[asterisk-users] Call Failed
After so many rings when the party does not answer, my SIP phone says Call Failed. Why doesn't it just keep ringing? Here's the dial plan rule: exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],,r) exten = _NX,n,Hangup() ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Include zaptel in kernel...
Is there any plan to include the zaptel drivers into the main Linux kernel? If not, there should be one. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Someone please explain FXO/FXS module/channel relationships in Zaptel configuration to me
Please explain the relationship between modules from the driver (wctdm), the /etc/zaptel.conf file and zapata.conf. Specifically, if I have a FXS module 0 and FXO module 1, what should be used in zaptel.conf and what should be used in zapata.conf? Then finally, in extensions.conf, what is the Zap channel for dialing out? Zap/? % dmesg Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Not installed Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) % cat /etc/zaptel.conf fxoks=1 fxsks=2 % cat zapata.conf ... signalling=fxo_ks context=outgoing-analog echocancel=yes callerid=asreceived channel = 1 signalling=fxs_ks context=incoming-analog echocancel=yes callerid=asreceived channel = 2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: incoming zaptel calls fail
On Apr 11, 2007, at 2:38 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Try to update your zaptel to latest 1.4 svn. I just fixed a bug in a patch that was committed not too long ago. It should fix it. Thanks. I will try that. I did start using the 1.4.2 tar release to get things working. It worked for nearly a week and then today it stopped working: [Apr 16 20:34:42] ERROR[10238]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-27) [Apr 16 20:34:42] WARNING[10238]: chan_zap.c:6359 ss_thread: CallerID feed failed: Success [Apr 16 20:34:42] WARNING[10238]: chan_zap.c:6459 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 16 20:34:44] WARNING[10238]: chan_zap.c:4027 zt_handle_event: Ring/Off-hook in strange state 6 on channel 2 [Apr 16 20:35:33] NOTICE[10244]: chan_zap.c:6327 ss_thread: Got event 18 (Ring Begin)... [Apr 16 20:35:45] NOTICE[10248]: chan_zap.c:6327 ss_thread: Got event 18 (Ring Begin)... [Apr 16 20:35:53] NOTICE[10253]: chan_zap.c:6327 ss_thread: Got event 17 (Polarity Reversal)... [Apr 16 20:36:01] WARNING[10253]: chan_zap.c:6459 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 16 20:36:03] WARNING[10253]: chan_zap.c:4027 zt_handle_event: Ring/Off-hook in strange state 6 on channel 2 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming zaptel calls fail
Using the latest SVN of zaptel and asterisk, I can no longer receive incoming analog calls. The caller just hears it ringing but Asterisk doesn't pick up. I am seeing these error messages: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:39:34] WARNING[16580]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' and also: % dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.4-r2397M Zaptel Echo Canceller: MG2 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Not installed Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) Registered tone zone 0 (United States / North America) Zaptel Transcoder support loaded zaptel.c:764 (pid 3176: asterisk) got signal 8000 zaptel.c:764 (pid 16469: asterisk) got signal 8100 What could be the problem? How can I fix it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: incoming zaptel calls fail
Neglected to mention the host operating system: Linux myhost 2.6.20-1.2307.fc5 #1 Sun Mar 18 20:44:48 EDT 2007 i686 i686 i386 GNU/Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: incoming zaptel calls fail
On Apr 9, 2007, at 1:51 PM, Kevin P. Fleming [EMAIL PROTECTED] wrote: You also neglected to mention the version of Asterisk you are running; 'latest SVN' means nothing when there are 20+ branches of Asterisk on our SVN server. Sorry about that. It is the 1.4 trunk: Asterisk SVN-branch-1.4-r60850, Copyright (C) 1999 - 2006 Digium, Inc. and others. and to recap: OS: Fedora Core 5: Linux myhost 2.6.20-1.2307.fc5 #1 Sun Mar 18 20:44:48 EDT 2007 i686 i686 i386 GNU/Linux ZAPTEL: Zaptel Version: SVN-branch-1.4-r2397M ERROR MSGS: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:39:34] WARNING[16580]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.4 svn fails to compile (xbus-core.c)
This compile error started happening about 2 weeks ago with zaptel. /mysrc/asterisk/zaptel-1.4/xpp/xbus-core.c: In function ‘debugfs_open’: /mysrc/asterisk/zaptel-1.4/xpp/xbus-core.c:171: error: ‘struct inode’ has no member named ‘u’ make[4]: *** [/mysrc/asterisk/zaptel-1.4/xpp/xbus-core.o] Error 1 BTW - Is there a separate zaptel mailing list? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.4 svn doesn't compile
Is there a zaptel mailing list? Here's the error: CC [M] zaptel-1.4/xpp/xbus-core.o zaptel-1.4/xpp/xbus-core.c: In function ‘debugfs_open’: zaptel-1.4/xpp/xbus-core.c:171: error: ‘struct inode’ has no member named ‘u’___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 svn voicemail broken?
Ever since upgrading to 1.4 SVN, the advanced options on voicemail have disappeared. When I press 3 for advanced options, it just reviews the message. It used to present me with a menu to 1 = reply, 2 = call the person back, 3 = play message envelope. What gives? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attention all Aastra IP phone users...
If you own Aastra phones, here's a group dedicated to your specific needs. BTW - The Asterisk-users mailing list is great but it has way too much volume to be useful for specific problems. It needs to be broken up into smaller more manageable lists. Homepage: http://groups.google.com/group/aastra-asterisk-users Group email:[EMAIL PROTECTED] Description: Aastra-asterisk-users is a group where owners of Aastra IP telephones can discuss tips and issues with Asterisk-based PBX systems. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 svn voicemail bug / crash
I cannot access my voicemail and get the following warning in my console: [Nov 25 10:26:43] WARNING[5628]: app.c:935 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/8900/Old': File exists I have also noticed that Asterisk will crash several minutes later after this warning message. I am using the latest SVN 1.4 branch of Asterisk (Revision 48007) and Zaptel (Revision 1640) on Fedora Core 5 (2.6.18-1.2239.fc5) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 svn voicemail bug / crash
I retested this with 1.4.0-beta3 and I still can't access my voicemail. I dial the voicemail extension and I just get silence for a few seconds and it hangs up. HELP! I have 295 messages in my old mailbox and I want to retrieve my new messages. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SOLVED - 1.4 svn voicemail bug / crash
There was a stale lock file in the mailbox directory. This is a bug though. Asterisk should clean up all lock files on startup. Lastly, I can't explain the intermittent crash and wasn't able to catch it using gdb either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-addons 1.4 SVN fails to compile
It seems like asterisk-addons in SVN has been broken for the last few weeks: gcc -DHAVE_CONFIG_H -I. -I. -I. -I./ooh323c/src -I./ooh323c/src/h323 - DGNU -D_GNU_SOURCE -D_REENTRANT -D_COMPACT -c src/chan_h323.c -MT chan_h323.lo -MD -MP -MF .deps/chan_h323.TPlo -fPIC -DPIC -o .libs/ chan_h323.lo src/chan_h323.c: In function 'ooh323_new': src/chan_h323.c:250: error: too few arguments to function 'ast_channel_alloc' ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Speeding up SayDigits?
I would like SayDigits to say a phone number faster. Is there a way to control the speed? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail idea and a question
When you listen to old messages, it would be better if Asterisk reversed the order so that it starts at the most recent message and then forwarding goes to the next oldest message, etc... The last message would be the oldest. This makes more sense for old messages. Also, is there a way to have it so that after one message plays, the next one plays automatically without having to press 6? This would be very useful when checking your messages remotely say from a handsfree car phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while
On Oct 19, 2006, at 3:00 PM, [EMAIL PROTECTED] wrote:Date: Thu, 19 Oct 2006 09:30:38 -0500 From: "Eric \"ManxPower\" Wieling" [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Do you have callprogress=yes or busydetect=yes in your /etc/asterisk/zapata.conf ? No. They are not set. i.e. default___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Date: Thu, 19 Oct 2006 09:30:38 -0500 From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Robert La Ferla wrote: I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the analog line on the FXO, * dials all the extensions (SIP and analog.) I have a Digium card with 1 FXO and 1 FXS. Do you have callprogress=yes or busydetect=yes in your /etc/asterisk/zapata.conf ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk hangs up on incoming analog calls after a while
I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the analog line on the FXO, * dials all the extensions (SIP and analog.) I have a Digium card with 1 FXO and 1 FXS. How can I diagnose this problem? Has anyone experienced anything similar? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5
I found an init.d script for asterisk BUT not for asterisk/zaptel modules. I'm still looking for a good solution. It seems to me that the correct solution would involve /etc/modprobe.d/modpobe.conf. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] modprobe wctdm fails in /etc/rc.local on FC5
If I boot my server and manually type modprobe wctdm, it correctly loads both wctdm and zaptel. If I put the modprobe in /etc/rc.local and reboot, it fails. Why? I am running the latest svn source of zaptel on Fedora Core 5 (w/latest updates as of 8/15) Here are the error messages from modprobe: Aug 15 15:55:39 WARNING[1860] chan_zap.c: Unable to specify channel 1: No such device or address Aug 15 15:55:39 ERROR[1860] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Aug 15 15:55:39 ERROR[1860] chan_zap.c: Unable to register channel '1' Aug 15 15:55:39 WARNING[1860] loader.c: chan_zap.so: load_module failed, returning -1 Aug 15 15:55:39 WARNING[1860] loader.c: Loading module chan_zap.so failed! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5
Can someone send me their modprobe.conf file? I think that may be the problem. A zaptel file is created during install in /etc/ modprobe.d but modprobe.conf must need to reference it... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No ringing on outgoing SIP calls.
When I dial out, I can't hear any ringing. I am using the latest SVN code (SVN-branch-1.2-r37458M). Is this a problem with Asterisk? Or with my VOIP provider? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] to.gsm and the.gsm
Can someone send me a link to a GSM sound file (US-English) for the words to and the? BTW - These should be put in the standard asterisk-sounds distribution. I couldn't find them in mine or in the SVN repository. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to ask for number to dial and then dial it?
I want to create an extension say 8000 that prompts the user to
enter a number and then dial that entered number according to a set
of rules. The rules for dialing out are in different context (dial-
out-rules).
[mymenu]
exten = 8000,1,Answer()
[dial-out-rules]
; toll-free numbers out pots line
exten = _1800XXX,1,Dial(${ANALOG_POTS}/${EXTEN})
exten = _1800XXX,n,Hangup()
; long-distance out voip line
exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],30)
exten = _NX,n,Hangup()
etc...
How do I do it?
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[Asterisk-Users] Latest SVN of asterisk-addons doesn't compile
build_tools/mkdep -fPIC -fPIC app_addon_sql_mysql.c app_saycountpl.c cdr_addon_mysql.c res_config_mysql.c app_addon_sql_mysql.c:15:22: error: asterisk.h: No such file or directory app_saycountpl.c:10:22: error: asterisk.h: No such file or directory cdr_addon_mysql.c:22:22: error: asterisk.h: No such file or directory res_config_mysql.c:41:22: error: asterisk.h: No such file or directory gcc -fPIC -fPIC -c -o app_saycountpl.o app_saycountpl.c app_saycountpl.c:10:22: error: asterisk.h: No such file or directory In file included from /usr/include/asterisk/linkedlists.h:23, from /usr/include/asterisk/chanvars.h:26, from /usr/include/asterisk/channel.h:111, from /usr/include/asterisk/file.h:30, from app_saycountpl.c:13: /usr/include/asterisk/lock.h: In function ‘ast_mutex_init’: /usr/include/asterisk/lock.h:534: error: ‘PTHREAD_MUTEX_RECURSIVE’ undeclared (first use in this function) /usr/include/asterisk/lock.h:534: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:534: error: for each function it appears in.) $ uname -a Linux localhost 2.6.16-1.2133_FC5 #1 Tue Jun 6 00:52:14 EDT 2006 i686 i686 i386 GNU/Linux $ svn update Fetching external item into 'menuselect' External at revision 17. Fetching external item into 'mxml' External at revision 3. At revision 254. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk hangs up on incoming PSTN line to analog extension
I have encountered the following problem with the latest Asterisk source (as of 4/23/2006): Someone calls me on my PSTN line, it then dials my analog extension (I have both SIP and analog phones where all analog phones are a shared extension.) After a while, I get a busy signal. How can I further diagnose this? What could be the problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Users Mailing List Traffic
The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for docs on adjusting txgain/rxgain
I am looking for docs on how to diagnose and adjust the rx/tx gain in zapata.conf. The wiki has a link to this article but it no longer exists on the server. http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival tts
Steven [EMAIL PROTECTED] wrote: Hi I have installed Festival on the same box as asterisk and followed the instructions to integrate it with asterisk. Festival seems to work fine on its own performing text to speech from the command line or via a file. Asterisk answers the call but there is no speech. I can see no errors in the Festival log file I asked the same question to this list a while back but got no replies. What OS are you using? How did you install Festival? What version of *? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting to the last old voicemail message
If you have many old voicemail messages, to get to the most recent one, you have to keep hitting 6 until you reach the last one. It would be better if you could hit 4 from the first message to get to the last message and/or have a digit that takes you the first and last messages respectively. Anyone have any patches for this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Getting to the last old voicemail message
I made a small change to apps/app_voicemail.c to permit circular
navigation when listening to messages. If you are at the first message,
and press 4, it takes you to the last message. If you are already at
the last message and press 6, it takes you to the first message. I
did a quick test and it seems to work. If you apply it and find any
problems, please let me know and I'll fix it.
I don't have a diff but here's the code in function vm_execmain():
case '4':
if (vms.curmsg) {
vms.curmsg--;
cmd = play_message(chan, vmu, vms);
} else {
/* cmd = ast_play_and_wait(chan, vm-nomore); */
vms.curmsg = vms.lastmsg;
cmd = play_message(chan, vmu, vms);
}
break;
case '6':
if (vms.curmsg vms.lastmsg) {
vms.curmsg++;
cmd = play_message(chan, vmu, vms);
} else {
/* cmd = ast_play_and_wait(chan, vm-nomore); */
vms.curmsg = 0;
cmd = play_message(chan, vmu, vms);
}
break;
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[Asterisk-Users] Adjusting gain, Milliwatt and ztmonitor
I have been trying to adjust the gain as per this document without any success: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html I have a PSTN and VoIP (SIP) connection via *. I disabled all echo cancel/training in zapata.conf and set tx/rxgain to 0. I then changed my extensions.conf so that when I call the VoIP line from the PSTN line, it plays the Milliwatt application tone. However, when I call, I don't hear the tone and ztmonitor doesn't change at all. What could be the problem? I am running the latest svn source, Aastra 9133i sip phone and Digium TDM11B. BTW - My PSTN is thru Verizon (MA) and I don't have their test tone #. If you know it, please e-mail it to me. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AlarmReceiver?
Anyone using the AlarmReceiver? Does it work? Mine doesn't seem to communicate properly. How can I tweak the DTMF settings? Is it in the zaptel.conf or somewhere else?? -- Executing AlarmReceiver(Zap/1-1, ) in new stack AlarmReceiver: Setting read and write formats to ULAW AlarmReceiver: Answering channel AlarmReceiver: Waiting for connection to stabilize AlarmReceiver: Waiting for first event from panel AlarmReceiver: Sending 1400Hz 100ms burst (ACK) AlarmReceiver: Sending 2300Hz 100ms burst (ACK) AlarmReceiver: DTMF Digit Timeout on Zap/1-1 == AlarmReceiver: Incomplete string: , trying again... AlarmReceiver: Sending 1400Hz 100ms burst (ACK) AlarmReceiver: Sending 2300Hz 100ms burst (ACK) AlarmReceiver: DTMF Digit Timeout on Zap/1-1 == AlarmReceiver: Incomplete string: , trying again... AlarmReceiver: Sending 1400Hz 100ms burst (ACK) AlarmReceiver: Sending 2300Hz 100ms burst (ACK) AlarmReceiver: DTMF Digit Timeout on Zap/1-1 == AlarmReceiver: Incomplete string: , trying again... AlarmReceiver: Sending 1400Hz 100ms burst (ACK) AlarmReceiver: Sending 2300Hz 100ms burst (ACK) AlarmReceiver: DTMF Digit Timeout on Zap/1-1 == AlarmReceiver: Incomplete string: , trying again... AlarmReceiver: Sending 1400Hz 100ms burst (ACK) AlarmReceiver: Sending 2300Hz 100ms burst (ACK) AlarmReceiver: DTMF Digit Timeout on Zap/1-1 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 removal
Kevin Bockman wrote: If you are using 1.2, I would use native (codec, not MP3). There should be an example in the sample config file in /usr/src/asterisk/configs/musiconhold.conf.sample - I don't see it on the Wiki. It should be there, somewhere. Must be buried. For this option, you will need to have the sound files in .ul, .gsm, or whatever codec you use mostly. I only allow ulaw, so all of my MOH files are .ul. This way it doesn't have to transcode at all. How can you convert mp3 to gsm? mencoder? Do you have an example? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 removal
Chris Albertson wrote: Second even if there were one the mpg123 process is not long lived. A new one is started for each MOH session. I hate to say it but there is a problem where the mpg123 process never terminates. This occurs with the latest SVN-branch-1.2-r7917 version and has been doing this for at least a few weeks now. I just uninstalled it tonight because of this. I am in the process of switching over to the format_mp3 addon. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Jobs
Douglas Garstang wrote: I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? I think that the Asterisk customer profile can shed some light on this. If you are a big company, you'll buy into an expensive system because you can afford it and rely on it. If you are a small company, you will look to Asterisk as an inexpensive way to set up your telephone system. You will also likely have staff that is willing to work with it and not enough money or need to hire external consultants exclusively for Asterisk. You may have a telecom or networking consultant that will put together the network and set up the system but Asterisk is a small piece of it. I'd say Asterisk is more of a plus in a job description but not a requirement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra 9133i and NAT: Can it work?
Jamie J. Begin wrote: I've been pulling out my hair all day on this one. If anyone can help, I'd really appreciate it. :-( I've got an Aastra 9133i (with the latest firmware version) and a Cisco 7960 sitting behind a NAT device on my LAN. The Asterisk server is hosted offsite and has a public IP address. I've set up port-forwarding on the firewall for both phones to tunnel the SIP messages initiated by the Asterisk box. It works like a charm with the Cisco phone by using the following config info: voip_control_port: 5077 nat_enable: 1 nat_address: nat_received_processing: 0 Every time the Cisco phone registers with Asterisk, it does so using port 5077 and with the corresponding port-forwarding rule added to the firewall, it works great. However, for the life of me, I can't get the Aastra to do the same thing. I thought that should be able to do it by using sip nat port: 5078 in the Aastra config file, but no dice. Incoming calls work for about two minutes after the phone is rebooted and then it loses its registration. Any suggestions? Using your browser, type http:// plus the IP address of the 9133i and login to the phone. Then select Global SIP from the menu. Go down to the Registration Period and put 100 in it. That should fix it. If not, try the NAT settings under the Network menu. If it still doesn't work post your settings so everyone can look at them. BTW - I have a 9133i working behind a NAT but * is also behind it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trailing silence in voicemail messages
Is there some way * can trim the trailing silence in a voicemail message? There's the maxsilence setting for silence detection which is related to what I'm asking but not the same. Let's say I set the maxsilence to 8 seconds. During the recording of a voicemail, if someone doesn't say anything for 8 seconds, the recording ends. However, the recording still has 8 seconds of silence at the end. I'm asking how to automatically chop off that last 8 seconds of silence. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FC3 or FC1 (or something else?)
Brett, Gary wrote: My question is which OS would be preferred in this configuration Fedora Core 1 or Fedora Core 3, and are there any install guides out there that are recent enough for asterisk 1.2 Try Fedora Core 4 (FC4). Works great. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regular Crashes
Did you try running * under gdb? When it crashes, do a bt to get a back trace and post it to the mailing list. e.g. % gdb /usr/sbin/asterisk GNU gdb Red Hat Linux (6.3.0.0-1.84rh) Copyright 2004 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type show copying to see the conditions. There is absolutely no warranty for GDB. Type show warranty for details. This GDB was configured as i386-redhat-linux-gnu...Using host libthread_db library /lib/libthread_db.so.1. (gdb) run wait for crash (gdb) bt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FC3 or FC1 (or something else?)
Michael Stearne wrote: I am having trouble with FC3. After doing a yum update (of 1264 packages) I still cannont compile 1.2.1 from source: make[1]: `libedit.a' is up to date. make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline' make[1]: Entering directory `/usr/src/asterisk-1.2.1/db1-ast' make[1]: `libdb1.a' is up to date. make[1]: Leaving directory `/usr/src/asterisk-1.2.1/db1-ast' make[1]: Entering directory `/usr/src/asterisk-1.2.1/stdtime' make[1]: *** No rule to make target `/usr/lib/gcc/i386-redhat-linux/3.4.2/include/stddef.h', needed by `localtime.o'. Stop. make[1]: Leaving directory `/usr/src/asterisk-1.2.1/stdtime' make: *** [stdtime/libtime.a] Error 2 and when I try to update from binary: [EMAIL PROTECTED] ~]# rpm -Uvh asterisk-1.2.1-15.rhfc3.at.i386.rpm warning: asterisk-1.2.1-15.rhfc3.at.i386.rpm: V3 DSA signature: NOKEY, key ID 66534c2b error: Failed dependencies: libpri.so.1 is needed by asterisk-1.2.1-15.rhfc3.at.i386 libspandsp.so.0 is needed by asterisk-1.2.1-15.rhfc3.at.i386 libtonezone.so.1 is needed by asterisk-1.2.1-15.rhfc3.at.i386 I have compiled from source 1.0.9 without problem on this machine. Any ideas why my attempts are now failing? To use the binary, it appears that you need libpri. Look at http://www.asterisk.org (top right hand side) Did you try getting the latest source via svn subversion? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival clicks instead of sound and disconnects.
When I dial a festival extension (1222), all I hear is a series of fast clicks and then it hangups. I do not have a sound card installed but I would think I don't need one. Is a sound card necessary? Should I use a script instead of the scheme code? Can someone who has this working send me their festival configuration? Thanks! CONFIG INFO: I have festival 1.95 installed on Fedora Core 4. % rpm -qa | grep fest festival-1.95-3 % cat festival.conf [general] host=localhost port=1314 usecache=yes cachedir=/var/lib/asterisk/festivalcache/ festivalcommand=(tts_textasterisk %s 'file)(quit)\n I added this to the festival.scm: ;;; Command for Asterisk begin (define (tts_textasterisk string mode) (tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe functions. (utt.send.wave.client (utt.wave.resample (utt.wave.rescale (utt.synth (eval (list 'Utterance 'Text string))) 5) 8000))) ;;; Command for Asterisk end I start the festival server BEFORE Asterisk on the same server: % festival_server -p 1314 -l /tmp I added this test extension: ; testing festival (text-to-speech app) exten = 1222,1,Answer() exten = 1222,n,Wait(1) exten = 1222,n,Festival(mary had a little lamb) ; do NOT use quotes around the string! if you use commas, you will have to escape them with a \ (backslash). exten = 1222,n,Hangup() When I dial 1222, all I hear is a click and then hangup. % festival --server /etc/myfestconfig serverMon Jan 2 15:27:37 2006 : Festival server started on port 1314 client(1) Mon Jan 2 15:27:51 2006 : accepted from localhost client(1) Mon Jan 2 15:27:51 2006 : disconnected ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival clicks instead of sound and disconnects.
Let me add that text2wave works fine. Something is wrong with the Asterisk = Festival server communications. Ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] To write Sphinx Interface in EAGI or app_xxx.c?
You must have seen this page but maybe not because the site was down for a while recently: http://turnkey-solution.com/asterisk-sphinx.html I am also interested in getting Sphinx to work with Asterisk. Please report anything you find. I know that there are a few different versions of Sphinx (v2, v3, v4) so I'm curious what versions work with *. Sphinx 4 sounds like it is not only the best performer (in terms of accuracy and speed) but also it's in Java so it's easier to modify. Does Sphinx 4 work with *? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Got 200 OK on REGISTER that isn't a register
What does this warning mean? WARNING[11065]: chan_sip.c:9596 handle_response_register: Got 200 OK on REGISTER that isn't a register ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wctdm module goes missing after a reboot - Gentoo?
Moises Silva wrote: Hello Ryan. Check out the file /etc/modules.conf, /etc/modules.d/zaptel ... if for some reason you have empty the modules.conf, modules-update force will fix it, tough. In order to provide you with further help, please provide more clues. What about systems that use /etc/modprobe.conf? depmod should handle it but it doesn't work when I tried it either but then again I haven't given this a lot of thought... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Go directly to new messagesfromVoiceMailMain?
Alexander Lopez wrote: I vote for 'a' as the auto-play option. http://bugs.digium.com/view.php?id=6090 In thinking about this more, the auto-play option can be a quickie fix but a more complete implementation is needed. Think about the scenario when checking your voicemail from an automobile or a crowded subway where you can't easily dial. Wouldn't it be great if you could dial in to your * box and if there are no new messages, it tells you Sorry, you have no new messages. and the call is terminated. If you have new messages, it plays ALL of them back to back without the need to press anything. If you do press a key, it takes you to the standard menu where you have full navigation. At the end of all the new messages, it hangs up. This way you can, by default, hear all of your messages without having to navigate anything. I can think of some additional options but this core functionality is a good start. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] name that vendor...
[EMAIL PROTECTED] wrote: The seller refuses to tell me who the vendor is. That should send up the big red flag to not buy anything from that seller. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Go directly to new messages from VoiceMailMain?
Tomislav Parcina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
I want to create an extension that goes directly to my new messages
without having to press 1. How do I do that? I can call
VoiceMailMain but then I have to choose 1 from the menu. I'd like it
to go there and play the first message or say There are no new
messages and hangup. How can I do this?
exten = 298,1,Ringing
exten = 298,2,Wait(2)
exten = 298,3,VoiceMailMain(s${CALLERIDNUM}) ; if pass is the same lik
extension number
Thanks but that doesn't address my problem. The s option is for the
password. I'm asking about how to get it to play new messages right
away without the user having to press 1
___
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[Asterisk-Users] What does Page application do?
Why would you use this? Can someone please elaborate on the below description? I'm missing the intent of it. localhost*CLI show application Page [Synopsis] Pages phones [Description] Page(Technology/ResourceTechnology2/Resource2[|options]) Places outbound calls to the given technology / resource and dumps them into a conference bridge as muted participants. The original caller is dumped into the conference as a speaker and the room is destroyed when the original caller leaves. Valid options are: d - full duplex audio q - quiet, do not play beep to caller ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Page application do?
So I can set it up to call a bunch of extensions and broadcast a message to them without the user picking up? Can I do this with Aastra phones? This would be great for announcing incoming calls. You have a call from XXX . Press 1 to pickup Press 2 to send them to voicemail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] silent dial/ring?
Is it possible to dial with a silent ring? If so, is it configurable with * or does the phone have to support it? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Go directly to new messagesfromVoiceMailMain?
Alexander Lopez wrote: I vote for 'a' as the auto-play option. http://bugs.digium.com/view.php?id=6090 I second the vote. I thought of using the same letter after reading your reply. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Page application do?
Andrew Latham wrote: I think most all of the phones have a hack to get it working. Aastra analog ADSI phones even work as I read some where... It works with the 9133i. This is a great feature! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SetAccount missing?
William M. Sandiford wrote: I just upgraded my system to the latest svn-trunk I previously made extensive use of the SetAccount() function, but now I'm getting the following error Dec 29 20:54:08 WARNING[4925]: pbx.c:1679 pbx_extension_helper: No application 'SetAccount' for extension (voipsubscriber-in, x, 100) Has this function been deprecated? If so, what method is used to replace its functionality. I have noticed a lot of deprecated features, variables, etc, and the wiki usually explains that the application / variable is deprecated and what to use in its replacement. The wiki entry for set account doesn't say anything http://www.voip-info.org/wiki-Asterisk+cmd+SetAccount Any Ideas, as you can see...its missing Just a guess but did you try: Set(ACCOUNTCODE=xxx) ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Go directly to new messages from VoiceMailMain?
I want to create an extension that goes directly to my new messages without having to press 1. How do I do that? I can call VoiceMailMain but then I have to choose 1 from the menu. I'd like it to go there and play the first message or say There are no new messages and hangup. How can I do this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk seg fault (SVN-branch-1.2-r7641)
-- Executing VoiceMail(SIP/999-e59b, 500|g4) in new stack Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1212703824 (LWP 4440)] 0x003d0110 in rawmemchr () from /lib/libc.so.6 (gdb) bt #0 0x003d0110 in rawmemchr () from /lib/libc.so.6 #1 0x003c582b in _IO_str_init_static_internal () from /lib/libc.so.6 #2 0x003bb657 in vsscanf () from /lib/libc.so.6 #3 0x003b6bf2 in sscanf () from /lib/libc.so.6 #4 0x0055e87d in vm_exec (chan=0x8b92fc8, data=0xb7b76fe8) at app_voicemail.c:5501 #5 0x0808d033 in pbx_extension_helper (c=0x8b92fc8, con=Variable con is not available. ) at pbx.c:544 #6 0x0808e4d4 in __ast_pbx_run (c=0x8b92fc8) at pbx.c:2220 #7 0x0808f0dc in pbx_thread (data=0x8b92fc8) at pbx.c:2507 #8 0x004d6b80 in start_thread () from /lib/libpthread.so.0 #9 0x0042e9ce in clone () from /lib/libc.so.6 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with date time on Aastra 480i since release 1.3
Jacques Leisy wrote: Thanks Robert. I tried of course with time server disabled: 0 too. Is it working for you? Which time server are you using, an external one? Works for me and I'm using an internal one which is then synced to an external one. Try ONLY these entries. Remove the time format and date format and backup ntp servers: time server disabled: 0 time server1: 192.168.0.10 If this doesn't work, you should check your firewall rules (if any) and the versions of ntpd (4.2?) that you are running. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check Digium TE410P firmware version?
Franz Wu wrote: I have one TE410P and want to know how to. Sending back to Digium should be a good idea. Is it possible to upgrade the firmware for a TDM400P? If so, where do you download new versions and what's the upgrade procedure? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iptables rules for forwarding SIP/RTP to Asterisk server from behind nat firewall/router
Can someone please send me your iptables rules for forwarding SIP/RTP udp to your * server? I tried this but I think I need more rules like DNAT or something... iptables -A FORWARD -i $EXT_IF -o $INT_IF -p udp -m udp --sport 5060 -d $ASTERISK_IP --dport 5060 -j ACCEPT iptables -A FORWARD -i $EXT_IF -o $INT_IF -p udp -m udp --sport 1:2 -d $ASTERISK_IP --dport 1:2 -j ACCEPT ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy signal for incoming calls from broadvoice
The solution lies in sip.conf and extensions.conf. BroadVoice's instructions are incomplete. You need to put your 10 digit phone number as the extension in the register command in sip.conf and add entries to extension.conf for your 10-digit extension under [from-broadvoice]. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy signal for incoming calls from broadvoice
Neil wrote: The problem appears to be with your settings. I have an identical configuration with my * box running behind a NAT firewall with the same firewall ports open. I have experienced the same problem before. If port forwarding is switched on then do NOT use the nat=yes and externalip/localnet settings - this breaks it. If using regular Asterisk, suggest you copy the exact settings from here: http://www.broadvoice.com/support_install_asterisk.html Thanks. I removed the nat=yes and externalip/localnet settings and followed the BroadVoice instructions but it still does not work.One deviation from the BroadVoice settings is that I don't supply an extension in the register entry. I tried supplying one and it still doesn't work. == my setting: register = phonenumber@sip.broadvoice.com:password:phonenumber@sip.broadvoice.com broadvoice setting: register = phonenumber@sip.broadvoice.com:password:phonenumber@sip.broadvoice.com/extension # Replace phonenumber with your account phone number, # Replace password with your password # Replace extension with one of your accessible extensions in the dial plan. === What should extension be? I want * to answer all incoming calls and not a specific extension. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why no sound from festival?
I can't get festival to output any sound. Any ideas? I have festival 1.95 installed on Fedora Core 4. % rpm -qa | grep fest festival-1.95-3 % cat festival.conf [general] host=localhost port=1314 usecache=yes cachedir=/var/lib/asterisk/festivalcache/ festivalcommand=(tts_textasterisk %s 'file)(quit)\n I added this to the festival.scm: ;;; Command for Asterisk begin (define (tts_textasterisk string mode) (tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe functions. (utt.send.wave.client (utt.wave.resample (utt.wave.rescale (utt.synth (eval (list 'Utterance 'Text string))) 5) 8000))) ;;; Command for Asterisk end I start the festival server BEFORE Asterisk on the same server: % festival_server -p 1314 -l /tmp I added this test extension: ; testing festival (text-to-speech app) exten = 1222,1,Answer() exten = 1222,n,Wait(1) exten = 1222,n,Festival(mary had a little lamb) ; do NOT use quotes around the string! if you use commas, you will have to escape them with a \ (backslash). exten = 1222,n,Hangup() When I dial 1222, all I hear is a click and then hangup. The festival_server.log file reads: Load server start /tmp/festival_server.scm stival port=1314 wrapper Mon Dec 26 00:48:29 EST 2005 : USING DEFAULT CONFIGURATION wrapper Mon Dec 26 00:48:30 EST 2005 : waiting serverMon Dec 26 00:48:30 2005 : Festival server started on port 1314 client(1) Mon Dec 26 00:48:43 2005 : accepted from myhost client(1) Mon Dec 26 00:48:43 2005 : disconnected ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with date time on Aastra 480i since release 1.3
Jacques Leisy wrote: Since the release 1.3 the 480i displays the wrong date and time. Something in 1947 ! I have followed the settings in the aastra.cfg. time server disabled: 1 time server1: 192.168.0.10 time server2: 192.168.0.11 # time server3: 128.121.51.132 time format: 1 date format: 0 My servers are running the proper time server. Same problem when I connect to the roku time server. Am I missing one entry? To enable the time server, you need: time server disabled: 0 1 means disabled 0 means enabled ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy signal for incoming calls from broadvoice
When someone calls me via BroadVoice, they get a busy signal. My * box is behind a NAT firewall. I have enabled port forwarding of UDP 5060 and 1:2 to the * box. I added nat=yes externalip and localnet to the sip.conf under [general]. It still doesn't work. I just want * to be able to answer the phone and send the caller to voicemail directly. What could be the problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy signal for incoming calls from broadvoice
trixter aka Bret McDanel wrote: On Sat, 2005-12-24 at 20:17 -0500, Robert La Ferla wrote: When someone calls me via BroadVoice, they get a busy signal. My * box is behind a NAT firewall. I have enabled port forwarding of UDP 5060 and 1:2 to the * box. I added nat=yes externalip and localnet to the sip.conf under [general]. It still doesn't work. I just want * to be able to answer the phone and send the caller to voicemail directly. What could be the problem? Did this start today or so? Rumor has it that BV is broken right now and others are having problems completing calls. I haven't been with BroadVoice long enough for my data to be relevent. i.e. less than 24 hrs. I tried calling their tech support and wasn't able to reach a live person and my call got dropped a few times. Haven't had problems otherwise. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra firmware 1.3.x (Far-End sound level issue)
Taco Scargo wrote: Hello, Just bought two 480i's which I updated to firmware 1.3 I experience the 'Far-End sound level issue' now. I tried configuring the handset tx gain: value but can only make it sound softer, not louder. If there is someone that has managed to get decent Far-end sound level, could he or she please e-mail their used values ? I have a similar issue with the Aastra 9133i and recorded .wav voicemail files. The recorded wav is too soft. I need to find a way to boost the volume level. Does anyone have any solutions or ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Please recommend VoIP service providers for US to Japan calls
Looking for a low-cost but good quality (i.e. value) service provider for US to Japan calls. Either a low monthly unlimited rate under $30 or very low per minute rates. I'm using SIP and analog phones w/Asterisk. The called party in Japan probably has PSTN phones with few exceptions. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aastra 9133i directory list downloading
How do you configure aastra.cfg to download directory list entries to each phone? The Aastra documentation is very sketchy. Anyone have an example??? You can use the Aastra Web UI (Operation-Directory) or the configuration files (aastra.cfg and mac.cfg) to download the Directory List. You use the following new configuration parameters in the configuration files: • directory 1 • directory 2 The software that reads directory files from the server, loads the file’s contents into the phone's NVRAM when the phone is booting. Directory entries in the NVRAM that originate from a server directory file are 'owned' by the server. During the boot process both directory files are read, combined into a single list, and any duplicate entries are deleted from the list. Any entries in this list that are not already in the phone's NVRAM are added to the NVRAM and flagged as being owned by the server. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring and zapata.conf
Thanks. Can anyone explain what the three values for the ring pattern signify? I assume it's a ring cadence pattern (in ms) but shouldn't it be 4 values (ring on, ring off, ring on, ring off) So is Asterisk ignoring the last ring off? And does Asterisk have some tolerance value for the ring timing? ie. 323 ms +/- 10ms or is it exact? e.g. dring1=323,0,0 Ring Patterns (Non-Asterisk) http://resource.intel.com/telecom/support/tnotes/gentnote/dl_soft/tn235.htm ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring and zapata.conf
Does anyone have distinctive ring working with Asterisk? Could you share your zapata.conf and relevent extensions.conf? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] low audio volume on recorded .wav voicemail messages
The audio volume of voicemail messages (msgNNN.wav) is rather low. Is there a parameter/option to adjust gain? In my voicemail.conf, I use these formats: format=wav49|gsm|wav Maybe I should use a different format? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring and zapata.conf
I am trying to configure zapata.conf to handle distinctive ring. Everytime someone calls my main number, I get a ring pattern of 0,0,0 which works consistently. The problem is that every time someone calls one of the other phone numbers (same number each time), I get a different ring pattern in the console. I also could not find documentation on what the three numbers in the ring pattern mean. My guess is that the first number is time spent ringing (in milliseconds), the second number is the time between rings and the third is how long the second ring rings for. Is this correct? The fact that I'm getting different numbers tells me that perhaps there's a range of values involved in the detection. Someone please elucidate! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] indications.conf for Japan?
Anyone have an indications.conf entry for Japan? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra 480i
Carlos Chavez wrote: On Fri, 2005-12-16 at 12:48 -0800, Dave wrote: I had a lot of issues with 480i too and this is how I resolved it: 1) Make sure that the file on the tftp server is called firmware followed by the type they suggest (I do not remember the type name) 2) Once this is done, your phones should download the firmware and reboot properly Actually this is no longer true since firmware 1.2. Now the firmware has to be named like the model of the phone and the extension .st for it to download automatically. So for the 480i you need a file called 480i.st and for the 9133i you need 9133i.st. I think this is no longer true for firmware 1.3 or greater. At least with the 9133i (1.2), I still had problems downloading it until I created a firmware.st file for the 1.3 update. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Rich Adamson wrote: The traditional pbx vendors (back then) would always use the same words that Kevin used, emphasizing the differences between key systems and pbx's. However, many of the pbx manufacturers finally realized they were loosing revenue due to those limitations, and began implementing key-system-type functions in their pbx's. They were not trying to address the key system market, but rather make their pbx products more valuable from a user's perspective. Those that are influencing or controlling the direction of asterisk haven't learned that lesson as yet, partially because of the lack of functionality in the sip phones themselves and partially because asterisk is being developed through the open source community (limited development resources and no published long term plan). Those individuals that have worked towards developing the sip rfc standards have recognized some of the key system vs pbx needs, and have added to the sip standards. However, it takes a while for the sip phone manufacturers (and voip pbx manufacturers) to implement those standards, and in some cases, the manufacturers purposefully leave out certain functions in their sip products to protect their investments in proprietary products. It certainly is not difficult to visualize how voip switching products (such as asterisk or any of the commercial products) could be oriented towards being a switch and address the needs of key systems, pbx's, and central office switching in the same basic product. All of the same functions are required in each case. Asterisk will get there, it will just take a little longer since there isn't any published long term plan to influence the short term development. (No offense intended to any asterisk individual or group; just the nature of most open source development.) I can also assure you that several large companies (most of those company names likely wouldn't be recognized by many of the readers here) are watching the asterisk development closely, and likely are in fear of various open source products negatively impacting their core business. They will adjust their product development (and plans) in an effort to remain one (or more) steps ahead from a marketing sales perspective. Thanks for the history on PBX and key systems. History has a way of repeating itself. I think Asterisk will have to implement features of a key system in the near future. Just judging from the reaction from friends and family who are fascinated by my Asterisk installation, there is huge demand for this kind of system. Digium is just losing out on sales. Is there an open source key system? What other alternative systems are there? How about OpenPBX? Are they integrating any key system support? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Is there a way for another extension to join a call in progress? i.e. If I can't share the line with all extensions, it would be nice to have a single button (dial sequence) that allows any extension to join the call. How can this be configured? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sharing a line w/multiple extensions
I'd like to configure Asterisk so that incoming calls from one POTS line are shared amongst multiple extensions. i.e. If one SIP phone answers the call, another SIP extension phone can pick up and join the conversation. How do I configure this in extensions.conf? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Let me revise this a little: I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Sean Cook wrote: also you can ring multiple extensions: Dial(SIP/101SIP/102SIP/103) I have that but once one extension picks up, others can't join in. Well, at least when I tried it with mixed SIP and Zap, it didn't work. Maybe all SIP does but I need it to work for all phones SIP and analog (via Zap). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Kevin P. Fleming wrote: Robert La Ferla wrote: I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? Asterisk is not a key system. It does not behave this way. What do you mean by 'another SIP phone can pick up (...) the conversation'? Exactly what would the SIP phone user do to accomplish that? Think residential installation where someone picks up the phone in one room but someone in another room wants to join the conversation. Ideally, I'd like to have Line 1 on every phone (SIP or analog) behave this way. Another poster pointed out a good potential approach using meetme. When an incoming call comes in, it dials all SIP + analog phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Kevin P. Fleming wrote: Robert La Ferla wrote: phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? Probably, but it would take some very creative dialplan programming and an external application to transfer the parties into a meetme room. You will not get 'pickup' behavior on the SIP phones regardless, they will have to press a speed dial button which would attempt to join the meetme. In other words: you can get there, but it will _not_ behave like a key system, and people will expect it to, so they will be frustrated when it doesn't. We've been down this road many times before, and many Asterisk installations have been taken out because the installers thought they could achieve key system behavior (or retrain the users) but failed. If you want to try, feel free... I'm only telling you what has happened before :-) Thanks. That's very helpful because being new to Asterisk, I don't know the history of what people have attempted to use Asterisk for. It's unfortunate that there's no way to do it because it sounds like others are looking for this same functionality. I wonder what it would take to implement this in Asterisk natively. Does Digium take feature requests? Certainly, this would have appeal for residential systems. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing analog extensions from SIP?
Doug Lytle wrote: I agree with Eric on this one. On my Polycom IP501s, I had to change the digit map to allow for # and * matching. For testing, remove the # and try again. Remove it from the phone's dial plan or all together? Also, my phone has a local dial plan that is set to this: X+#|XX+* I can't find any documentation on it and it doesn't seem to match up with the patterns in Asterisk. i.e. ; X - any digit from 0-9 ; Z - any digit from 1-9 ; N - any digit from 2-9 ; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9) ; . - wildcard, matches anything remaining (e.g. _9011. matches ; anything starting with 9011 excluding 9011 itself) ; ! - wildcard, causes the matching process to complete as soon as ; it can unambiguously determine that no other matches are possible So what do + and | and * do? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing analog extensions from SIP?
Is it possible to group all analog (regular phone) extensions so that you can dial it from a SIP extension? i.e. for use as an intercom I tried this: [default] exten = #3001,1,Dial(Zap/1,25,t,r) exten = #3001,2,Hangup but I just get a Call Failed and busy signal. I would think this is possible but I'm not sure how to configure it. I do have analog-SIP working just not SIP-analog. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing analog extensions from SIP?
Doug Lytle wrote: Is it possible to group all analog (regular phone) extensions so that you can dial it from a SIP extension? i.e. for use as an intercom I tried this: [default] exten = #3001,1,Dial(Zap/1,25,t,r) exten = #3001,2,Hangup Change your dial to: exten = #3001,1,Dial(ZAP/25,tr) Doug This didn't work. I still get Call Failed followed by a fast busy tone. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing analog extensions from SIP?
Eric ManxPower Wieling wrote: The phone's built in dialplan is prolly blocking the call. Check the docs for your SIP device. Remember SIP devices collect all digits, then pass them on to Asterisk as one packet. Also what Zap port is your analog phone connected to? What card are you using? Thanks. I'm using the Digium TDM11B card and an Aastra 9133i SIP phone. The phone has a local dial plan set to this: X+#|XX+* I'm not sure what this regex means??? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wait for X rings before answering?
How do I set up extensions.conf to wait for x rings (ringing all extensions) before answering? I'm trying to mimic a regular answering machine on an multiple analog phone system. Currently, Asterisk picks up after 1 ring and just tries to dial one extension. I want all extensions to ring. [incoming] exten = s,1,Dial(SIP/myext,25,t,r) exten = s,2,Voicemail(myext) exten = s,3,Hangup() Also, I couldn't find documentation on the r option for Dial(). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wait for X rings before answering?
Derek Whitten wrote: [incoming] exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r) exten = s,2,Voicemail(myext) exten = s,3,Hangup() Thanks. This will call/ring multiple extensions but what about waiting for X rings before going to voicemail? How do I do that? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wait for X rings before answering?
I realize that it's a timeout but what's implicit in that is that Asterisk can't detect # of rings just the amount of time spent ringing? I have been looking at the reference manual on asteriskguru.com. They say it's a timeout but they don't indicate the units. Is it milliseconds, microseconds or seconds? Dave Cotton wrote: On Fri, 2005-12-09 at 11:41 -0500, Robert La Ferla wrote: Derek Whitten wrote: [incoming] exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r) exten = s,2,Voicemail(myext) exten = s,3,Hangup() Thanks. This will call/ring multiple extensions but what about waiting for X rings before going to voicemail? How do I do that? What do you think the 25 does? Maybe it's a time or something. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Synthesized Voice for Asterisk
Dakota wrote: Are there any cool free software I can use to create automated voice message greetings for my PBX? Take a look at Festival/Festvox. I'm not sure about the output format but you could use sox to convert to gsm. http://festvox.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Synthesized Voice for Asterisk
snacktime wrote: The festival tts engine is free, but the the quality leaves a lot to be desired. Definitly not something you would use in a business. I'd say it depends on the use. Try it for yourself and see. Be sure to try different voices because some sound better than others. Online demo http://www.cs.cmu.edu/~awb/festival_demos/userin.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra 9133i Configurations - are the file namesto be lower case or upper case or does it matter?
[EMAIL PROTECTED] wrote: Thanks, I did that with upper and lower case, using 1.3. I have another issue then because it is still not loading, it appears the phone is loading but when I check the configs aren't there. I looked at this last night. You need to have an aastra.cfg file in your directory in addition to the mac.cfg. If the aastra.cfg file is not present, your phone will not download anything. This is a bug that was supposedly fixed in 1.3 but I found that it still is broken. So, add the aastra.cfg. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I set up extensions.conf to dial out on analog telephone line?
I have one SIP phone (and soon a 2nd phone) and a Digium TDM11B (1 FXO + 1 FXS) card. I would like to be able to dial out the analog line via Asterisk. How do I configure that? i.e I'd like any extension to be able to dial 411, 911, 0, (617) 555-1212, 16175551212, etc... and have these routed out the POTS line. Just like a regular telephone. % cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 FXSKS (In use) 3 WCTDM/0/2 4 WCTDM/0/3 % cat /etc/zaptel.conf fxoks=1 fxsks=2 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra 9133i Configurations - are the file names to be lower case or upper case or does it matter?
Lists wrote: According to the wiki page http://www.voip-info.org/tiki-index.php?page=Aastra+480i+Configuration it shows lowercase file name and then there is a comment at the bottom that it needs to be capitalized. I have tried it both ways with no luck. Could someone comment on which way the cfg files need to be in the /tftpboot directory? Thanks in advance. You need to look up the MAC address of your 9133i. It's on the bar code on the bottom of the phone. If your MAC address looked like this: 00 01 2d 09 58 c1 You would create a file (in uppercase except the .cfg which is in lowercase) called: 00012D0958C1.cfg in the /tftpboot directory. This will configure that SPECIFIC PHONE because it's tied to the MAC address. The case sensitivity only applies to pre-1.3 firmware versions. 1.3 can handle upper and lowercase. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i
Let me simplify my problem. I have a single Aastra 9133i SIP phone and latest Asterisk from SVN source running on Fedora Core 4. The phone currently says No Service I would like to be able to dial 1234 from the phone and get Asterisk to play back an audio message or say some digits. I can't get this to work with either SayDigits or Playback. Please help. == sip.conf == [general] port = 5060 bindaddr = 0.0.0.0 context=tutorial [3006] type=friend username=3006 secret=mypassword host=dynamic canreinvite=no permit=192.168.0.0/24 allow=all mailbox=3006 === extensions.conf === [tutorial] exten = 1234,1,Answer exten = 1234,2,SayDigits(123456789) ** TFTP directory ** = mymacaddress.cfg = sip line1 auth name: 3006 sip line1 password: mypassword sip line1 user name: 3006 sip line1 display name: myname sip line1 screen name: myname === aastra.cfg === dhcp: 1# DHCP enabled. sip silence suppression: 2 # 0 = off, 1 = on, 2 = default sip proxy port: 5060 # 5060 is set by default. sip registrar ip: 192.168.0.99# IP of registrar. --- THIS IS THE IP of my Asterisk and tftp server sip registrar port: 5060 # 5060 is set by default. sip digit time out: 6 time server disabled: 0 # Time server disabled. time server1: 192.168.0.99# Enable time server and enter at ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i
One more thing. I upgraded the firmware of the 9133i to 1.3. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i
Pete Barnwell wrote: I wasted a lot of time getting 9112is to work with almost identical setup. The problem I eventually found was that the 9112is look for the config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas the documentation says they look for lower case, so they were ignoring my tftp settings. The 9133i may well be the same. The other thing I had to do was to provide the line next-server tftpserver ip; in dhcpd.conf to get them to pick everything up. (IIRC that last bit was only to do with timedate format though). I read about the mac address case sensitivity so I used an all uppercase filename which works fine. The downloading of the firmware works fine too. I also have the ntp time/date working. I just can't get Asterisk to respond to the phone! Help! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i
Dave Cotton wrote: One thing is to do a factory reset to reinit everything, I did that with my 9112i after upgrading the firmware. I just did that. Now Asterisk is giving me the follow error: (0.99 is my Asterisk server and 0.111 is the phone) Dec 5 12:04:10 NOTICE[14222]: chan_sip.c:10817 handle_request_register: Registration from 'No User sip:[EMAIL PROTECTED]:5060' failed for '192.168.0.111' - Username/auth name mismatch -- Registered SIP '3006' at 192.168.0.111 port 5060 expires 300 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
