
extensions but I was thinking about larger
Re: [Asterisk-Users] asterisk registered in ser proxy
Jenna Cole wrote: thanx for the reply. i tried it, and now asterisk is doing something. but the problem is that instead of sendind a REGISTER message to the SIP PROXY, it is sendind an OPTIONS message, and the PROXY responds with 404 NOT FOUND ihave in my sip.conf file: register = 7771::[EMAIL PROTECTED]/7771 [10.0.0.115] type=peer context=default secret= username=7771 fromdomain=10.0.0.115 canreinvite=yes dtmfmode=RFC2833 qualify=yes host=10.0.0.115 insecure=very fromuser=7771 Remove the qualify=yes and Asterisk will stop sending the options packets. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 Only terrorists use the r option to Dial. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel warning
Probably complaining about the dialed number. You say you are dialing the pstn - and I assume in north america. What is the number 91713545 supposed to dial? Last time I checked pstn calls were either 7 or 10/11 digits. perhaps you forgot to strip the 9 off? Perhaps the pstn is returning an error signal? On Aug 5, 2005, at 11:23 AM, VoIP Newbie wrote: Hi all, When I was making calls from an IP phone, through a X100P, to PSTN, the following error was encountered. -- Executing Dial(SIP/25086937-aa6c, Zap/1/91713545) in new stack -- Called 1/91713545 -- Zap/1-1 answered SIP/25086937-aa6c Aug 6 00:12:53 WARNING[3983]: chan_zap.c:4717 zt_indicate: Don't know how to set condition 16 on channel Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Hungup 'Zap/1-1' Any idea for this problem? Many thanks. Newbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nortel Option 11 and TE110P of Digium
Hi list: I have a client that needs to connect a Asterisk PBX with a TE110P of Digium and one Nortel Option 11. Actually the Nortel Option 11 have a AMI E1 card. With it the have problems of clock sync. They can change the AMI CARD to a PRI CARD, te questions are: 1) Which model of PRI is suggest for this ? 2) Some one have already do this ? 3) Is there form of correct de AMI problem ? Well i hope that you will answered me. Alvaro Parres P.D. If any one from Mexica have done this before pleas contact me (33) 35636261 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA186 can not generate dtmf
Hello: I have problems sending dtmf signal to an ATA186 my configuration is: ATA186 -- asterisk -- ATA186 -- FXS to FXO Converter -- PSTN The ATA186 are set to send dtmf RFC2833, but it seems that the ATA186 can't generate dtmf so I can dial to a PSTN number. Is there a setting that can fix my problem, inband dtmf does not work because I'm using G729 codec Thanks Erick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel warning
The next question is, was your call successful? I see you dialed an 8 digit number. Is that what's required on your line? MARK. Eric Wieling aka ManxPower wrote: VoIP Newbie wrote: Hi all, When I was making calls from an IP phone, through a X100P, to PSTN, the following error was encountered. -- Executing Dial(SIP/25086937-aa6c, Zap/1/91713545) in new stack -- Called 1/91713545 -- Zap/1-1 answered SIP/25086937-aa6c Aug 6 00:12:53 WARNING[3983]: chan_zap.c:4717 zt_indicate: Don't know how to set condition 16 on channel Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Hungup 'Zap/1-1' It's a warning, not an error. You don't have /etc/asterisk/indications.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk registered in ser proxy
if i remove that line, asterisk stop sendind the OPTIONS message to the SIP PROXY, but it's still NOT sending the REGISTER message. i would alse need to register more than one number --- Eric Wieling aka ManxPower [EMAIL PROTECTED] escribió: Jenna Cole wrote: thanx for the reply. i tried it, and now asterisk is doing something. but the problem is that instead of sendind a REGISTER message to the SIP PROXY, it is sendind an OPTIONS message, and the PROXY responds with 404 NOT FOUND ihave in my sip.conf file: register = 7771::[EMAIL PROTECTED]/7771 [10.0.0.115] type=peer context=default secret= username=7771 fromdomain=10.0.0.115 canreinvite=yes dtmfmode=RFC2833 qualify=yes host=10.0.0.115 insecure=very fromuser=7771 Remove the qualify=yes and Asterisk will stop sending the options packets. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 Only terrorists use the r option to Dial. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! ¡Abrí tu cuenta ya! - http://correo.yahoo.com.ar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone
Switch to IAXCOMM and use an IAX extension. Problem solved. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin KronstadSent: Friday, August 05, 2005 7:03 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone Hi! The bandwith is not the problem, uploadspeed is about 400 kbits. I think I found the solution, I need to have a Proxy in the middle, or set up a IAX2 client and server at each end I will be testng this next week. BR Martin Kronstad What is the upload speed on B? Looks to me as you have bandwidth problem! Martin Kronstad wrote: Hi! Problem: I can_t hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) - Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone(Location B) I am having problems with sound, I have opened the following ports: Location A: 10 000 - 20 000 (TCP and UDP) 5060 (TCP and UDP) 8000 (TCP and UDP) Location B: 8000 (TCP and UDP) 5060 (TCP and UDP) I am using [EMAIL PROTECTED] 1.3 , and xlite as softphone. I have tried to set the softphone I have set the extention parameters(in sip.conf) to: ;; Location A [200] username=200 type=friend secret=1234 record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="Location A" 200 ;; Location B [201] username=201 type=friend secret=1234 record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="Location B" 201 My sip.conf : port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) externip=80.202.50.16 disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown language=no Best Regard Martin Kronstad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, Tenovis, Fritz, capi problem
On Thu, 4 Aug 2005, Joseph Rothstein wrote: Background: We are currently implementing an Asterisk based solution for a customer to enable teleworker phone access. We have connected an Asterisk box running SUSE 9.3 with an AVM Fritz PCI ISDN card installed running CAPI to a Tenovis box. Softphones using SIP (referred to as SIP user) have been configured and can register no problem with Asterisk. The SIP users can call each other with no problem. Problem: Incoming calls to the SIP users work fine, but outgoing calls do not. Outgoing calls ring the called number no problem (dialing using chan_capi works fine), but when the called number answers, Asterisk does not receive any notification that the call has been answered, and hence the softphone keeps ringing. If the hash (#) is pressed on the called phone, the call is then shown as answered, Asterisk sees it as answered, but there is only oneway voice. The called party can hear the SIP user, but the SIP user cannot hear the called party. Asterisk also does not get notification that the call was terminated if the called party disconnects the call. Attempted Solutions: We believe this to be a DTMF problem, but are not sure. We have tried changing the DTMF in the sip.conf and capi.conf files, but nothing seems to solve the problem. What versions of Asterisk/chan_capi do you use ? Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: ARTCP Project (was RE: [Asterisk-Users] Features you'd like toseeina GUI?)
Why does the system have to be based on a linux distro? I think that's the wrong way to go. It's one thing to create a linux distro around a popular piece of software, but it's another to create software that can only be used as an entire linux distribution. If I were you I would take an existing application server platform of some type that is already popular, and build a management interface on top of that. Say something like Zope or mod perl/Mason. Preferrably you want a platform that has a good web application server and can also be extended using a good general purpose programming language like Perl or Python. Then after the core product is done add in secondary things like backups and monitoring. And although I know php is hugely popular and would make it easy for many to contribute, I would think twice about it. In real life it can get messy really quick on large projects, and it's not the best general purpose programming language. my favorite would be python, but that's just me. As for databases use an abstraction layer. Even if you aren't familiar with databases other than mysql, someone else will be. Some of these types of decisions will be what decides whether your project goes anywhere or not. And also, be prepared to do most of the work yourself with little help until a first usable version is produced. Lot's of people jump on the bandwagon once you get something going, but few will jump in and help a lot right from the start. If you don't have the time yourself, or can't put together at least a handful of people that do have the time, you are kind of doomed from the start. Everyone (like me) will be more than ready to give you their opinions on how to do things, but you won't see many of them when it comes time to actually do any work:) Good luck ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk MWI and Realtime
I'm testing my asterisk system and the realtime backend. My Asterisk build is rather aged, 03/18/2005 CVS. I have successfully moved Sip peers and Voicemail boxes to the realtime database backend and this works very well except for MWI. I don't seem to be able to get MWI to work when I store the voicemail information in a database backend, from a flat file it does work fine. I'm using rtcachefriends=yes for my sip users, per the WIKI, I'm presuming asterisk can't see these mailboxes, and therefore can't poll them to send the alerts when necessary. Is there anything that can be done to make this work properly, short of going back to a flat file for voicemail.conf? Regards Michael Baird ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Abwesenheitsnotiz: [Asterisk-Users] Nortel Option 11 and TE110P o f Digium
??? i dont understand. On 8/5/05, Siegel, Joerg [EMAIL PROTECTED] wrote: Ich bin am 9.8. wieder im Hause! Mit freundlichen Grüßen, Jörg Siegel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Uniden UIP200 Opinions
Hi, I've read through the archives, and wanted to get an updated opinion on the Uniden UIP200 phone. Seems like there were a lot of opinions that it was a good phone, but there were a few items that people were waiting for firmware updates for, but that was in 2004. I'm going to be using them in an office, 12 phones, on a LAN connected to an asterisk box. Thanks for any advice or opinions. -jim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is this echo problem down to IP Phone hardware?
Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more expensive phone - eg a Snom to see what they were like for echo. Is there something I can do with the Asterisk? codec to use? Anything? Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is this echo problem down to IP Phone hardware?
On Fri, 5 Aug 2005, Angus Comber wrote: Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more expensive phone - eg a Snom to see what they were like for echo. Is there something I can do with the Asterisk? codec to use? Anything? This has nothing to do with the IP. This is a badly designed phone. The mic is picking up the speaker. So when the end party talks, they hear themselves. Pick up a 7940/60. You will not have this issue. Michael Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is this echo problem down to IP Phone hardware?
This known as is 'acoustic echo' or 'room reverb' and involves mathematics that is quite a bit different from that used when cancelling regular 'reflected electrical signal' echos, as the signal is being acousically distorted as it echos around the room. On many handsfree handsets it doesn't manifest itself until you move into a physically large room, which increases the reflection delay and overwhelms the internal mechanisms. It would need to be handled internally by the handset or you would need to insert a hardware echo canceller capable of dealing with this type of echo, assuming your signal is exposed on a T1 somewhere. If it's IP all the way for you then you're really just down to the handset vendors as far as I know - Asterisk doesn't currently offer any form of echo cancellation on the VoIP side. Hope that helps. Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Angus Comber Sent: Friday, August 05, 2005 10:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Is this echo problem down to IP Phone hardware? Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more expensive phone - eg a Snom to see what they were like for echo. Is there something I can do with the Asterisk? codec to use? Anything? Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7914
How does one go about programming a Cisco 7914 sidecar to be used as a busy lamp field? Thanks Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7914
On Fri, 5 Aug 2005, Craig Bruenderman wrote: How does one go about programming a Cisco 7914 sidecar to be used as a busy lamp field? This can be done with SCCP only. CHeck the wiki. Thanks Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA and a PayPhone
Andres wrote: Help is on the way:) This is quite simple to achieve on Sipura units. There is a parameter called Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2) It defines the frequencies and duration of the tone. The 10 you see near the end is the duration. Simply change it to 60 like this and you're done: Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];60(*/0/1+2) I just tried it and it works like you want it. I'm not the OP and do plan on deploying several spa3k's, is there somewhere this kind of stuff is documented for the spa's? The Sipura Admin guide covers also the spa3k. The Dial Tone parameter is the same for all SPAs. You can ask your reseller for the Admin guide if you don't have it. Cheers. These are great suggestions!! I will try them on Monday! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how may channels
how many channels using codec g729 can be done by an internet bandwidth to 512kb dedicated. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 360 and firmware 4.0 problem
I have a pair of snom 360s at a customer and they were giving me Low Memory errors. The distributor suggested updating the firmware. I did that, to the one just below 4.0 (which wasn't released yet). One of the phones is still giving the Low Memory error every 3-4 days. The other one had a broken display that was just RMA'd, so it' hasn't been up long enough to know if the error occurs on that one, too. The distributor's latest suggestion was to go to the newest firmware, 4.0. I did that on the new 360 (from the RMA) and with the same account settings as the one it was replacing, it could not register with *. Since I was in a pinch, I updated the firmware down to the latest below 4.0 and the phone works just fine. Does anyone with more knowledge than I know what might be going on? Maybe a new default setting in 4.0 that's breaking things? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how may channels
how many channels using codec g729 can be used by an internet bandwidth to 512kb dedicated. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how may channels
keep approx. 32kb per channel.. -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 5 Aug 2005 11:16:32 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how may channels how many channels using codec g729 can be done by an internet bandwidth to 512kb dedicated. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: ARTCP Project (was RE: [Asterisk-Users] Features you'd liketoseeina GUI?)
The reason the system is going to be a linux distro is because it will be a complete out of the box asterisk system ready to be installed. Just like [EMAIL PROTECTED], only much much more integrated and having more features. As far as Linux not being a popular server platform? Maybe I missed something As I've said before, I'm sticking with what I know. I'm not looking to make money, or even fame. The system is a distro because it's supposed to be a distro. There will be a release of just the management/user interface but the main project will be a complete distro for installing and configuring a fullyfeatured stable asterisk environment with all the modules that the interface connects to. As far as using PHP, I know php, that's the point. There's no reason for the project to be messy. There's several huge projects out there that are php based and they're not messy at all. Zope is a python based CMS, but there's also POSTNuke and PHPNuke, which are both solid CMS's as well... I'll put it this way...We can go round and round and round on my choices of language, project type, and system. There's no point to it. What features do you want to see in the management aspect and/or the asterisk system itself? Do you wish to contribute to the project as specified? Those are the only questions at this point. I appreciate the interest, but please stop trying to change my mind on the project. It's not about money, success of the project, or fame. It's about getting the idea out of my head and into a working system, and possibly helping other people who might need it at the same time. Cheers, Sherwood McGowan --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -snacktime -Sent: Friday, August 05, 2005 1:04 PM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: ARTCP Project (was RE: [Asterisk-Users] Features -you'd liketoseeina GUI?) - -Why does the system have to be based on a linux distro? I think -that's the wrong way to go. It's one thing to create a linux -distro around a popular piece of software, but it's another -to create software that can only be used as an entire linux -distribution. - -If I were you I would take an existing application server -platform of some type that is already popular, and build a -management interface on top of that. Say something like Zope -or mod perl/Mason. Preferrably you want a platform that has -a good web application server and can also be extended using -a good general purpose programming language like Perl or -Python. Then after the core product is done add in secondary -things like backups and monitoring. - - -And although I know php is hugely popular and would make it -easy for many to contribute, I would think twice about it. -In real life it can get messy really quick on large projects, - and it's not the best general purpose programming language. -my favorite would be python, but that's just me. As for -databases use an abstraction layer. Even if you aren't -familiar with databases other than mysql, someone else will be. - -Some of these types of decisions will be what decides whether -your project goes anywhere or not. And also, be prepared to -do most of the work yourself with little help until a first -usable version is produced. Lot's of people jump on the -bandwagon once you get something going, but few will jump in -and help a lot right from the start. If you don't have the -time yourself, or can't put together at least a handful of -people that do have the time, you are kind of doomed from the -start. Everyone (like me) will be more than ready to give -you their opinions on how to do things, but you won't see -many of them when it comes time to actually do any work:) - -Good luck -___ -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] number 'register = ' in sip.conf
how many 'register =' I can have in sip.conf___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID Problems.
Hi everyone. I need to get CallerID to route incoming calls, but i keep getting this on the CLI for the callerid = -- Starting simple switch on 'Zap/1-1' Aug 5 13:18:50 ERROR[2756]: callerid.c:260 callerid_feed: fsk_serie made mylen 0 (-85) Aug 5 13:18:50 WARNING[2756]: chan_zap.c:5434 ss_thread: CallerID feed failed: Success Aug 5 13:18:50 WARNING[2756]: chan_zap.c:5476 ss_thread: CallerID returned with error on channel 'Zap/1-1' == Anyone has fixed this, thanks in advance Sincerely Otto Krumm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need Help Troubleshooting Broadvoice Connection
Tim P wrote: [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED]/2068660133 You need to add the number to the back so you can route it with asterisk. Ok I can register with BV fine (as far as I can tell from asterisk - see below). I am able to make outgoing calls but all incoming calls get a fast busy. I have opened and forwarded the following ports to my pbx: 5060-5063 UDP + TCP 69 UDP (BV claims they need this) 1-2 UDP I tried switching proxies as well, tried both LAX and CHI with the same problem. Called BV they said they can conenct andd call it with a softphone so it must be a configuration issue. Here are some outputs that might be helpful: Asterisk -r sip show registry asterisk1*CLI HostUsername Refresh State sip.broadvoice.com:5060 [EMAIL PROTECTED]23 Registered asterisk1*CLI sip show peers asterisk1*CLI Name/usernameHostDyn Nat ACL Mask Port Status bv/2068660133147.135.12.128 N 255.255.255.255 5060 Unmonitored /(Unspecified)D 255.255.255.255 0 Unmonitored 1005/1005(Unspecified)D 255.255.255.255 0 Unmonitored 1004/1004(Unspecified)D 255.255.255.255 0 Unmonitored 1003/1003(Unspecified)D 255.255.255.255 0 Unmonitored 1002/1002(Unspecified)D 255.255.255.255 0 Unmonitored [Kasterisk1*CLI sip show peer bv asterisk1*CLI * Name : bv Secret : Set MD5Secret: Not set Context : from-pstn Language : FromUser : 2068660133 FromDomain : sip.broadvoice.com Callgroup: (0) Pickupgroup : (0) Mailbox : LastMsgsSent : -1 Dynamic : No Expire : -1 seconds Expiry : 900 Insecure : Very Nat : Always ACL : No CanReinvite : No PromiscRedir : No DTMFmode : inband LastMsg : 0 ToHost : sip.broadvoice.com Addr-IP : 147.135.12.128 Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Username : 2068660133 Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw|alaw) Status : UNKNOWN Useragent: Full Contact : (not sure about that Status = UNKNOWN, is that a problem?) Get full output on outgoing calls and they connect sucessfully Get zero output on incoming calls, pbx never seem to get them Here is my sip.conf [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED] [sip.broadvoice.com] username=2068660133 user=2068660133 type=user secret=mypass nat=yes insecure=very host=sip.broadvoice.com fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband context=from-pstn authname=2068660133 Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MFC/R2 Mexico Unicall Blocked
My E1 has 10 lines from my telco, 10 lines are blocked and 20 are idle. I guess those 10 blocked are my lines(channels). Also reading this: http://voip-info.org/tiki-index.php?page=Asterisk+MFC+R2 i come to this lines: ... cas=110-124:1101 The 4 characters after the colon in the cas statements define the idle patternfor the signalling bits. For China and Thailand you should use instead of 1101. 1101 should be correct for all other countries using MFC/R2. This pattern puts the trunk in the blocked state, so when no application software is using the trunk it behaves in a sensible way. (...) You should get a green light. If you make calls into the E1 you find the E1 is blocked. This is the correct state before asterisk is started. I use 1101, that should be correct, and it should be correct to have 10 lines blocked, 1 to 10. It is also correct that those 10 lines are blocked _before_ asterisk is started. But the problem is that asterisk says those lines are still blocked when i try to simulate a call. Using libunicall-0.0.2's testcall executable i get this output, and as you can see initialli y have 10 blocked lines :-(, errors from line 11 to 31 (i only have 10 lines) and then messages of local end unblocked! for each of the 31 lines. ./testcall 2005/08/06 07:45:02 MFC/R2 Chan 1: call control(8) 2005/08/06 07:45:02 MFC/R2 Chan 1: unblock 2005/08/06 07:45:02 MFC/R2 Chan 1: 1001 - [1/4000/Idle /Idle ] 2005/08/06 07:45:02 MFC/R2 Chan 2: call control(8) 2005/08/06 07:45:02 MFC/R2 Chan 2: unblock 2005/08/06 07:45:02 MFC/R2 Chan 2: 1001 - [1/4000/Idle /Idle ] (...snip...) 2005/08/06 07:45:02 MFC/R2 Chan 30: call control(8) 2005/08/06 07:45:02 MFC/R2 Chan 30: unblock 2005/08/06 07:45:02 MFC/R2 Chan 30: 1001 - [1/4000/Idle /Idle ] 2005/08/06 07:45:02 MFC/R2 Chan 31: call control(8) 2005/08/06 07:45:02 MFC/R2 Chan 31: unblock 2005/08/06 07:45:02 MFC/R2 Chan 31: 1001 - [1/4000/Idle /Idle ] Chan 1: -- Far end blocked! :-( Chan 2: -- Far end blocked! :-( Chan 3: -- Far end blocked! :-( Chan 4: -- Far end blocked! :-( Chan 5: -- Far end blocked! :-( Chan 6: -- Far end blocked! :-( Chan 7: -- Far end blocked! :-( Chan 8: -- Far end blocked! :-( Chan 9: -- Far end blocked! :-( Chan 10: -- Far end blocked! :-( Chan 11: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern Chan 12: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern (...snip...) Chan 31: -- Protocol failure on channel 0, cause (32773) Unexpected CAS bit pattern 2005/08/06 07:45:02 MFC/R2 Chan 1: local_unblocking_expired Chan 1: -- Local end unblocked! :-) 2005/08/06 07:45:02 MFC/R2 Chan 2: local_unblocking_expired Chan 2: -- Local end unblocked! :-) 2005/08/06 07:45:02 MFC/R2 Chan 3: local_unblocking_expired Chan 3: -- Local end unblocked! :-) (...snip...) 2005/08/06 07:45:02 MFC/R2 Chan 30: local_unblocking_expired Chan 30: -- Local end unblocked! :-) 2005/08/06 07:45:02 MFC/R2 Chan 31: local_unblocking_expired Chan 31: -- Local end unblocked! :-) Athiel E. Criollo Merino wrote: Seems like your carrier is assigning you channels from 11 and up to make calls, why dont test making a definition for group 1 from lines 11 to 20... Regards In spanish. Ariel, parece que Telmex te está asignando los timeslots 11 al 20 para tus lineas. por que no pruebas con tu grupo 1, asignandole solamente desde la linea o timeslot 11 hasta el 20. No se mucho sobre señalizacion, pero tiene algo de logica lo que te estoy diciendo. Suerte. Athiel Criollo 2005/8/3, Ariel Molina R. [EMAIL PROTECTED]: I've been trying to configure an E1 in Mexico using unicall, i went into vozdigital, googled this list, and finally followed this instructions: http://voip-info.org/tiki-index.php?page=Asterisk+MFC+R2 I have 10 PSTN numbers and 10 lines assigned, so i only have 10 channels assigned from my telco. However when i try to simulate a call using this call file: call file-- Channel: UniCall/g1/1 Callerid: 4772140099 MaxRetries: 0 RetryTime: 600 WaitTime: 600 Context: principal_in Extension: 014433988789 Priority: 1 -- I get this messages -- Aug 4 11:46:06 WARNING[9420]: chan_unicall.c:1240 unicall_call: Make Call failed - Blocked Aug 4 11:46:06 NOTICE[9420]: channel.c:1827 __ast_request_and_dial: Unable to request channel UniCall/g1/1 -- Hungup 'UniCall/11-1' Aug 4 11:46:06 NOTICE[9420]: pbx_spool.c:229 attempt_thread: Call failed to go through, reason 0 -- So i can see Unicall channels are configured but blocked (as UC show channel). There is not much info about unicall so i require your advice, what can i do? Where do i look? Also i constantly receive messages Aug 4 11:54:07 WARNING[9402]:
Re: [Asterisk-Users] Is this echo problem down to IP Phone hardware?
Kris Boutilier wrote: This known as is 'acoustic echo' or 'room reverb' and involves mathematics that is quite a bit different from that used when cancelling regular 'reflected electrical signal' echos, as the signal is being acousically distorted as it echos around the room. On many handsfree handsets it doesn't manifest itself until you move into a physically large room, which increases the reflection delay and overwhelms the internal mechanisms. The maths is exactly the same. However, it is certainly true that a lot of acoustic echo cancellers don't deal with long enough echoes to be effective in large spaces. It would need to be handled internally by the handset or you would need to insert a hardware echo canceller capable of dealing with this type of echo, assuming your signal is exposed on a T1 somewhere. If it's IP all the way for you then you're really just down to the handset vendors as far as I know - Asterisk doesn't currently offer any form of echo cancellation on the VoIP side. In the IP world the echo must be killed by the phone itself. You cannot echo cancel on the IP side of a switch like Asterisk. The echo path length needs to be constant for any known echo cancellation process to work. IP path lengths are not constant. Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. The Grandstreams are much maligned, but they actually do a better job in this area than most products. As said above, if you are using this in a large space the echo canceller in the phone may not cancel a long enough echo to be very effective. If it fails to kill the echo in a small room something is wrong. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more expensive phone - eg a Snom to see what they were like for echo. Is there something I can do with the Asterisk? codec to use? Anything? A snom might be a poor choice. People tell me they don't even echo cancel the handset. If a hard of hearing user turns up the handset volume the caller hears considerable echo. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how may channels
http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption On 8/5/05, Innocent Evil [EMAIL PROTECTED] wrote: keep approx. 32kb per channel.. -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 5 Aug 2005 11:16:32 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how may channels how many channels using codec g729 can be done by an internet bandwidth to 512kb dedicated. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7914
On Fri, 2005-08-05 at 14:09 -0400, Craig Bruenderman wrote: How does one go about programming a Cisco 7914 sidecar to be used as a busy lamp field? In the sccp.conf file, o As a Line: You can assign a line to the button/lamp which is really neat. The lamp is green when you are on the line, blinking green when you put the line on hold, blinks orange when you call that line. If you had a 7960 and wanted a line on the 7914 you could do it this way: autologin = ,,79140,79141 ; This makes it go to the 7th button ; for the first line button. o As a speed dial (lamp is either off or red) You setup a speed dial like this: speeddial = 10,John Doe,[EMAIL PROTECTED] And then to make this work you need to have the exten = 10,hint,SCCP/10 ;sccp phone exten = 10,hint,SIP/10 ;Sip phone As a line you don't need the hint. Note: a speed dial with hint shows an icon of a phone just like a line. And when the lamp is on with the hint, it shows an icon of a phone with an X though it. This is the case with the 7940/7960 speed dials as well. The icon gives you the same line status as the lamps even without the 7914 sidecar. Thanks Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some echo?
Robbie: I fought with echocancel and various parameters for a long time with little luck. Then I uncommented AGGRESSIVE_SUPPRESSOR and DISABLED the Fax/tone detection in in zconfig.h since we're not faxing via Asterisk. Recompiled and all echo disappeared. Hope that helps. -Rob -- Robert Goodyear Brand Up LLC http://www.brand-up.com On Aug 4, 2005, at 2:10 PM, Robbie Hughes wrote: I have a 12 channel PRI with SNOM 190's and asterisk CVS from January. Most calls are fine, all incoming calls are fine, but I am getting echo on a significant number of outgoing calls. The person on the other side hears a perfect call, but the SIPphone side gets to hear themselves. It happens 100% of the time to some numbers (outgoing only), and only sporadically to others. Has anyone ever experienced this? the RTT to the phones from the server is less than 10ms and it is a 100mbit network with no traffic and cisco switches. zapata.conf attached below: Note: The commented out gain of +2 on outgoing seems to make no difference to the effect. Has anyone got any ideas? ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] group = 1,16 [channels] spanmap = 1,1,1 language=en context=from-pstn rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 ;txgain=2.0 txgain=0.0 rxgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no pridialplan=unknown overlapdial=yes signalling=pri_cpe switchtype=euroisdn channel= 1-12 faxdetect=both ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and firmware 4.0 problem
We have experienced some Snom firmware issues, although the are not related to the symptoms you describe. We found that the sidecards will not power on unless the 360 host phone is running the latest firmware rev. Cory Andrews Purchasing / EVP VOIPSupply.com v – 716.630.1555 X22 e – [EMAIL PROTECTED] Michael George wrote: I have a pair of snom 360s at a customer and they were giving me Low Memory errors. The distributor suggested updating the firmware. I did that, to the one just below 4.0 (which wasn't released yet). One of the phones is still giving the Low Memory error every 3-4 days. The other one had a broken display that was just RMA'd, so it' hasn't been up long enough to know if the error occurs on that one, too. The distributor's latest suggestion was to go to the newest firmware, 4.0. I did that on the new 360 (from the RMA) and with the same account settings as the one it was replacing, it could not register with *. Since I was in a pinch, I updated the firmware down to the latest below 4.0 and the phone works just fine. Does anyone with more knowledge than I know what might be going on? Maybe a new default setting in 4.0 that's breaking things? Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
