Re: [asterisk-users] Call recording problems from queue
Ex Vito [EMAIL PROTECTED] writes: I don't have access to an asterisk system right now (nor any other sort of information source) but I seem to recall that from 1.4 onwards the config option for recording queue calls is named differently... Is it mixmonitor ? Check you 1.4 queues.conf sample. PS: I'm not really sure about this one! Hi exvito, Mysteriously it started working today. Maybe Asterisk just needed a restart after playing with the configuration all day, I'll see if it keeps working. Thanks! ---Scott. ___ -- 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
Re: [asterisk-users] LDAP
Please check: http://bugs.digium.com/view.php?id=12112 which we had to fix ourselves. There are still problems using: 1. patterns in extensions 2. queue members 3. sip.conf, iax.conf, voicemail,etc should all work fine. Note the schema include with the distribution is invalid for the supplied res_ldap.conf. You will have to fix the schema yourself or modify res_ldap.conf to match your schema. the multi_ldap function in res_config_ldap.c is flawed. Any call to this function will result in no matches being returned. Quoting Gonzalo Servat [EMAIL PROTECTED]: Hi again :) I've downloaded, compiled installed 1.6.0-beta4 --with-ldap. After a few hours of messing with it, I've managed to get it to say that it has connected successfully to the LDAP backend (by looking at the output of realtime ldap status). I've modified extconfig.conf to what it should be (after reading many different configs on the subject). The trouble I'm having now is actually authenticating with a SIP user. I am running slapd in debug mode (slapd -d 4095) and I would have expected to see lots of activity on the console when I attempt to authenticate as a SIP user, but I see none at all. Is this normal? Thanks! Regards, Gonzalo On Thu, Mar 6, 2008 at 12:37 AM, Gonzalo Servat [EMAIL PROTECTED] wrote: Hi All, I've just compiled Asterisk 1.4.18 and I'm planning on using an LDAP tree where the users will each have their account, SIP username/password, extension number, context, etc. My first question is: can this be done with 1.4.x? If so, where can I get the res_config_ldap from?? I googled quite a bit and found a res_config_ldap that looks to be coded for 1.2. Is anyone running Asterisk with LDAP? Is it stable? Thanks in advance. Regards, Gonzalo -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz ___ -- 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] [Invitation] VoIP Users Conference @ Fri Mar 7 12:00 - 13:00 ()
BEGIN:VCALENDAR
PRODID:-//Google Inc//Google Calendar 70.9054//EN
VERSION:2.0
CALSCALE:GREGORIAN
METHOD:REQUEST
BEGIN:VEVENT
DTSTART:20080307T11Z
DTEND:20080307T12Z
DTSTAMP:20080306T082900Z
ORGANIZER;CN=Randy R:MAILTO:[EMAIL PROTECTED]
UID:[EMAIL PROTECTED]
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=NEEDS-ACTION;RSVP=
TRUE;X-NUM-GUESTS=0:MAILTO:asterisk-users@lists.digium.com
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=NEEDS-ACTION;RSVP=
TRUE;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=NEEDS-ACTION;RSVP=
TRUE;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=ACCEPTED;RSVP=TRUE
;CN=Randy R;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
CLASS:PRIVATE
CREATED:20080306T082859Z
DESCRIPTION:Every week we try to get guests with ideas\, products and servi
ces you haven't had time to check out to come and talk about what they're d
oing. \n\nTomorrow\, Pika Technologies will be with us.\n\nFriday\, March 7
that 12:00 PM (Eastern US) 9AM PST\, 5PM GMT\n\n*** Call (724) 444-7444 o
r SIP:[EMAIL PROTECTED] ***\n\nAfter the call connects\, enter the conf
: 22622# and your_PIN# (or 1# if you have no PIN)\n\nIf ( (${You_are_Regist
ered}) (${PIN} == callerID) ) you will not need to enter an ID\;\n\nhtt
p://www.VoIPUsersConference.org for how to listen and join.\n\nAccording to
their site at http://www.pikatechnologies.com\, Pika offers reliable medi
a processing building blocks connect computer systems to TDM and IP network
s. Brand name companies design groundbreaking IVR\, call center\, custom PC
/IP PBX\, fax and logging solutions using PIKA Technologies' components.\n
\nhttp://food4wine.ning.com is the VUC Community Site (archive recordings\,
forum)\n\nIRC Freenode.Net #voip-users-conference is the channel to ask qu
estions if you can't call\n\nJoin us\, we look forward to hearing you. (Ech
o? I don hear no stinking)\n\n/r\nView your event at http://www.google.
com/calendar/event?action=VIEWeid=ZHM2bGkzM2VxaTZkZmdzaTJtZDNuMWk2dGcgYXN0
ZXJpc2stdXNlcnNAbGlzdHMuZGlnaXVtLmNvbQtok=MjMjc3BhbXN1Y2tzMjAwNUBnbWFpbC5j
b21kZmQ4ZWYzYjgyNDYzY2Y2ZmZmOTI3OWU2Y2RkNWZiMGViYjhiMWM4ctz=Europe%2FParis
hl=en.
LAST-MODIFIED:20080306T082859Z
LOCATION:http://voipusersconference.org
SEQUENCE:0
STATUS:CONFIRMED
SUMMARY:VoIP Users Conference
TRANSP:OPAQUE
END:VEVENT
END:VCALENDAR
invite.ics
Description: application/ics
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[asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST
BEGIN:VCALENDAR
PRODID:-//Google Inc//Google Calendar 70.9054//EN
VERSION:2.0
CALSCALE:GREGORIAN
METHOD:REQUEST
BEGIN:VEVENT
DTSTART:20080307T11Z
DTEND:20080307T12Z
DTSTAMP:20080306T082900Z
ORGANIZER;CN=Randy R:MAILTO:[EMAIL PROTECTED]
UID:[EMAIL PROTECTED]
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=NEEDS-ACTION;RSVP=
TRUE;X-NUM-GUESTS=0:MAILTO:asterisk-users@lists.digium.com
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=NEEDS-ACTION;RSVP=
TRUE;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=NEEDS-ACTION;RSVP=
TRUE;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=ACCEPTED;RSVP=TRUE
;CN=Randy R;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
CLASS:PRIVATE
CREATED:20080306T082859Z
DESCRIPTION:Every week we try to get guests with ideas\, products and servi
ces you haven't had time to check out to come and talk about what they're d
oing. \n\nTomorrow\, Pika Technologies will be with us.\n\nFriday\, March 7
that 12:00 PM (Eastern US) 9AM PST\, 5PM GMT\n\n*** Call (724) 444-7444 o
r SIP:[EMAIL PROTECTED] ***\n\nAfter the call connects\, enter the conf
: 22622# and your_PIN# (or 1# if you have no PIN)\n\nIf ( (${You_are_Regist
ered}) (${PIN} == callerID) ) you will not need to enter an ID\;\n\nhtt
p://www.VoIPUsersConference.org for how to listen and join.\n\nAccording to
their site at http://www.pikatechnologies.com\, Pika offers reliable medi
a processing building blocks connect computer systems to TDM and IP network
s. Brand name companies design groundbreaking IVR\, call center\, custom PC
/IP PBX\, fax and logging solutions using PIKA Technologies' components.\n
\nhttp://food4wine.ning.com is the VUC Community Site (archive recordings\,
forum)\n\nIRC Freenode.Net #voip-users-conference is the channel to ask qu
estions if you can't call\n\nJoin us\, we look forward to hearing you. (Ech
o? I don hear no stinking)\n\n/r\nView your event at http://www.google.
com/calendar/event?action=VIEWueid=ds6li33eqi6dfgsi2md3n1i6tg.
LAST-MODIFIED:20080306T082859Z
LOCATION:http://voipusersconference.org
SEQUENCE:0
STATUS:CONFIRMED
SUMMARY:VoIP Users Conference
TRANSP:OPAQUE
END:VEVENT
END:VCALENDAR
invite20080307T12.ics
Description: application/ics
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[asterisk-users] Fax in AGX Extra Addons for Asterisk causes Asterisk to die
I am using the rxfax and txfax application with Asterisk 1.4.18. When ever I try sending or receiving a fax, my Asterisk dies. I tried to enable debug to see what happens, but I have no clue why it happens. Please help me out. -- Regards, Nasir. ___ -- 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] Accepted: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()
BEGIN:VCALENDAR METHOD:REPLY PRODID:Microsoft CDO for Microsoft Exchange VERSION:2.0 BEGIN:VTIMEZONE TZID:(GMT) Greenwich Mean Time/Dublin/Edinburgh/London X-MICROSOFT-CDO-TZID:1 BEGIN:STANDARD DTSTART:16010101T02 TZOFFSETFROM:+0100 TZOFFSETTO:+ RRULE:FREQ=YEARLY;WKST=MO;INTERVAL=1;BYMONTH=10;BYDAY=-1SU END:STANDARD BEGIN:DAYLIGHT DTSTART:16010101T01 TZOFFSETFROM:+ TZOFFSETTO:+0100 RRULE:FREQ=YEARLY;WKST=MO;INTERVAL=1;BYMONTH=3;BYDAY=-1SU END:DAYLIGHT END:VTIMEZONE BEGIN:VEVENT DTSTAMP:20080306T082900Z DTSTART;TZID=(GMT) Greenwich Mean Time/Dublin/Edinburgh/London:20080307T1 1 SUMMARY:Accepted: [asterisk-users] [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 () UID:[EMAIL PROTECTED] ATTENDEE;ROLE=REQ-PARTICIPANT;PARTSTAT=ACCEPTED;RSVP=TRUE;CN=Morgan Gilroy :MAILTO:[EMAIL PROTECTED] ORGANIZER:MAILTO:asterisk-users@lists.digium.com LOCATION:http://voipusersconference.org DTEND;TZID=(GMT) Greenwich Mean Time/Dublin/Edinburgh/London:20080307T120 000 SEQUENCE:0 PRIORITY:5 CLASS:Private CREATED:20080306T084202Z LAST-MODIFIED:20080306T084202Z STATUS:TENTATIVE TRANSP:OPAQUE X-MICROSOFT-CDO-BUSYSTATUS:BUSY X-MICROSOFT-CDO-INSTTYPE:0 X-MICROSOFT-CDO-REPLYTIME:20080306T083633Z X-MICROSOFT-CDO-INTENDEDSTATUS:BUSY X-MICROSOFT-CDO-ALLDAYEVENT:FALSE X-MICROSOFT-CDO-IMPORTANCE:1 X-MICROSOFT-CDO-OWNERAPPTID:-1 X-MICROSOFT-CDO-APPT-SEQUENCE:0 X-MICROSOFT-CDO-ATTENDEE-CRITICAL-CHANGE:20080306T083633Z X-MICROSOFT-CDO-OWNER-CRITICAL-CHANGE:20080306T082900Z END:VEVENT END:VCALENDAR ___ -- 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] Accepted: VoIP Users Conference
BEGIN:VCALENDAR METHOD:REPLY PRODID:Microsoft Exchange Server 2007 VERSION:2.0 BEGIN:VTIMEZONE TZID:Eastern Standard Time BEGIN:STANDARD DTSTART:16010101T02 TZOFFSETFROM:-0400 TZOFFSETTO:-0500 RRULE:FREQ=YEARLY;INTERVAL=1;BYDAY=1SU;BYMONTH=11 END:STANDARD BEGIN:DAYLIGHT DTSTART:16010101T02 TZOFFSETFROM:-0500 TZOFFSETTO:-0400 RRULE:FREQ=YEARLY;INTERVAL=1;BYDAY=2SU;BYMONTH=3 END:DAYLIGHT END:VTIMEZONE BEGIN:VEVENT ATTENDEE;PARTSTAT=ACCEPTED;CN=Steven C. Blair:MAILTO:[EMAIL PROTECTED] COMMENT: SUMMARY:Accepted: VoIP Users Conference DTSTART;TZID=Eastern Standard Time:20080307T06 DTEND;TZID=Eastern Standard Time:20080307T07 UID:[EMAIL PROTECTED] CLASS:PRIVATE PRIORITY:5 DTSTAMP:20080306T083641Z TRANSP:OPAQUE STATUS:CONFIRMED SEQUENCE:0 LOCATION:http://voipusersconference.org X-MICROSOFT-CDO-APPT-SEQUENCE:0 X-MICROSOFT-CDO-OWNERAPPTID:-1 X-MICROSOFT-CDO-BUSYSTATUS:BUSY X-MICROSOFT-CDO-INTENDEDSTATUS:BUSY X-MICROSOFT-CDO-ALLDAYEVENT:FALSE X-MICROSOFT-CDO-IMPORTANCE:1 X-MICROSOFT-CDO-INSTTYPE:0 END:VEVENT END:VCALENDAR ___ -- 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
Re: [asterisk-users] Asterisk based UNIX
On Thu, 2008-03-06 at 08:21 +0100, randulo wrote: On Thu, Mar 6, 2008 at 5:32 AM, Carole Migden [EMAIL PROTECTED] wrote: Generally what you know is best This is close to the best advice I've seen on this list in the last 5 years! The rest is a question of religion ;) Should have read: The Future Of Telephony (asterisk bible) It says the same already in the first edition of the book ___ -- 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] New Time Proposed: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()
BEGIN:VCALENDAR METHOD:REPLY PRODID:Microsoft CDO for Microsoft Exchange VERSION:2.0 BEGIN:VTIMEZONE TZID:(GMT+01.00) Sarajevo/Warsaw/Zagreb X-MICROSOFT-CDO-TZID:2 BEGIN:STANDARD DTSTART:16010101T03 TZOFFSETFROM:+0200 TZOFFSETTO:+0100 RRULE:FREQ=YEARLY;WKST=MO;INTERVAL=1;BYMONTH=10;BYDAY=-1SU END:STANDARD BEGIN:DAYLIGHT DTSTART:16010101T02 TZOFFSETFROM:+0100 TZOFFSETTO:+0200 RRULE:FREQ=YEARLY;WKST=MO;INTERVAL=1;BYMONTH=3;BYDAY=-1SU END:DAYLIGHT END:VTIMEZONE BEGIN:VEVENT DTSTAMP:20080306T082900Z DTSTART;TZID=(GMT+01.00) Sarajevo/Warsaw/Zagreb:20080307T12 SUMMARY:New Time Proposed: [asterisk-users] [Invitation] VoIP Users Confere nce @ Fri Mar 712:00 - 13:00 () UID:[EMAIL PROTECTED] ATTENDEE;ROLE=REQ-PARTICIPANT;PARTSTAT=TENTATIVE;RSVP=TRUE;CN=Giles Cooche y:MAILTO:[EMAIL PROTECTED] ORGANIZER:MAILTO:asterisk-users@lists.digium.com LOCATION:http://voipusersconference.org DTEND;TZID=(GMT+01.00) Sarajevo/Warsaw/Zagreb:20080307T13 SEQUENCE:0 PRIORITY:5 COMMENT:\N\N--\NNew Meeting Time Proposed:\N10 March 2010 1 2:00-13:00 (GMT+01:00) Brussels\, Copenhagen\, Madrid\, Paris.\N CLASS:Private CREATED:20080306T084436Z LAST-MODIFIED:20080306T084436Z STATUS:TENTATIVE TRANSP:OPAQUE X-MICROSOFT-CDO-BUSYSTATUS:BUSY X-MICROSOFT-CDO-INSTTYPE:0 X-MICROSOFT-CDO-REPLYTIME:20080306T084434Z X-MICROSOFT-CDO-INTENDEDSTATUS:BUSY X-MICROSOFT-CDO-ALLDAYEVENT:FALSE X-MICROSOFT-CDO-IMPORTANCE:1 X-MICROSOFT-CDO-OWNERAPPTID:-1 X-MICROSOFT-CDO-APPT-SEQUENCE:0 X-MICROSOFT-CDO-ATTENDEE-CRITICAL-CHANGE:20080306T084434Z X-MICROSOFT-CDO-OWNER-CRITICAL-CHANGE:20080306T082900Z END:VEVENT END:VCALENDAR ___ -- 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
Re: [asterisk-users] Asterisk based UNIX
On Thu, Mar 6, 2008 at 9:42 AM, Hans Witvliet [EMAIL PROTECTED] wrote: Should have read: The Future Of Telephony (asterisk bible) It says the same already in the first edition of the book Actually, it's fairly common wisdom outside of mailing lists and IRC :) ___ -- 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] Asterisk authentication by SIP Proxy
Hi, I have a setup where asterisk (1.4.18) is connected to a SIP proxy. Now the SIP proxy challenges REGISTER and INVITE request from Asterisk. Asterisk is able to handle REGISTER request challenge but for INVITE request, it seems to handle authentication for only one user that exists under the global authentication. Even if I configure all users in global authentication, asterisk is using only the first username:password pair for that realm (realm and domain is same for all users). Can anyone help me understand why this is happening and if there is solution? Regards, Mayur ___ -- 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
Re: [asterisk-users] codec_g729-v34 Builds Now Available
Hi, I have just checked again and the Solaris build of the codec appears to be v33 and not v34 as advertised. Thanks Bruce Bruce McAlister wrote: Hi, The Solaris build still appears to be at v32. Am I being a little hasty :) Thanks Bruce The Asterisk Development Team wrote: Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux builds for both 1.4 and 1.6 have been updated for various API changes. * All of the Linux builds include changes so that an Ethernet interface explicitly named eth0, or eth1, etc., is no longer required. All of the builds are available from the following URL: * http://downloads.digium.com/pub/telephony/codec_g729/ Thank you for your support! ___ -- 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 -- +---+ | Bruce McAlister Blueface Ltd | | [EMAIL PROTECTED] http://www.blueface.ie | +---+ ___ -- 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] Tentative: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()
BEGIN:VCALENDAR METHOD:REPLY PRODID:Microsoft CDO for Microsoft Exchange VERSION:2.0 BEGIN:VTIMEZONE TZID:(GMT-05.00) Eastern Time (US Canada) X-MICROSOFT-CDO-TZID:10 BEGIN:STANDARD DTSTART:16010101T02 TZOFFSETFROM:-0400 TZOFFSETTO:-0500 RRULE:FREQ=YEARLY;WKST=MO;INTERVAL=1;BYMONTH=11;BYDAY=1SU END:STANDARD BEGIN:DAYLIGHT DTSTART:16010101T02 TZOFFSETFROM:-0500 TZOFFSETTO:-0400 RRULE:FREQ=YEARLY;WKST=MO;INTERVAL=1;BYMONTH=3;BYDAY=2SU END:DAYLIGHT END:VTIMEZONE BEGIN:VEVENT DTSTAMP:20080306T082900Z DTSTART;TZID=(GMT-05.00) Eastern Time (US Canada):20080307T06 SUMMARY:Tentative: [asterisk-users] [Invitation] VoIP Users Conference @ Fr i Mar 712:00 - 13:00 () UID:[EMAIL PROTECTED] ATTENDEE;ROLE=REQ-PARTICIPANT;PARTSTAT=TENTATIVE;RSVP=TRUE;CN=Kenneth T. V an Wie II:MAILTO:[EMAIL PROTECTED] ORGANIZER:MAILTO:asterisk-users@lists.digium.com LOCATION:http://voipusersconference.org DTEND;TZID=(GMT-05.00) Eastern Time (US Canada):20080307T07 SEQUENCE:0 PRIORITY:5 CLASS:Private CREATED:20080306T123812Z LAST-MODIFIED:20080306T123813Z STATUS:TENTATIVE TRANSP:OPAQUE X-MICROSOFT-CDO-BUSYSTATUS:BUSY X-MICROSOFT-CDO-INSTTYPE:0 X-MICROSOFT-CDO-REPLYTIME:20080306T123844Z X-MICROSOFT-CDO-INTENDEDSTATUS:BUSY X-MICROSOFT-CDO-ALLDAYEVENT:FALSE X-MICROSOFT-CDO-IMPORTANCE:1 X-MICROSOFT-CDO-OWNERAPPTID:-1 X-MICROSOFT-CDO-APPT-SEQUENCE:0 X-MICROSOFT-CDO-ATTENDEE-CRITICAL-CHANGE:20080306T123844Z X-MICROSOFT-CDO-OWNER-CRITICAL-CHANGE:20080306T082900Z END:VEVENT END:VCALENDAR ___ -- 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] Declined: VoIP Users Conference
BEGIN:VCALENDAR PRODID:-//Microsoft Corporation//Outlook 11.0 MIMEDIR//EN VERSION:2.0 METHOD:REPLY BEGIN:VEVENT DTSTART:20080307T11Z DTEND:20080307T12Z LOCATION:http://voipusersconference.org TRANSP:OPAQUE SEQUENCE:0 UID:[EMAIL PROTECTED] DTSTAMP:20080306T094011Z SUMMARY:Declined: VoIP Users Conference PRIORITY:5 X-MICROSOFT-CDO-IMPORTANCE:1 CLASS:PRIVATE ATTENDEE;PARTSTAT=DECLINED:MAILTO:[EMAIL PROTECTED] END:VEVENT END:VCALENDAR No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.21.5/1314 - Release Date: 05/03/2008 18:38 ___ -- 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] Tentative: VoIP Users Conference
BEGIN:VCALENDAR PRODID:-//Microsoft Corporation//Outlook 11.0 MIMEDIR//EN VERSION:2.0 METHOD:REPLY BEGIN:VEVENT DTSTART:20080307T11Z DTEND:20080307T12Z LOCATION:http://voipusersconference.org TRANSP:OPAQUE SEQUENCE:0 UID:[EMAIL PROTECTED] DTSTAMP:20080306T094112Z DESCRIPTION:\n\nhi\,\n\nyou wrote :\n\nFri Mar 7 12:00 – 13:00 \n(Timezone: Paris) \nhttp://voipusersconference.org (map http://maps.google.fr/maps?q=http%3A%2F%2Fvoipusersconference.orghl=en ) \nCalendar: \n\nEvery week we try to get guests with ideas\, products and services you haven't had time to check out to come and talk about what they're doing. \n\nTomorrow\, Pika Technologies will be with us. \n\nFriday\, March 7that 12:00 PM (Eastern US) 9AM PST\, 5PM GMT \n\n\nnow\, Fri Mar 7 12:00 – 13:00 (Timezone: Paris) is ok for me\nbut Friday\, March 7that 12:00 PM (Eastern US) 9AM PST\, 5PM GMT is not\n\nwhat time is it then ? 12-13 Paris time or eastern us ?\n\nbest regards\n\nt. jacobson\n\n \n\n SUMMARY:Tentative: VoIP Users Conference PRIORITY:5 X-MICROSOFT-CDO-IMPORTANCE:1 CLASS:PRIVATE ATTENDEE;PARTSTAT=TENTATIVE:MAILTO:[EMAIL PROTECTED] END:VEVENT END:VCALENDAR No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.21.5/1314 - Release Date: 05/03/2008 18:38 ___ -- 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
Re: [asterisk-users] {s} - extension
Thank you all for answers. As I understand s - i and others is device specific. I will not need them in my SIP configuration. 2008/3/5, Andres Jimenez [EMAIL PROTECTED]: On Wed, Mar 5, 2008 at 1:36 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: This is not needed. If the extension is not found, there is a fallthrough to 's' (Right? Or is it chan_zap-specific)? I would say it's chan_zap-specific. From http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf For some kinds of connections — such incoming calls from an outside telephone line — the user has not dialed an extension. In that case, Asterisk behaves as if the user had dialed a special extension named s (for Start). Asterisk will look for an extension number s in the definition of the context for that channel for instructions about what it should do to handle the call. The key factor is that s is used when NO EXTENSION has been specified (when the call is not clearly directed to an specific number). As far as I know, that's the way analog lines behave. The line just receives the call, but no information says to which number the call was sent. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- 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 ___ -- 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
Re: [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST
On Thu, Mar 6, 2008 at 11:38 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: If you want to reply to this message regarding the schedule, please reply to the author. Your messages look very badly in the archives. And there is really no need to have 500 replies to this message on-list. Sincere apologies to all, I seriously screwed up by adding the calendar item. That obviously was a stupid idea, I did NOT expect the result, mea culpa. /r ___ -- 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] AEL - SQL and TIMEDIFF()
Hello list,
I'm having some problem integrating the SELECT TIMEDIFF() and SELECT
DATEDIFF() in my code. The syntax I'm using works without any problems if I
run them directly from the MySQL Client, but from the Asterisk Dialplan it
just wont work. Is there a limitation in the MySQL() application for the
Asterisk dialplan that produces this error?
CODE
context testsql {
s = {
MYSQL(Connect connid ${DBHOST} ${DBUSER} ${DBPASS} ${DB});
MYSQL(Query resultid ${connid} SELECT TIMEDIFF(callend,callstart) FROM
tblCall WHERE id=7);
MYSQL(fetch fetchid ${resultid} temp);
MYSQL(Disconnect ${connid});
}
}
/CODE
The error I'm getting is below:
[Mar 6 08:59:35] WARNING[27116]: app_addon_sql_mysql.c:268 aMYSQL_query:
aMYSQL_query: mysql_query failed. Error: You have an error in your SQL
syntax; check the manual that corresponds to your MySQL server version for
the right syntax to use near ') FROM tblCall WHERE id=7' at line 1
Has anyone done this kind of calculation before, or is there a better way to
do it?
Thanks,
Best regards,
Tobias
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Re: [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST
Hi On Thu, Mar 06, 2008 at 09:29:01AM +0100, randulo wrote: Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Tomorrow, Pika Technologies will be with us. Friday, March 7that 12:00 PM (Eastern US) 9AM PST, 5PM GMT If you want to reply to this message regarding the schedule, please reply to the author. Your messages look very badly in the archives. And there is really no need to have 500 replies to this message on-list. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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
Re: [asterisk-users] Newbie Polycom: IP600 Headset Problem
Lee, John (Sydney) wrote: Has anyone encountered such problems before? On the IP501 and IP301, yes. The handset cord dies. I've had to replace 3 so far. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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
Re: [asterisk-users] AEL - SQL and TIMEDIFF()
context testsql {
s = {
MYSQL(Connect connid ${DBHOST} ${DBUSER} ${DBPASS} ${DB});
MYSQL(Query resultid ${connid} SELECT
TIMEDIFF(callend,callstart) FROM tblCall WHERE id=7);
MYSQL(fetch fetchid ${resultid} temp);
MYSQL(Disconnect ${connid});
}
}
/CODE
The error I'm getting is below:
[Mar 6 08:59:35] WARNING[27116]: app_addon_sql_mysql.c:268
aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You
have an error in your SQL syntax; check the manual that
corresponds to your MySQL server version for the right syntax
to use near ') FROM tblCall WHERE id=7' at line 1
I think the solution would be to escape the , with a backslash, so
your query would look like this:
SELECT TIMEDIFF(callend\,callstart) FROM tblCall WHERE id=7
Maybe even the brackets ()
--
Andreas Sikkema
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[asterisk-users] Declined: VoIP Users Conference
BEGIN:VCALENDAR METHOD:REPLY PRODID:Microsoft Exchange Server 2007 VERSION:2.0 BEGIN:VTIMEZONE TZID:Eastern Standard Time BEGIN:STANDARD DTSTART:16010101T02 TZOFFSETFROM:-0400 TZOFFSETTO:-0500 RRULE:FREQ=YEARLY;INTERVAL=1;BYDAY=1SU;BYMONTH=11 END:STANDARD BEGIN:DAYLIGHT DTSTART:16010101T02 TZOFFSETFROM:-0500 TZOFFSETTO:-0400 RRULE:FREQ=YEARLY;INTERVAL=1;BYDAY=2SU;BYMONTH=3 END:DAYLIGHT END:VTIMEZONE BEGIN:VEVENT ATTENDEE;PARTSTAT=DECLINED;CN=Evan Ruff:MAILTO:[EMAIL PROTECTED] om COMMENT: SUMMARY:Declined: VoIP Users Conference DTSTART;TZID=Eastern Standard Time:20080307T06 DTEND;TZID=Eastern Standard Time:20080307T07 UID:[EMAIL PROTECTED] CLASS:PRIVATE PRIORITY:5 DTSTAMP:20080306T131621Z TRANSP:OPAQUE STATUS:CONFIRMED SEQUENCE:0 LOCATION:http://voipusersconference.org X-MICROSOFT-CDO-APPT-SEQUENCE:0 X-MICROSOFT-CDO-OWNERAPPTID:-1 X-MICROSOFT-CDO-BUSYSTATUS:BUSY X-MICROSOFT-CDO-INTENDEDSTATUS:BUSY X-MICROSOFT-CDO-ALLDAYEVENT:FALSE X-MICROSOFT-CDO-IMPORTANCE:1 X-MICROSOFT-CDO-INSTTYPE:0 END:VEVENT END:VCALENDAR ___ -- 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] chan_iax2.c:3904 iax2_trunk_queue: Maximum trunk data space exceeded
Hello, [Mar 6 07:07:51] WARNING[9994]: chan_iax2.c:3904 iax2_trunk_queue: Maximum trunk data space exceeded to ***.***.***.***:52213 I am seeing a ton of these errors on an IAX2 trunk to a second server with only 1 call on the trunk. I have found some information regarding the MTU size being an issue, however this is *only* with 1 G711 call on a trunk. One server is on a 100mbit connection in my datacenter, and the other is on a T1 at an office. Feedback? Google has not been of much help! Thanks! -- Brooks R. Bridges Telecommunications Manager Ifbyphone, Inc. Phone: (847) 983-3000 Fax: (847) 676-6553 [EMAIL PROTECTED] http://www.ifbyphone.com ___ -- 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] IAX user identification.
Is there a way to set up a user/peer in iax.conf where it matches incoming calls based entirely on IP? I have a provider that sets the username (as well as the extension) to the phone number that has been dialled, I'd prefer calls from that provider to all be identified as the same trunk. TIA for any help with this. ___ -- 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
Re: [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST
On Thu, Mar 06, 2008 at 08:43:43AM -0500, OCG Technical Support wrote: I (like many others probably have) added the sender of the invite to my spam filter. That avoids the many replies - and also blocks future email from someone stupid enough to spam multiple entire list with an invite! The sender of the invite has earned his credit in this mailing list. Feel free to ignore him and miss some useful sound ;-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] bristuff qozap support for beronet cards
Hi all, In the changelog of bristuff, as of version 0.4.0test4(test5) the beronet cards should be supported. Can anyone confirm if the beronet 2,4 and 8 ports version are supported by qozap now? Regards, stoffell ___ -- 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] Accepted: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()
BEGIN:VCALENDAR METHOD:REPLY PRODID:Microsoft CDO for Microsoft Exchange VERSION:2.0 BEGIN:VTIMEZONE TZID:(GMT-05.00) Eastern Time (US Canada) X-MICROSOFT-CDO-TZID:10 BEGIN:STANDARD DTSTART:16010101T02 TZOFFSETFROM:-0400 TZOFFSETTO:-0500 RRULE:FREQ=YEARLY;WKST=MO;INTERVAL=1;BYMONTH=11;BYDAY=1SU END:STANDARD BEGIN:DAYLIGHT DTSTART:16010101T02 TZOFFSETFROM:-0500 TZOFFSETTO:-0400 RRULE:FREQ=YEARLY;WKST=MO;INTERVAL=1;BYMONTH=3;BYDAY=2SU END:DAYLIGHT END:VTIMEZONE BEGIN:VEVENT DTSTAMP:20080306T082900Z DTSTART;TZID=(GMT-05.00) Eastern Time (US Canada):20080307T06 SUMMARY:Accepted: [asterisk-users] [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 () UID:[EMAIL PROTECTED] ATTENDEE;ROLE=REQ-PARTICIPANT;PARTSTAT=ACCEPTED;RSVP=TRUE;CN=Jeff Johnson :MAILTO:[EMAIL PROTECTED] ORGANIZER:MAILTO:asterisk-users@lists.digium.com LOCATION:http://voipusersconference.org DTEND;TZID=(GMT-05.00) Eastern Time (US Canada):20080307T07 SEQUENCE:0 PRIORITY:5 CLASS:Private CREATED:20080306T142753Z LAST-MODIFIED:20080306T142754Z STATUS:TENTATIVE TRANSP:OPAQUE X-MICROSOFT-CDO-BUSYSTATUS:BUSY X-MICROSOFT-CDO-INSTTYPE:0 X-MICROSOFT-CDO-REPLYTIME:20080306T142756Z X-MICROSOFT-CDO-INTENDEDSTATUS:BUSY X-MICROSOFT-CDO-ALLDAYEVENT:FALSE X-MICROSOFT-CDO-IMPORTANCE:1 X-MICROSOFT-CDO-OWNERAPPTID:-1 X-MICROSOFT-CDO-APPT-SEQUENCE:0 X-MICROSOFT-CDO-ATTENDEE-CRITICAL-CHANGE:20080306T142756Z X-MICROSOFT-CDO-OWNER-CRITICAL-CHANGE:20080306T082900Z END:VEVENT END:VCALENDAR ___ -- 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
Re: [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST
I (like many others probably have) added the sender of the invite to my spam filter. That avoids the many replies - and also blocks future email from someone stupid enough to spam multiple entire list with an invite! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: March-06-08 5:39 AM To: Asterisk Users List Subject: Re: [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST Hi On Thu, Mar 06, 2008 at 09:29:01AM +0100, randulo wrote: Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Tomorrow, Pika Technologies will be with us. Friday, March 7that 12:00 PM (Eastern US) 9AM PST, 5PM GMT If you want to reply to this message regarding the schedule, please reply to the author. Your messages look very badly in the archives. And there is really no need to have 500 replies to this message on-list. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 ___ -- 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
Re: [asterisk-users] Declined: VoIP Users Conference
Evan Ruff wrote: Since when is the users list a transport for calendar scheduling? -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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
Re: [asterisk-users] {s} - extension
Hi - Thank you all for answers. As I understand s - i and others is device specific. I will not need them in my SIP configuration. The s extension is not zap-specific. You can use it for any type of device. It's just the generic extension that a call will go to when no other matching extensions are present. As Tzafrir pointed out, you had no s extension in the default context, and your sip device was in the default context. Therefore, you were only able to dial extensions that you had explicitly declared. To access the s extension from your sip device, you'd either need to add your sip device to the context where your s extension is, or include that context in the default context. NOTE: Andres' example using _. will work, too (but you should make sure you put in at the end of a context if you want to put other extensions in that context as it will match all calls). - Noah 2008/3/5, Andres Jimenez [EMAIL PROTECTED]: On Wed, Mar 5, 2008 at 1:36 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: This is not needed. If the extension is not found, there is a fallthrough to 's' (Right? Or is it chan_zap-specific)? I would say it's chan_zap-specific. From http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf For some kinds of connections — such incoming calls from an outside telephone line — the user has not dialed an extension. In that case, Asterisk behaves as if the user had dialed a special extension named s (for Start). Asterisk will look for an extension number s in the definition of the context for that channel for instructions about what it should do to handle the call. The key factor is that s is used when NO EXTENSION has been specified (when the call is not clearly directed to an specific number). As far as I know, that's the way analog lines behave. The line just receives the call, but no information says to which number the call was sent. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- 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 ___ -- 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 ___ -- 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] FXS channel banks
Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel banks on reasonably sizeable installs I'd be interested to hear what device(s) you used, how simple or complex they were to configure, and whether there'd be any issues attaching multiple units to a single server. This install would be in the UK, so we do need to factor in the different conditions expected by UK POTS handsets (line impedance, etc.). Are most channel banks country-neutral, or do specific models need to be purchased for different line conditions in each country? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- 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
Re: [asterisk-users] Declined: VoIP Users Conference
Doug Lytle wrote: Evan Ruff wrote: Since when is the users list a transport for calendar scheduling? Since when are humans infallible? Randy made a mistake. He apologized for it. Let's move on... -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- 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
Re: [asterisk-users] Declined: VoIP Users Conference
Hey screwed up and has already apologized for it ok. Randy tried something and didn't realize all the replies were going to be 'resent' to the list. At least he's out there trying something different. And as he's tirelessly promoting asterisk every Friday afternoon putting his own time and energy out there on a regular basis you should cut him some slack. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Thursday, 6 March 2008 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Declined: VoIP Users Conference Evan Ruff wrote: Since when is the users list a transport for calendar scheduling? -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 ___ -- 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
Re: [asterisk-users] FXS channel banks
Chris Bagnall wrote: Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel banks on reasonably sizeable installs I'd be interested to hear what device(s) you used, how simple or complex they were to configure, and whether there'd be any issues attaching multiple units to a single server. This install would be in the UK, so we do need to factor in the different conditions expected by UK POTS handsets (line impedance, etc.). Are most channel banks country-neutral, or do specific models need to be purchased for different line conditions in each country? Thanks in advance. Regards, Chris www.citel.com I used them a few years back in a pilot install with legacy Nortel phones and it worked well. I gather they have grown tremendously from there. I'm in North America, don't know how well they support UK stuff. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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
Re: [asterisk-users] FXS channel banks
On Thu, Mar 06, 2008 at 03:21:47PM -, Chris Bagnall wrote: I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel banks on reasonably sizeable installs I'd be interested to hear what device(s) you used, how simple or complex they were to configure, and whether there'd be any issues attaching multiple units to a single server. This install would be in the UK, so we do need to factor in the different conditions expected by UK POTS handsets (line impedance, etc.). Are most channel banks country-neutral, or do specific models need to be purchased for different line conditions in each country? You might want to check the archives from, I think, early '07; I was looking into doing a hotel/motel system for a client, and asked almost exactly this question. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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
Re: [asterisk-users] Asterisk based UNIX
Actually, UNIX [tm] Describes meeting a standard, and not development history. http://en.wikipedia.org/wiki/Unix#Branding Absolutely! Which is why I referred to Linux as Unix-like and not UNIX. Linux is NOT licensed to use UNIX(r) per The Open Group's specs. BSD and Mac OS X are licensed to use UNIX(r). All this according to the wikipedia entry - I'm certainly no UNIX expert! Which, of course is a good thing. Otherwise, SCO would be hauling my butt in to court to testify!!! LOL ___ -- 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
Re: [asterisk-users] Asterisk based UNIX
On Thu, Mar 06, 2008 at 09:39:05AM -0600, Bill Andersen wrote: Actually, UNIX [tm] Describes meeting a standard, and not development history. http://en.wikipedia.org/wiki/Unix#Branding Absolutely! Which is why I referred to Linux as Unix-like and not UNIX. Linux is NOT licensed to use UNIX(r) per The Open Group's specs. BSD and Mac OS X are licensed to use UNIX(r). BSD has many flavours. I don't think any of them is actually certified by the OpenGroup. Yeah, and we really care. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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
Re: [asterisk-users] Declined: VoIP Users Conference
Dean Collins wrote: Hey screwed up and has already apologized for it ok. Please note the time the message was sent. 8:58AM Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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
Re: [asterisk-users] FXS channel banks
On Thu, Mar 06, 2008 at 10:38:36AM -0500, Drew Gibson wrote: www.citel.com I used them a few years back in a pilot install with legacy Nortel phones and it worked well. I gather they have grown tremendously from there. I'm in North America, don't know how well they support UK stuff. Citel are, are they not, the company that specializes in FXS channel banks specific to legacy digital phones? Do they do analog-POTS banks as well? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Declined: VoIP Users Conference
BEGIN:VCALENDAR PRODID:-//Microsoft Corporation//Outlook 11.0 MIMEDIR//EN VERSION:2.0 METHOD:REPLY BEGIN:VEVENT DTSTART:20080307T11Z DTEND:20080307T12Z LOCATION:http://voipusersconference.org TRANSP:OPAQUE SEQUENCE:0 UID:[EMAIL PROTECTED] DTSTAMP:20080306T160242Z SUMMARY:Declined: VoIP Users Conference PRIORITY:5 X-MICROSOFT-CDO-IMPORTANCE:1 CLASS:PRIVATE ATTENDEE;PARTSTAT=DECLINED:MAILTO:[EMAIL PROTECTED] END:VEVENT END:VCALENDAR ___ -- 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] Declined: VoIP Users Conference
BEGIN:VCALENDAR PRODID:-//Microsoft Corporation//Outlook 11.0 MIMEDIR//EN VERSION:2.0 METHOD:REPLY BEGIN:VEVENT ORGANIZER:MAILTO:asterisk-users@lists.digium.com DTSTART:20080307T11Z DTEND:20080307T12Z LOCATION:http://voipusersconference.org TRANSP:OPAQUE SEQUENCE:0 UID:[EMAIL PROTECTED] DTSTAMP:20080306T160255Z SUMMARY:Declined: VoIP Users Conference PRIORITY:5 X-MICROSOFT-CDO-IMPORTANCE:1 CLASS:PRIVATE ATTENDEE;PARTSTAT=DECLINED:MAILTO:[EMAIL PROTECTED] END:VEVENT END:VCALENDAR ___ -- 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] format of UNIQUEID variable
What is the format of the UNIQUEID variable? It seems to be something like: 40651204817492.56 Does it always have the pattern long_number.short_number? Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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
Re: [asterisk-users] Asterisk in the call center - how do you do it?
On Mar 5, 2008, at 5:46 PM, [EMAIL PROTECTED] wrote: If you are running a call centre (large or small) using Asterisk, I'd be interested to know how you log your agents in out: E.g. - Do you use AgentLogin (to force calls onto the agents, perhaps)? - Do you still use AgentCallbackLogin? - If you use AddQueueMember, are you - running it through the agent's phones (i.e. in the dialplan)? - through a manager interface some software (homebrew or otherwise)? - Do you allow agent hot-desking? - if so, how do you determine which agent is logged in at which desk at what time? - how do you deal with authentication, or don't you bother? It'd also be useful if you could tell me what version of Asterisk you're using. And, finally, a pure fishing expedition: - What kind of reporting (if any) do you currently get out of the Asterisk, and are you happy with it? We are a medium sided center, I'd guess, mostly inbound. We don't use the Queue app, since it seemed rather inadequate for us, so we rolled our own solution that does skills-based routing and various other enhanced features (all database driven.) Along with a custom client, we pass custom headers to handle client-server communication. Any agent can log into any workstation and things just work, and our app handles authentication of agents. (We also authenticate the workstations, but that's hard coded into the app.) As for reporting, again, a totally custom developed system that's an extension to what we were using with our old phone switch. On top of that, I've developed a number of web-based applications (using Apache Tapestry) to slice and dice our data for reporting (mostly graphically) that we use a lot. Since it's all quite specific to how we work and our custom solutions, it wouldn't help anyone, I'm sure. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- 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
Re: [asterisk-users] FXS channel banks
On Thu, Mar 6, 2008 at 10:49 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Mar 06, 2008 at 10:38:36AM -0500, Drew Gibson wrote: www.citel.com I used them a few years back in a pilot install with legacy Nortel phones and it worked well. I gather they have grown tremendously from there. I'm in North America, don't know how well they support UK stuff. Citel are, are they not, the company that specializes in FXS channel banks specific to legacy digital phones? Do they do analog-POTS banks as well? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) Citel is the worst product I have ever dealt with, worse than Grandstream but for different reasons. Anyways, for smaller port density I love the Quintum Tenor AX 24 port FXS, They may make a 48, I am not sure. This is a SIP connection, and there are probably a multitude of other products that do the same, Quintum blew me away with the sheer amount of options and configuration (that you will probably never use). I have heard people suggest MaxTNT for high port densities, which looks great, I just have no experience or need for such a device yet. The other option is a channel bank that connects via T1 or I guess E1 (although I have never seen an E1 30 port channel bank, I am in the US so it is not surprising) Thanks, Steve Totaro ___ -- 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] Best Java library to interact with Asterisk
Hi, I need to interact with my Asterisk and need a good Java class library. What do you think is the best? Thanks ___ -- 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
Re: [asterisk-users] format of UNIQUEID variable
On Thursday 06 March 2008 10:07:26 Vieri wrote: What is the format of the UNIQUEID variable? It seems to be something like: 40651204817492.56 Does it always have the pattern long_number.short_number? UniqueID is composed of the epoch when a call starts, plus a monotonically incrementing integer. Together, they will be unique for all calls originating from a single machine, as long as they are treated as a string and not as a float. Note that you can set asterisk to prefix the hostname to a uniqueid from asterisk.conf, which should make uniqueIDs globally unique (as long as you aren't repeating hostnames). -- Tilghman ___ -- 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] Call Manager as trunk
I have Asterisk 1.4 tied via SIP to a Cisco Callmanager 6.1 system. Calls between the systems (ie. extension to extension) work perfectly. However when I attempt to make an outside call from an Asterisk extension through Call Manager to the outside world, it connects but only for a few seconds, and on the Asterisk console I get: Got SIP response 503 Service Unavailable back from (ip of call manager) Coming the other way, if I call into the Call Manager system (from my cell to be exact), then transfer my call to the Asterisk SIP phone (an Aastra 57i), on the cell I can hear the voice on the Aastra, but the Aastra can only hear the Asterisk music on hold! As I mentioned though, going the other way and calling out from Asterisk to my cell works perfectly...for between 5 and 10 seconds (it varies), then disconnects with the above error. My sip.conf looks like this: [callman] type=friend context=incoming host=(ip of call manager) disallow=all allow=ulaw allow=alaw nat=yes canreinvite=yes qualify=yes I've tried experimenting with the externip and localnet parameters to no effect. Any ideas? ___ -- 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
Re: [asterisk-users] format of UNIQUEID variable
In article [EMAIL PROTECTED], Vieri [EMAIL PROTECTED] wrote: What is the format of the UNIQUEID variable? It seems to be something like: 40651204817492.56 Does it always have the pattern long_number.short_number? If the system has been running a long time with many calls, it could be long_number.long_number :-) The first (long) number is the Unix time_t timestamp (number of seconds since 00:00:00 GMT on 1 Jan 1970) of when the channel was created. The second (short) number is a sequence number, starting at 0 for the first created channel since Asterisk started up, and incrementing by 1 for each subsequent channel. In Asterisk 1.4 or later, an optional system name can be defined in asterisk.conf, and if defined, the unique ID becomes: system_name-timestamp.seq_num Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- 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] Declined: VoIP Users Conference
BEGIN:VCALENDAR PRODID:-//Microsoft Corporation//Outlook 12.0 MIMEDIR//EN VERSION:2.0 METHOD:REPLY X-MS-OLK-FORCEINSPECTOROPEN:TRUE BEGIN:VEVENT ATTENDEE;PARTSTAT=DECLINED:mailto:[EMAIL PROTECTED] CLASS:PRIVATE CREATED:20080306T172312Z DTEND:20080307T12Z DTSTAMP:20080306T172312Z DTSTART:20080307T11Z LAST-MODIFIED:20080306T172312Z LOCATION:http://voipusersconference.org PRIORITY:5 SEQUENCE:0 SUMMARY:Declined: VoIP Users Conference TRANSP:OPAQUE UID:[EMAIL PROTECTED] X-MICROSOFT-CDO-BUSYSTATUS:BUSY X-MICROSOFT-CDO-IMPORTANCE:1 X-MS-OLK-AUTOFILLLOCATION:FALSE X-MS-OLK-CONFTYPE:0 END:VEVENT END:VCALENDAR ___ -- 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
Re: [asterisk-users] C compiler cannot create executables when building zaptel
I believe I have all the necessary packages installed. Having done some research, one link suggests using strace and in that case I don't get the error: strace -f -o /tmp/trace -e trace=process ./configure ... configure: *** Zaptel build successfully configured *** That's from the end of the configure script. Can you post your config.log ? the config.log from strace -f -o /tmp/trace -e trace=process ./configure configure:2066: $? = 0 configure:2073: gcc -v 5 Using built-in specs. Target: i386-redhat-linux Configured with: ../configure --prefix=/usr --mandir=/usr/share/man --infodir=/usr/share/info --enable-shared --enable-threads=posix --enable-checking=release --with-system-zlib --enable-__cxa_atexit --disable-libunwind-exceptions --enable-libgcj-multifile --enable-languages=c,c++,objc,obj-c++,java,fortran,ada --enable-java-awt=gtk --disable-dssi --with-java-home=/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre --with-cpu=generic --host=i386-redhat-linux Thread model: posix gcc version 4.1.1 20070105 (Red Hat 4.1.1-51) configure:2076: $? = 0 configure:2083: gcc -V 5 gcc: '-V' option must have argument configure:2086: $? = 1 configure:2109: checking for C compiler default output file name configure:2136: gccconftest.c 5 configure:2139: $? = 0 configure:2177: result: a.out configure:2194: checking whether the C compiler works configure:2204: ./a.out configure:2207: $? = 0 configure:2224: result: yes configure:2231: checking whether we are cross compiling configure:2233: result: no configure:2236: checking for suffix of executables configure:2243: gcc -o conftestconftest.c 5 configure:2246: $? = 0 configure:2270: result: configure:2276: checking for suffix of object files configure:2302: gcc -c conftest.c 5 configure:2305: $? = 0 configure:2328: result: o configure:2332: checking whether we are using the GNU C compiler configure:2361: gcc -c conftest.c 5 configure:2367: $? = 0 configure:2384: result: yes configure:2389: checking whether gcc accepts -g configure:2419: gcc -c -g conftest.c 5 configure:2425: $? = 0 configure:2524: result: yes configure:2541: checking for gcc option to accept ISO C89 configure:2615: gcc -c -g -O2 conftest.c 5 configure:2621: $? = 0 configure:2644: result: none needed configure:2667: checking how to run the C preprocessor configure:2707: gcc -E conftest.c configure:2713: $? = 0 configure:2744: gcc -E conftest.c conftest.c:9:28: error: ac_nonexistent.h: No such file or directory configure:2750: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME | #define PACKAGE_TARNAME | #define PACKAGE_VERSION | #define PACKAGE_STRING | #define PACKAGE_BUGREPORT | #define _GNU_SOURCE 1 | /* end confdefs.h. */ | #include ac_nonexistent.h configure:2783: result: gcc -E configure:2812: gcc -E conftest.c configure:2818: $? = 0 configure:2849: gcc -E conftest.c conftest.c:9:28: error: ac_nonexistent.h: No such file or directory configure:2855: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME | #define PACKAGE_TARNAME | #define PACKAGE_VERSION | #define PACKAGE_STRING | #define PACKAGE_BUGREPORT | #define _GNU_SOURCE 1 | /* end confdefs.h. */ | #include ac_nonexistent.h configure:2936: checking for a BSD-compatible install configure:2992: result: /usr/bin/install -c configure:3003: checking whether ln -s works configure:3007: result: yes configure:3014: checking for GNU make configure:3029: result: make configure:3055: gcc -c -g -O2 conftest.c 5 configure:3061: $? = 0 configure:3087: checking for grep configure:3105: found /bin/grep configure:3118: result: /bin/grep configure:3128: checking for sh configure:3159: result: /bin/sh configure:3169: checking for ln configure:3187: found /bin/ln configure:3200: result: /bin/ln configure:3211: checking for wget configure:3229: found /usr/bin/wget configure:3242: result: /usr/bin/wget configure:3306: checking for grep that handles long lines and -e configure:3380: result: /bin/grep configure:3385: checking for egrep configure:3463: result: /bin/grep -E configure:3468: checking for ANSI C header files configure:3498: gcc -c -g -O2 conftest.c 5 configure:3504: $? = 0 configure:3603: gcc -o conftest -g -O2 conftest.c 5 configure:3606: $? = 0 configure:3612: ./conftest configure:3615: $? = 0 configure:3632: result: yes configure:3656: checking for sys/types.h configure:3677: gcc -c -g -O2 conftest.c 5 configure:3683: $? = 0 configure:3699: result: yes configure:3656: checking for sys/stat.h configure:3677: gcc -c -g -O2 conftest.c 5 configure:3683: $? = 0 configure:3699: result: yes configure:3656: checking for stdlib.h configure:3677: gcc -c -g -O2 conftest.c 5 configure:3683: $? = 0 configure:3699: result: yes configure:3656: checking for string.h configure:3677: gcc -c -g -O2 conftest.c 5 configure:3683: $? = 0 configure:3699: result: yes configure:3656: checking for memory.h configure:3677: gcc -c -g -O2 conftest.c 5 configure:3683: $? = 0
Re: [asterisk-users] Passing variables between two DUNDi/IAX2 peers
--- Craig Guy [EMAIL PROTECTED] wrote:
I believe that IAXVAR in Asterisk 1.6 will do what
you want. I have a
backport of this for Asterisk 1.2.14 or so floating
around somewhere but it
hasn't been maintained or used for months, may not
be compatible with the
1.6 implementation and I offer it with no support
whatsoever.
I'd like to give it a try if you can tell me where to
download it.
I noticed that overloading the EXTEN variable does
actually work but not if I use SIP's regcontext
feature and DUNDi's nopartial option.
I want to take advantage of regcontext but I also need
to pass a variable. Basically, I need to send the
UNIQUEID var to the remote peer for logging purposes
(and maybe other values).
If I use regcontext/nopartial and place a call to
4065^${UNIQUEID}, the asterisk log reports:
Mar 6 18:30:06 DEBUG[10457] pbx_dundi.c: Got
canonical message 13 (0), 96 bytes data
Mar 6 18:30:06 DEBUG[10457] pbx_dundi.c: Got
canonical message 1 (0), 42 bytes data
Mar 6 18:30:06 DEBUG[10457] pbx_dundi.c: Answering
query for '[EMAIL PROTECTED]'!
Mar 6 18:30:06 DEBUG[26691] pbx_dundi.c: Whee,
looking up '[EMAIL PROTECTED]' for
'00:1d:60:39:e9:1b'
Mar 6 18:30:06 DEBUG[26691] pbx_dundi.c: Registering
request for '[EMAIL PROTECTED]' on behalf of
'00:1d:60:39:e9:1b' crc ''
Mar 6 18:30:06 DEBUG[26691] db.c: Unable to find key
'001D6039E91B/4065^1204824606.83/priv/e' in
family 'dundi/cache'
Mar 6 18:30:06 DEBUG[26691] db.c: Unable to find key
'001D6039E91B/4065^1204824606.83/priv/e' in
family 'dundi/cache'
Mar 6 18:30:06 DEBUG[26691] db.c: Unable to find key
'001D6039E91B/4065^1204824606.83/priv/r001D6039E91B'
in family 'dundi/cache'
Mar 6 18:30:06 DEBUG[26691] db.c: Unable to find key
'hint/001D6039E91B/4/priv/e' in family
'dundi/cache'
(...cut to save space...)
Mar 6 18:30:06 DEBUG[26691] db.c: Unable to find key
'hint/001D6039E91B/4065^1204824606.83/priv/e'
in family 'dundi/cache'
Mar 6 18:30:06 DEBUG[26691] db.c: Unable to find key
'hint/001D6039E91B/4065^1204824606.83/priv/e'
in family 'dundi/cache'
Mar 6 18:30:06 DEBUG[26691] db.c: Unable to find key
'hint/001D6039E91B/4065^1204824606.83/priv/r001D6039E91B'
in family 'dundi/cache'
Mar 6 18:30:06 DEBUG[26691] pbx_dundi.c: Avoiding
'00:1d:60:39:e9:1b' in transaction
and it obviously fails because dundi is looking up
4065^1204824606.83 in regcontext but it doesn't
exist (4065 is the registered SIP extension).
So unless someone suggests a better solution, I'd be
happy to try out the 1.2 backport of IAXVAR (if you
can find it somewhere).
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Vieri
Sent: Thursday, 6 March 2008 2:57 AM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Passing variables
between two DUNDi/IAX2 peers
--- Richard Lyman [EMAIL PROTECTED] wrote:
Vieri wrote:
Hi.
I am trying to pass a variable from one Asterisk
PBX
to another.
I'm using DUNDi with IAX2. Is there a way to do
it?
I tried the following but it fails.
On peer1:
[dundi-outgoing]
switch = DUNDI/priv
exten = s,1,Set(CDR(userfield)=test)
exten = s,2,Set(DUNDIVAR=${ARG1}#TEST)
exten = s,3,NoOp(Passing ${DUNDIVAR} to DUNDi
peer.)
exten = s,4,Goto(${DUNDIVAR},1)
On peer2:
[dundi-incoming]
exten = _X.,1,NoOp(Received EXTEN ${EXTEN}.)
exten = _X.,n,Set(EXTTODIAL=${CUT(EXTEN|#|1)})
exten = _X.,n,Set(DUNDIVAR=${CUT(EXTEN|#|2)})
exten = _X.,1,NoOp(Extracted extension
${EXTTODIAL}
and DUNDi variable ${DUNDIVAR})
exten =
_X.,n,Goto(local-extensions,${EXTTODIAL},1)
If I try a test call then nothing ever reaches
peer2.
However, if I remove #TEST from DUNDIVAR in
dundi-outgoing and
Goto(local-extensions,${EXTEN},1)
in dundi-incoming then the call is established
correctly.
I guess the _X. pattern match is wrong?
How can I match an alphanumeric string?
Thanks,
Vieri
you would have to use type 'friend' as user/peer
do
not pass channel
variables (unless it has been changed in
1.4/1.6/trunk).
In iax.conf I have (on both peers):
[priv]
type=friend
dbsecret=dundi/secret
context=dundi-incoming
and I am running Asterisk 1.2.21.1 on peer1 and
1.2.26.2 on peer2.
Any ideas as to why it's not working?
Or could anyone please suggest an alternative
method?
Thanks!
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[asterisk-users] Provider recommendation in USA
Hi, I would like to seek an opinion or list of providers in USA or particularly in California. We would need someone who can offer maximum ports and lowest rates. Thanks very much, Vivek ___ -- 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] Allowguest=yes language
I have an Asterisk server with voicemail(), in the sip.conf I have: [general] allowguest=yes language=pt_BR I have the sound files for pt_BR in /var/lib/asterisk/sounds/pt_BR, and the others dirs (dgits, phonetic and so on). The problem I have is: when a guest tries to place a call and is directed to the voicemail, the sounds are in english and not in the default language, how can I change this ? Best Regards, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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
Re: [asterisk-users] Provider recommendation in USA
Vivek, What do you need, DID or Termination? BTW We are in California. Send me you Contact info and we can discuss more about your needs. -Jai On Thu, Mar 6, 2008 at 10:25 AM, Vivek Shrivastava [EMAIL PROTECTED] wrote: Hi, I would like to seek an opinion or list of providers in USA or particularly in California. We would need someone who can offer maximum ports and lowest rates. Thanks very much, Vivek ___ -- 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 ___ -- 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
Re: [asterisk-users] C compiler cannot create executables when building zaptel
On Thu, Mar 06, 2008 at 06:50:40AM +1300, CSB wrote: When attempting to build zaptel I get the following error: configure:2184: error: C compiler cannot create executables Where do you actually get the error from? From the 'make' command? If so: go chase errors in menuselect/configure -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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
Re: [asterisk-users] OT How to Change Polycom Web Admin User:Pass via Web
I am using version 2.2.0. __Yehavi: Date: Thu, 6 Mar 2008 15:01:26 +1100 From: Lee, John (Sydney) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How to Change Polycom Web Admin User:Pass As far as I recall it can be done from the config file only. Here is the relevant line from sip.cfg: device device.set=1 device.auth.localAdminPassword.set=1 device.auth.loc alAdminPassword=YOUR-PASSWORD-HERE / What sip release are you referring to? I am looking at sip 1.6.x and sip.cfg only allows you to set the length of the user and admin passwords. You cannot set the password in cfg. ___ -- 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 ___ -- 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
Re: [asterisk-users] Best Java library to interact with Asterisk
Asterisk-java, http://asterisk-java.org is a very good one, it has a pretty good documentation. On Thu, Mar 6, 2008 at 1:01 PM, equis software [EMAIL PROTECTED] wrote: Hi, I need to interact with my Asterisk and need a good Java class library. What do you think is the best? Thanks ___ -- 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 -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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
Re: [asterisk-users] C compiler cannot create executables when building zaptel
When attempting to build zaptel I get the following error: configure:2184: error: C compiler cannot create executables Where do you actually get the error from? From the 'make' command? If so: go chase errors in menuselect/configure ./configure Cameron ___ -- 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] zaptel compile question
Hi all, I am wanting to use an option from the ./configure script with zaptel to compile zaptel for a different kernel than the running kernel. How do I do that exactly. Example: Current kernel is 2.6.18-8.1.4.el5 and I want to compile zaptel for 2.6.18-53.1.4.el5 I am using centos 5.1 or RHEL 5.1 So when I reboot zaptel is ready to go. How do I do that? Thanks, Jerry ___ -- 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] Net Neutrality
What do we want NET NEUTRALITY When do we want it? NOW AND FOREVER This video should be compulsory viewing for everyone in public office not just here in the USA but globally so this public resource cant be stolen from you !! http://deancollinsblog.blogspot.com/2008/03/net-neutrality.html http://deancollinsblog.blogspot.com/2008/03/net-neutrality.html Please pass it on. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ -- 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
Re: [asterisk-users] zaptel compile question
Jerry Geis wrote: I am wanting to use an option from the ./configure script with zaptel to compile zaptel for a different kernel than the running kernel. How do I do that exactly. Example: Current kernel is 2.6.18-8.1.4.el5 and I want to compile zaptel for 2.6.18-53.1.4.el5 I am using centos 5.1 or RHEL 5.1 So when I reboot zaptel is ready to go. How do I do that? There might be a better way, and I don't use ./configure script options, but what I do is set the KVERS variable before building. So in your case, I would export KVERS=2.6.18-53.1.4.el5 make install Shaun ___ -- 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] Asterisk 1.4 w/ realtime static zapata
i've been using *1.2 w/ realtime static zapata in mysql table fine. but after i upgraded to 1.4. it seems like the zapata table doesn't load correctly. i have to go in the console and use the zap restart to get the zap channels register. is this sounds like a bug or something i'm missing when upgrading to 1.4? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- 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] DTMFR2- UNICALL
Hi Asterisk-user, Steve; I´m using *libmfcr2-0.0.3.tar.gz, libsupertone-0.0.2.tar.gz, libunicall-0.0.3.tar.gz,spandsp-0.0.3pre22.tgz* with Fedora core 6 ,Asterisk 1.2.14, libpri 1.2.4 , zaptel 1.2.20; So everything is working perfectly with MFCR2, but sometimes i have problems with MultiFrame Alignment Signal (MFAS),i´m using standard G.732..I would like to Know, where can i mofify this?? Regards, Jessi ___ -- 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
Re: [asterisk-users] zaptel compile question
On Thu, Mar 06, 2008 at 03:19:21PM -0500, Jerry Geis wrote: Hi all, I am wanting to use an option from the ./configure script with zaptel to compile zaptel for a different kernel than the running kernel. How do I do that exactly. Example: Current kernel is 2.6.18-8.1.4.el5 and I want to compile zaptel for 2.6.18-53.1.4.el5 I am using centos 5.1 or RHEL 5.1 You can just set KVERS: make KVERS=2.6.18-53.1.4.el5 Assuming you have the respective kernel-devel package installed. See also: http://zaptel.tzafrir.org.il/#_kernel_source_headers (Which is the HTML-zed version of the README file in Zaptel) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] en25.com
Hi, Are those messages from [EMAIL PROTECTED] messages from Digium sent via some kind of spamming service? I did not subscribe to anything at en25.com so why would I have to unsubscribe? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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
Re: [asterisk-users] format of UNIQUEID variable
--- Tony Mountifield [EMAIL PROTECTED] wrote: In Asterisk 1.4 or later, an optional system name can be defined in asterisk.conf, and if defined, the unique ID becomes: system_name-timestamp.seq_num Thanks! So for the sake of backward compatibility, if I dont' define sysname in 1.4 then the uniqueid will be just like in 1.2. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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
Re: [asterisk-users] Asterisk in the call center - how do you do it?
On Thu, Mar 06, 2008 at 11:02:50AM +1100, Paul Hales wrote: And we found (recently) that if you send the right http packet to a snom phone you can make the screen say Agent 155 rather than the extension number. :) Or, y'know, INSERT COIN. http://www.hackszine.com/blog/archive/2007/10/change_the_message_on_hp_print.html?CMP=OTC-7G2N43923558 http://www.odetocode.com/Humor/68.aspx Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Cool New Website
Cool New Website For everyone to see! I think they are using a specially programmed version of Asterisk to do this. www.dialaway4free.com ___ -- 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
Re: [asterisk-users] Receiving double DTMF if I pressed 1, then asterisk box recognize it 11
Dear Michael; This problem happens even if I am in Zapata level (did not use any SIP trunk), it happens when I am calling to the asterisk box and need to enter the extension, then it reads the digit duplicated. Any advise? Regards Bilal -- I believe you need to set in the sip.conf the setting dtmfmode to either inband or rfc2833 for the connection. Michael Cargile Software Developer Explido Software USA Inc. www.explido.us On Wed, 2008-02-20 at 11:00 -0800, bilal ghayyad wrote: Hi All; I read below about resolving the problem of receiving the digit duplicated (for example, if u press 1 then asterisk see it 11), the below note helping to resolve it, but I did not understand how I can be able to apply it? Any help to apply the below: If you appear to be receiving doubled DTMF signals then you are likely to get both inband and RFC2833 or SIP INFO signalling on your Asterisk box; you will want to make the sening party use only one of these two methods. Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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
Re: [asterisk-users] Call Manager as trunk
I can't believe I fixed the problem, but here's what I did: 1. Checked the Use Media Termination Point in the profile for the SIP trunk in Call Manager. 2. Split the SIP config for Call Manager into separate inbound and outbound settings like so: 3. Added the insecure=very to the callmanout section. [callmanout] type=peer context=incoming insecure=very host=(ip of server) disallow=all allow=ulaw allow=alaw nat=no canreinvite=yes qualify=yes [callmanin] host=(ip of server) type=user context=incoming And suddenly it's working great! Aaron On Thu, Mar 6, 2008 at 9:54 AM, Aaron Fransen [EMAIL PROTECTED] wrote: I have Asterisk 1.4 tied via SIP to a Cisco Callmanager 6.1 system. Calls between the systems (ie. extension to extension) work perfectly. However when I attempt to make an outside call from an Asterisk extension through Call Manager to the outside world, it connects but only for a few seconds, and on the Asterisk console I get: Got SIP response 503 Service Unavailable back from (ip of call manager) Coming the other way, if I call into the Call Manager system (from my cell to be exact), then transfer my call to the Asterisk SIP phone (an Aastra 57i), on the cell I can hear the voice on the Aastra, but the Aastra can only hear the Asterisk music on hold! As I mentioned though, going the other way and calling out from Asterisk to my cell works perfectly...for between 5 and 10 seconds (it varies), then disconnects with the above error. My sip.conf looks like this: [callman] type=friend context=incoming host=(ip of call manager) disallow=all allow=ulaw allow=alaw nat=yes canreinvite=yes qualify=yes I've tried experimenting with the externip and localnet parameters to no effect. Any ideas? ___ -- 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
Re: [asterisk-users] Cool New Website
Apart from the BS crap about patent pending - looks like a great service and I'm sure they'll get a ton of traffic. Good use of technology. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Goran Donev Sent: Thursday, 6 March 2008 5:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cool New Website Cool New Website For everyone to see! I think they are using a specially programmed version of Asterisk to do this. www.dialaway4free.com ___ -- 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 ___ -- 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
Re: [asterisk-users] FXS channel banks
I have been told to use Rhino Channel Bank but I am yet to set it up and I appreciate if someone can show me some doco of using Rhino on an E1/T1 with TE410. Thanks. I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. ___ -- 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] OT: Upgrade Addpac AP200C
Hi guys, I have made a upgrade to my addpac ap200c, however it does not upload complete, now I can load addpac. Is there anyway that can I upload the old firwmare? Any help is appreciated. System Boot Loader, Version 2.2.5/DUAL(852) Copyright (c) by AddPac Technology Co., Ltd. Since 1999. System Bootstrap, Version 1.2 Decompressing the image: # ___ -- 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
Re: [asterisk-users] Cool New Website
yeah, /Searching US Patents Text Collection.../ *Results of Search in US Patents Text Collection db for: dialaway4free*: 0 patents. No patents have matched your query Original post also sounds a bit spammy to me... *shrug* Brooks R. Bridges Telecommunications Manager Ifbyphone, Inc. Phone: (847) 983-3000 Fax: (847) 676-6553 [EMAIL PROTECTED] http://www.ifbyphone.com Dean Collins wrote: Apart from the BS crap about patent pending - looks like a great service and I'm sure they'll get a ton of traffic. Good use of technology. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Goran Donev Sent: Thursday, 6 March 2008 5:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cool New Website Cool New Website For everyone to see! I think they are using a specially programmed version of Asterisk to do this. www.dialaway4free.com ___ -- 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] OT: Upgrade Addpac AP200C
Hi guys, I have made a upgrade to my addpac ap200c, however it does not upload complete, now I can not load addpac. Is there anyway that I can upload the old firwmare? Any help is appreciated. System Boot Loader, Version 2.2.5/DUAL(852) Copyright (c) by AddPac Technology Co., Ltd. Since 1999. System Bootstrap, Version 1.2 Decompressing the image: # ___ -- 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
Re: [asterisk-users] FXS channel banks
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What kind of information are you looking for? configuration or? If you look in our manuals our cards and the Digium cards configure the same in zaptel and zapata. Lee, John (Sydney) wrote: I have been told to use Rhino Channel Bank but I am yet to set it up and I appreciate if someone can show me some doco of using Rhino on an E1/T1 with TE410. Thanks. I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. ___ -- 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 !DSPAM:47d0790d14234975420232! - -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, Come with a Money Back gurantee and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH0HucdloC7YyaIOoRAv5zAJ9jdZQEkXbYfbvP7QbONR+DVVYSdQCfSmmb dv00H0l/fgJiTU9o4Z6++9Y= =wDLO -END PGP SIGNATURE- ___ -- 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
Re: [asterisk-users] Had it with Dell Garbage
Ex Vito wrote: On Tue, Feb 26, 2008 at 10:51 PM, Joshua Kinard [EMAIL PROTECTED] wrote: Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very likely, 380's as well). I just learned this the hard way. --J ...can you expand on that please ? I'm on my way to getting one of the newer Digium TE220B PCIe dual T1/E1 to put on such a system. So far I'm having nothing but problems with my DL360's and TE220B's. While many of the problems are slowly starting to seem to like problems with the telco, some of them definitely aren't. I can't pass the loop back test (patlooptest) unless the hpasmd (the system management software) has been stopped. Though I will try asterisk and zaptel 1.4 soon to see if that helps. ___ -- 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
Re: [asterisk-users] Cool New Website
I think they are using a specially programmed version of Asterisk to do this. Don't you mean: I am using a specially programmed version of Asterisk to do this. ? domain: dialaway4free.com created: 16-Jan-2008 last-changed:16-Jan-2008 registration-expiration: 16-Jan-2009 registrant-firstname:Goran registrant-lastname: Donev registrant-organization: Donev Technology Consulting Inc Also, clean up your grammar and spelling errors on the site if you want anyone to take it seriously. It's a good idea, and I hope you go far with it, but geez, that site looks like it was written by an over-caffeinated 12 year old. Brooks R. Bridges Telecommunications Manager Ifbyphone, Inc. Phone: (847) 983-3000 Fax: (847) 676-6553 [EMAIL PROTECTED] http://www.ifbyphone.com Goran Donev wrote: Cool New Website For everyone to see! I think they are using a specially programmed version of Asterisk to do this. www.dialaway4free.com ___ -- 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 ___ -- 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
Re: [asterisk-users] FXS channel banks
I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. I know I've missed the original message in this thread, so it'll be a bit out of place, but what about the Xorcom Channel banks? e.g.: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? However, even with E1 units, you're still looking at 7 E1 ports... (2 quad cards + the external channel bank) Gordon ___ -- 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
Re: [asterisk-users] Cool New Website
On Thu, 6 Mar 2008 17:54:04 -0500, Dean Collins wrote: Apart from the BS crap about patent pending - looks like a great service and I'm sure they'll get a ton of traffic. Good use of technology. Y'think? I have no patience for such adverts. It even bugs me to have to listen to the Talkshoe self-promo stuff when I miss a VOIP Users Conference and download the MP3 recording. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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
Re: [asterisk-users] Newbie Polycom: IP600 Headset Problem
We had a similar issue where the connector was not pushed in hard enough. I know that sounds like a joke, but it isn't! PaulH On Thu, 2008-03-06 at 18:27 +1100, Lee, John (Sydney) wrote: I have been testing with Polycom IP600 phones for a month or so. I found out that there are frequent problems with the handset. The problem is I can hear the other end but the other end cannot hear me. I have already downloaded the latest bootROM 3.1.3 and sip 2.1.2 However, there are no problems with the headset or speaker phone. Has anyone encountered such problems before? Thanks. ___ -- 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 ___ -- 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
Re: [asterisk-users] FXS channel banks
On Thu, Mar 06, 2008 at 11:50:43PM +, Gordon Henderson wrote: I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. I know I've missed the original message in this thread, so it'll be a bit out of place, but what about the Xorcom Channel banks? e.g.: http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html Trouble is, you'll need 7 32-port units to cover your needs and I'm not sure if USB2 is up to driving that many ... Tzafrir? One USB connector can take a number close to that easily. But even if USB were the bottleneck, you would just add another USB controller in the form of PCI card and get extra bandwidth. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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
Re: [asterisk-users] Had it with Dell Garbage
...can you expand on that please ? I'm on my way to getting one of the newer Digium TE220B PCIe dual T1/E1 to put on such a system. I know the subject line was anti-Dell, but just to put in a data point: We have 10 Dell PE2950's running with one or two TE220B's per system, and they have given us no problems. We did play with IRQs in the BIOS, but not sure if that was actually needed. Ron ___ -- 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
Re: [asterisk-users] DTMFR2- UNICALL
What kind of problems are you talking about and what you want to modify? On Thu, Mar 6, 2008 at 2:42 PM, Jessica Gonzalez Arriagada [EMAIL PROTECTED] wrote: Hi Asterisk-user, Steve; I´m using libmfcr2-0.0.3.tar.gz, libsupertone-0.0.2.tar.gz,libunicall-0.0.3.tar.gz,spandsp-0.0.3pre22.tgz with Fedora core 6 ,Asterisk 1.2.14, libpri 1.2.4 , zaptel 1.2.20; So everything is working perfectly with MFCR2, but sometimes i have problems with MultiFrame Alignment Signal (MFAS),i´m using standard G.732..I would like to Know, where can i mofify this?? Regards, Jessi ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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
Re: [asterisk-users] Had it with Dell Garbage
We have found that 860's with Te120p's seem to work well too. PaulH On Thu, 2008-03-06 at 19:29 -0500, Ron Joffe wrote: ...can you expand on that please ? I'm on my way to getting one of the newer Digium TE220B PCIe dual T1/E1 to put on such a system. I know the subject line was anti-Dell, but just to put in a data point: We have 10 Dell PE2950's running with one or two TE220B's per system, and they have given us no problems. We did play with IRQs in the BIOS, but not sure if that was actually needed. Ron ___ -- 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 ___ -- 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] is this possible..
I'm head of RD for a dot com company and we are looking to create a prototype using asterisk. Basically we people who visit our site and search for goods listed by other people. Once something is found, a phone number is listed in the results and person A calls person B to see if the item is available, cost, etc. I'd like for the person searching to be able to click on 10 items they are interested in then click another button which would have asterisk start at the first, call person B, ask if the item is available, if yes, then call person A and connect the two, if not, it says thanks, and calls the next person on the list. Is this possible with Asterisk? Second, anyone looking for some contract work to help get this prototype running? Thanks! ___ -- 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] How to return the status of a call to the calling server?
I think tilghman hacked out something like this in less time than it took
me to search through 20 pages of googlegook trying unsuccessfully to find
it :)
A caller calls host A. They select a service provided by host B which is
invoked using dial(iax2/[EMAIL PROTECTED]/${EXTEN}|2|g).
The service on host B needs to return a single digit to host A's dialplan.
(I really only need 1 bit that I can interpret as this or that.)
Executing hangup(${EXTEN}) on host B and testing ${HANGUPCAUSE} on host A
seemed obvious to me. Sad to say, it doesn't to the developers :(
Abusing CALLERID(name) didn't work either.
Writing a row to a MySQL database seems so cumbersome.
This is for 1.2 if it matters.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Newbie Polycom: IP600 Headset Problem
We had a similar issue where the connector was not pushed in hard enough. I know that sounds like a joke, but it isn't! PaulH Thanks Paul - it also happened to my phone! Thanks so much. ___ -- 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
Re: [asterisk-users] is this possible..
blackwater dev wrote: I'm head of RD for a dot com company and we are looking to create a prototype using asterisk. Basically we people who visit our site and search for goods listed by other people. Once something is found, a phone number is listed in the results and person A calls person B to see if the item is available, cost, etc. I'd like for the person searching to be able to click on 10 items they are interested in then click another button which would have asterisk start at the first, call person B, ask if the item is available, if yes, then call person A and connect the two, if not, it says thanks, and calls the next person on the list. Is this possible with Asterisk? Second, anyone looking for some contract work to help get this prototype running? ok i have to say it.. Go Fish! ___ -- 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
Re: [asterisk-users] is this possible..
I really see this is useless since we alreadu got pricegrabbers buy.com and froogle they all list the itme in stock on the site there is really no need for a $30k a year operator to read it for the person. just my $0.02 On 3/6/08, blackwater dev [EMAIL PROTECTED] wrote: I'm head of RD for a dot com company and we are looking to create a prototype using asterisk. Basically we people who visit our site and search for goods listed by other people. Once something is found, a phone number is listed in the results and person A calls person B to see if the item is available, cost, etc. I'd like for the person searching to be able to click on 10 items they are interested in then click another button which would have asterisk start at the first, call person B, ask if the item is available, if yes, then call person A and connect the two, if not, it says thanks, and calls the next person on the list. Is this possible with Asterisk? Second, anyone looking for some contract work to help get this prototype running? Thanks! ___ -- 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
Re: [asterisk-users] is this possible..
I think he's talking about an automated system. It's definitely possible with asterisk, whether or not it's a good idea. I really see this is useless since we alreadu got pricegrabbers buy.com and froogle they all list the itme in stock on the site there is really no need for a $30k a year operator to read it for the person. just my $0.02 On 3/6/08, blackwater dev [EMAIL PROTECTED] wrote: I'm head of RD for a dot com company and we are looking to create a prototype using asterisk. Basically we people who visit our site and search for goods listed by other people. Once something is found, a phone number is listed in the results and person A calls person B to see if the item is available, cost, etc. I'd like for the person searching to be able to click on 10 items they are interested in then click another button which would have asterisk start at the first, call person B, ask if the item is available, if yes, then call person A and connect the two, if not, it says thanks, and calls the next person on the list. Is this possible with Asterisk? Second, anyone looking for some contract work to help get this prototype running? Thanks! ___ -- 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 ___ -- 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
Re: [asterisk-users] is this possible..
It's certainly possible, and I would be interested in helping you get it going. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of blackwater dev Sent: Thursday, March 06, 2008 8:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] is this possible.. I'm head of RD for a dot com company and we are looking to create a prototype using asterisk. Basically we people who visit our site and search for goods listed by other people. Once something is found, a phone number is listed in the results and person A calls person B to see if the item is available, cost, etc. I'd like for the person searching to be able to click on 10 items they are interested in then click another button which would have asterisk start at the first, call person B, ask if the item is available, if yes, then call person A and connect the two, if not, it says thanks, and calls the next person on the list. Is this possible with Asterisk? Second, anyone looking for some contract work to help get this prototype running? Thanks! ___ -- 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
Re: [asterisk-users] Newbie Polycom: IP600 Headset Problem
It was one of those moments in life where I felt a lot less smart than I usually do... PaulH On Fri, 2008-03-07 at 15:28 +1100, Lee, John (Sydney) wrote: We had a similar issue where the connector was not pushed in hard enough. I know that sounds like a joke, but it isn't! PaulH Thanks Paul - it also happened to my phone! Thanks so much. ___ -- 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 ___ -- 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] Call flows of conference
Hi, I have an astrisk pbx installed on my system and i have registered two Aastra hardphones and one SJPhone(softphone) with that. Then i tested the following scenario A(Aastra) calledB(Aastra) B answered the call I pressed conference button on the A ( A put B on hold) A called C(SJPhone) (It send an invite with isfocus ) C answered the call I pressed conference button on the A again A B and C came in conference mode. Then when I hangup the phone A , call between the B and C is also disconnected. Any one could you explain me this scenario with the sip message sequence? What is the message sequence of a pbx centered conference? Thanks in advance. Preethy ___ -- 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
Re: [asterisk-users] Cool New Website
On Fri, Mar 7, 2008 at 12:58 AM, Michael Graves [EMAIL PROTECTED] wrote: Y'think? I have no patience for such adverts. It even bugs me to have to listen to the Talkshoe self-promo stuff when I miss a VOIP Users Conference and download the MP3 recording. Download and scrub. Repeat. :) ___ -- 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
Re: [asterisk-users] Provider recommendation in USA
On Thu, Mar 6, 2008 at 7:25 PM, Vivek Shrivastava [EMAIL PROTECTED] wrote: I would like to seek an opinion or list of providers in USA or particularly in California. We would need someone who can offer maximum ports and lowest rates. The usual suspects IMO (random order): Teliax, Nufone, Voicepulse, Junction I have accounts at all four of the above and have had them for a few years. Some others I've used were around and disappeared. Two others I still have don't come up to scratch anymore but I have a significant remaining credit. /r ___ -- 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] Newbie MeetMe: How to control max users in conference?
I was successful to control the max users (10) if I hardcode the
conference room number (in this case 101) as follows:
exten = 8600,1,Playback(conf-thereare)
exten = 8600,2,MeetMeCount(101)
exten = 8600,3,Playback(conf-peopleinconf)
exten = 8600,4,MeetMeCount(101,CONFCOUNT)
exten = 8600,5,GotoIf($[${CONFCOUNT} = 10]?6:9)
exten = 8600,6,MeetMe(101,i)
exten = 8600,7,Playback(vm-goodbye)
exten = 8600,8,Hangup()
exten = 8600,9,Playback(conf-full)
exten = 8600,10,Playback(vm-goodbye)
exten = 8600,11,Hangup()
On the other hand, I thought it would be a better idea if I just allow
user to enter their conf number when they dial a generic meetme
extension.
exten = 8600,1,MeetMe(,ciMps)
exten = 8600,2,Playback(vm-goodbye)
exten = 8600,3,Hangup()
However, if I choose the 2nd option, I do not seem to be able to control
the max number of users.
Any thoughts on how I can achieve this?
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Re: [asterisk-users] LDAP
Gonzalo, Please let us know what you mean by 'stops working' - it should spit out errors or wrong queries to ldap. Also please keep this list in your replies. I have no problems answering personal emails but both of us might get more feedback if we post our progress on the list! :) Quoting Gonzalo Servat [EMAIL PROTECTED]: Hi again :) I've downloaded, compiled installed 1.6.0-beta4 --with-ldap. After a few hours of messing with it, I've managed to get it to say that it has connected successfully to the LDAP backend (by looking at the output of realtime ldap status). I've modified extconfig.conf to what it should be (after reading many different configs on the subject). The trouble I'm having now is actually authenticating with a SIP user. I am running slapd in debug mode (slapd -d 4095) and I would have expected to see lots of activity on the console when I attempt to authenticate as a SIP user, but I see none at all. Is this normal? Thanks! Regards, Gonzalo On Thu, Mar 6, 2008 at 12:37 AM, Gonzalo Servat [EMAIL PROTECTED] wrote: Hi All, I've just compiled Asterisk 1.4.18 and I'm planning on using an LDAP tree where the users will each have their account, SIP username/password, extension number, context, etc. My first question is: can this be done with 1.4.x? If so, where can I get the res_config_ldap from?? I googled quite a bit and found a res_config_ldap that looks to be coded for 1.2. Is anyone running Asterisk with LDAP? Is it stable? Thanks in advance. Regards, Gonzalo -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz ___ -- 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
