[asterisk-users] web app to playback recorded phone calls.
1 of our customers records all phone calls and needs to be able to be played back via a searchable web app. I tried ARI but it is very limited. Anyone have any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk queue and agents
Has this been corrected? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Wednesday, March 07, 2007 11:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk queue and agents BJ Here is the sip.conf file. Hints work great. The only problem is the queue is sending calls to an agent that's on the phone. [general] rtcachefriends=yes videosupport=yes port=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) context=sip ; Send unknown SIP callers to this context allow=g729 allow=h263 ; H.263 is our video codec allow=h263p ; H.263p is the enhanced video codec ;allow=g711 ;allow=all ;allow=ulaw ;allow=gsm nat=1 host=dynamic type=peer qualify=yes notifyringing=yes Subscribecontext=sip call-limit=300 notifyhold = yes limitonpeer = yes notifyringing = yes; Notify subscriptions on RINGING state (default: no) notifyhold = yes [56405] ;Eric Test type=friend ; "friend" means this device takes and makes calls username=1 ; Username on device callerid=Eric Test Phone <56405> secret=56405; Password for device host=dynamic ; This host is not on the same IP addr every time context=sip ; Inbound calls from this host go here [EMAIL PROTECTED]; Activate the message waiting light if this canreinvite=no; Leave this alone for now; see archives for details nat=1 qualify=yes Subscribecontext=sip notifyringing=yes call-limit=300 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, March 07, 2007 10:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents I don't think this is a bug. From UPGRADE.txt: * Queues depend on the channel driver reporting the proper state for each member of the queue. To get proper signalling on queue members that use the SIP channel driver, you need to enable a call limit (could be set to a high value so it is not put into action) and also make sure that both inbound and outbound calls are accounted for. Example: [general] limitonpeer = yes [peername] type=friend call-limit=10 Please test with that and report your findings, and if it's still not working find us on IRC as we'd like to take a further look and see what might be wrong. BJ On 3/7/07, Hall, Eric M. <[EMAIL PROTECTED]> wrote: > Looks like it's a bug > > http://bugs.digium.com/view.php?id=9172&nbn=3 > > I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and > report back to the list. > > > > Eric Hall > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Octavio > Ruiz (Ta^3) > Sent: Wednesday, March 07, 2007 1:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk queue and agents > > >Have a question for the group > >If I have an agent is on the phone outside of the queue should that > > person > >still get queue calls ? > >Doing a show agents online I see Available however show hints I see > > inuse. > > There is a ringinuse feature for SIP devices on 1.4.X which is what you are > looking for. > > -- > Octavio Ruiz Cervera > Neocenter, SA. de CV. > http://www.neocenter.com/ > Soluciones para Centros de Contacto y Telefonía IP > Tel.: (+52 55) 8590-9000 Ext. 9016 > Cel.: (+55 55) 5514-087790 > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hints not working using SVN-branch-1.4-r59289 **** FIXED ****
Just wanted to update the list I found the problem. In my extensions.conf I had exten => 21,hint(SIP/21) It should be exten => 21,hint,SIP/21 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Wednesday, April 04, 2007 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Hints not working using SVN-branch-1.4-r59289 Just wanted to update the group. I copied the config file to a working server and the hints worked without any problems. Can anyone tell me if they have a working system using hits and SVN-branch-1.4-r59289 or newer. Eric Hall From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Tuesday, April 03, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hints not working using SVN-branch-1.4-r59289 Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI> show hints -= Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 2 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 4 --- - 6 hints registered Here is the sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes; Notify subscriptions on HOLD state (default: no) limitonpeers=yes allow=ulaw [21] ;Bill Salmons type=peer username=21 callerid=Bill Salmons <21> secret=21 host=dynamic context=default mailbox=21 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=30 [23] ;Teresa Trautman type=peer username=23 callerid=Teresa Trautman <23> secret=23 host=dynamic context=default mailbox=23 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [25] ;Bill Goldsmith type=peer username=25 callerid=Bill Goldsmith <25> secret=25 host=dynamic context=default mailbox=25 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [26] ;Joelle Harris type=peer username=26 callerid=Joelle Harris <26> secret=26 host=dynamic context=default mailbox=26 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [29] ;Amanda Anderson type=peer username=29 callerid=Amanda Anderson <29> secret=29 host=dynamic context=default mailbox=29 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [30] ;Joelle Harris type=peer username=30 callerid=Liz Williamson <30> secret=30 host=dynamic context=default mailbox=30 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [ata] type=peer username=ata host=dynamic context=default secret=ata here is the extensions.conf [default] include => parkedcalls exten => 21,hint(SIP/21) exten => 21,1,answer exten => 21,n,dial(sip/21|30|kw) exten => 21,n,voicemail([EMAIL PROTECTED]|u) exten => 23,hint(sip/23) exten => 23,1,answer exten => 23,n,dial(sip/23|30|kw) exten => 23,n,voicemail([EMAIL PROTECTED]|u) exten => 25,hint(SIP/25) exten => 25,1,answer exten => 25,n,dial(sip/25|30|kw) exten => 25,n,voicemail([EMAIL PROTECTED]|u) exten => 26,hint(SIP/26) exten => 26,1,answer exten => 26,n,dial(sip/26|30|kw) exten => 26,n,voicemail([EMAIL PROTECTED]|u) exten => 29,hint(SIP/29) exten => 29,1,answer exten => 29,n,dial(sip/29|30|kw) exten => 29,n,voicemail([EMAIL PROTECTED]|u) exten => 30,hint(SIP/30) exten => 30,1,answer exten => 30,
RE: [asterisk-users] Hints not working using SVN-branch-1.4-r59289
Just wanted to update the group. I copied the config file to a working server and the hints worked without any problems. Can anyone tell me if they have a working system using hits and SVN-branch-1.4-r59289 or newer. Eric Hall From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Tuesday, April 03, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hints not working using SVN-branch-1.4-r59289 Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI> show hints -= Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 2 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 4 --- - 6 hints registered Here is the sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes; Notify subscriptions on HOLD state (default: no) limitonpeers=yes allow=ulaw [21] ;Bill Salmons type=peer username=21 callerid=Bill Salmons <21> secret=21 host=dynamic context=default mailbox=21 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=30 [23] ;Teresa Trautman type=peer username=23 callerid=Teresa Trautman <23> secret=23 host=dynamic context=default mailbox=23 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [25] ;Bill Goldsmith type=peer username=25 callerid=Bill Goldsmith <25> secret=25 host=dynamic context=default mailbox=25 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [26] ;Joelle Harris type=peer username=26 callerid=Joelle Harris <26> secret=26 host=dynamic context=default mailbox=26 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [29] ;Amanda Anderson type=peer username=29 callerid=Amanda Anderson <29> secret=29 host=dynamic context=default mailbox=29 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [30] ;Joelle Harris type=peer username=30 callerid=Liz Williamson <30> secret=30 host=dynamic context=default mailbox=30 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [ata] type=peer username=ata host=dynamic context=default secret=ata here is the extensions.conf [default] include => parkedcalls exten => 21,hint(SIP/21) exten => 21,1,answer exten => 21,n,dial(sip/21|30|kw) exten => 21,n,voicemail([EMAIL PROTECTED]|u) exten => 23,hint(sip/23) exten => 23,1,answer exten => 23,n,dial(sip/23|30|kw) exten => 23,n,voicemail([EMAIL PROTECTED]|u) exten => 25,hint(SIP/25) exten => 25,1,answer exten => 25,n,dial(sip/25|30|kw) exten => 25,n,voicemail([EMAIL PROTECTED]|u) exten => 26,hint(SIP/26) exten => 26,1,answer exten => 26,n,dial(sip/26|30|kw) exten => 26,n,voicemail([EMAIL PROTECTED]|u) exten => 29,hint(SIP/29) exten => 29,1,answer exten => 29,n,dial(sip/29|30|kw) exten => 29,n,voicemail([EMAIL PROTECTED]|u) exten => 30,hint(SIP/30) exten => 30,1,answer exten => 30,n,dial(sip/30|30|kw) exten => 30,n,voicemail([EMAIL PROTECTED]|u) -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.25/744 - Release Date: 4/3/2007 5:32 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints not working using SVN-branch-1.4-r59289
Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI> show hints -= Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 2 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 4 --- - 6 hints registered Here is the sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes; Notify subscriptions on HOLD state (default: no) limitonpeers=yes allow=ulaw [21] ;Bill Salmons type=peer username=21 callerid=Bill Salmons <21> secret=21 host=dynamic context=default mailbox=21 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=30 [23] ;Teresa Trautman type=peer username=23 callerid=Teresa Trautman <23> secret=23 host=dynamic context=default mailbox=23 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [25] ;Bill Goldsmith type=peer username=25 callerid=Bill Goldsmith <25> secret=25 host=dynamic context=default mailbox=25 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [26] ;Joelle Harris type=peer username=26 callerid=Joelle Harris <26> secret=26 host=dynamic context=default mailbox=26 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [29] ;Amanda Anderson type=peer username=29 callerid=Amanda Anderson <29> secret=29 host=dynamic context=default mailbox=29 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [30] ;Joelle Harris type=peer username=30 callerid=Liz Williamson <30> secret=30 host=dynamic context=default mailbox=30 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [ata] type=peer username=ata host=dynamic context=default secret=ata here is the extensions.conf [default] include => parkedcalls exten => 21,hint(SIP/21) exten => 21,1,answer exten => 21,n,dial(sip/21|30|kw) exten => 21,n,voicemail([EMAIL PROTECTED]|u) exten => 23,hint(sip/23) exten => 23,1,answer exten => 23,n,dial(sip/23|30|kw) exten => 23,n,voicemail([EMAIL PROTECTED]|u) exten => 25,hint(SIP/25) exten => 25,1,answer exten => 25,n,dial(sip/25|30|kw) exten => 25,n,voicemail([EMAIL PROTECTED]|u) exten => 26,hint(SIP/26) exten => 26,1,answer exten => 26,n,dial(sip/26|30|kw) exten => 26,n,voicemail([EMAIL PROTECTED]|u) exten => 29,hint(SIP/29) exten => 29,1,answer exten => 29,n,dial(sip/29|30|kw) exten => 29,n,voicemail([EMAIL PROTECTED]|u) exten => 30,hint(SIP/30) exten => 30,1,answer exten => 30,n,dial(sip/30|30|kw) exten => 30,n,voicemail([EMAIL PROTECTED]|u) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [asterisk-users] cmd page crashesAsteriskSVN-branch-1.4-r57207
I take it back. It will not work if you hang up the calling phone first.
Still crashes
-- Executing [EMAIL PROTECTED]:1] SIPAddHeader("SIP/36651-b7d1cf48",
"Call-Info: answer-after=0") in new stack
-- Executing [EMAIL PROTECTED]:2] Page("SIP/36651-b7d1cf48",
"SIP/36651&SIP/36652&sip36655&sip/36653&sip/36651h|d") in new stack
-- Called 36652
[Mar 12 19:25:27] WARNING[7784]: app_page.c:129 page_exec: Incomplete
destination 'sip36655' supplied.
-- Called 36653
-- Called 36651h
-- Playing 'beep' (language 'en')
-- SIP/36653-09f68f78 is ringing
-- SIP/36652-09f679f8 is ringing
-- SIP/36651h-09f7fbd8 is ringing
-- SIP/36652-09f679f8 answered
-- Created MeetMe conference 1023 for conference '1689628562d'
-- SIP/36651h-09f7fbd8 answered
-- SIP/36653-09f68f78 answered
[Mar 12 19:25:31] WARNING[7784]: channel.c:1686 ast_hangup: Hard hangup
called by thread 21883824 on SIP/36652-09f679f8, while fd is blocked by
thread 49327024 in procedure ast_waitfor_nandfds! Expect a failure
[Mar 12 19:25:31] WARNING[7784]: channel.c:1686 ast_hangup: Hard hangup
called by thread 21883824 on SIP/36653-09f68f78, while fd is blocked by
thread 116542384 in procedure ast_waitfor_nandfds! Expect a failure
[Mar 12 19:25:31] WARNING[7784]: channel.c:1686 ast_hangup: Hard hangup
called by thread 21883824 on SIP/36651h-09f7fbd8, while fd is blocked by
thread 95366064 in procedure ast_waitfor_nandfds! Expect a failure
== Spawn extension (amaxx, **2, 2) exited non-zero on
'SIP/36651-b7d1cf48'
VoIP-PBX*CLI>
Disconnected from Asterisk server
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Monday, March 12, 2007 7:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: Spam? Re: [asterisk-users] cmd page
crashesAsteriskSVN-branch-1.4-r57207
Just wanted to update the group
I updated asterisk to SVN-branch-1.4-r58833M and page no longer crashes
Asterisk. My below example works great.
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Friday, March 02, 2007 3:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: Spam? Re: [asterisk-users] cmd page crashes
AsteriskSVN-branch-1.4-r57207
Did that. No change
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee
Jenkins
Sent: Friday, March 02, 2007 10:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [asterisk-users] cmd page crashes Asterisk
SVN-branch-1.4-r57207
Hall, Eric M. wrote:
> Group
>
> I'm having some trouble with asterisk and the page cmd.
> Any help would be great!
>
> This is what's in my extensions.conf
>
> exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0)
>
> exten => _**2,2,Page(SIP/36651)|d
>
> exten => _**2,3,Hangup
>
Looks like you have at least a syntax error. You have:
_**2,2,Page(SIP/36651)|d
And it should be
_**2,2,Page(SIP/36651|d)
Try fixing the "d" option by placing it within the right parenthesis and
try it again.
--
Warm Regards,
Lee
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___
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RE: Spam? Re: [asterisk-users] cmd page crashes AsteriskSVN-branch-1.4-r57207
Just wanted to update the group I updated asterisk to SVN-branch-1.4-r58833M and page no longer crashes Asterisk. My below example works great. Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Friday, March 02, 2007 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [asterisk-users] cmd page crashes AsteriskSVN-branch-1.4-r57207 Did that. No change -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Friday, March 02, 2007 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207 Hall, Eric M. wrote: > Group > > I'm having some trouble with asterisk and the page cmd. > Any help would be great! > > This is what's in my extensions.conf > > exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0) > > exten => _**2,2,Page(SIP/36651)|d > > exten => _**2,3,Hangup > Looks like you have at least a syntax error. You have: _**2,2,Page(SIP/36651)|d And it should be _**2,2,Page(SIP/36651|d) Try fixing the "d" option by placing it within the right parenthesis and try it again. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk queue and agents
I use AgentCallBackLogin I called that exten from my cell. However I have tested it calling into the Queue with the same outcome. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Thursday, March 08, 2007 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents Ok. One more thing - how are you logging the agent in? With AgentLogin or AgentCallBackLogin? Additionally, how did you get on that call 56405 to your cell? Was it directly to the SIP device or via the agent channel that the represents that SIP device? BJ On 3/8/07, Hall, Eric M. <[EMAIL PROTECTED]> wrote: > Sorry > Forgot to tell you I was on exten 56405 called to my cell. I then called > into the Queue with another cell and this is the output. > > Also forgot to include the show queue > > voipgw*CLI> show queue > dayton has 0 calls (max unlimited) in 'rrmemory' strategy (0s > holdtime), W:0, C:0, A:0, SL:0.0% within 0s >Members: > agent/56432 (Unavailable) has taken no calls yet > agent/56422 (Unavailable) has taken no calls yet > agent/56426 (Unavailable) has taken no calls yet > agent/56424 (Unavailable) has taken no calls yet > agent/56429 (Unavailable) has taken no calls yet > agent/56427 (Unavailable) has taken no calls yet > agent/56425 (Unavailable) has taken no calls yet > agent/56411 (Unavailable) has taken no calls yet > agent/56428 (Unavailable) has taken no calls yet >No Callers > > masion has 1 calls (max unlimited) in 'fewestcalls' strategy (0s > holdtime), W:0, C:0, A:2, SL:0.0% within 0s >Members: > agent/564321 (Unavailable) has taken no calls yet > agent/564221 (Unavailable) has taken no calls yet > agent/56405 (paused) (Not in use) has taken no calls yet > agent/56423 (Unavailable) has taken no calls yet > agent/56421 (paused) (Not in use) has taken no calls yet > agent/56420 (Unavailable) has taken no calls yet > agent/56416 (paused) (Not in use) has taken no calls yet >Callers: > 1. SIP/208.70.216.73-09780030 (wait: 0:12, prio: 0) > > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. > Sent: Thursday, March 08, 2007 7:24 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Asterisk queue and agents > > Asterisk SVN-branch-1.4-r58243 > > Voipgw*CLI> show agents > 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is > 'default') > 56420(Ran Dodds) not logged in (musiconhold is 'default') > 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold > is 'default') > 56423(Manager) not logged in (musiconhold is 'default') > 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') > 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') > 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') > 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') > 56426(HEATHER PRICE) not logged in (musiconhold is 'default') > 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') > 56429(JOE FERRAU) not logged in (musiconhold is 'default') > 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') > 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') > 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') > 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') > 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is > 'default') > 16 agents configured [3 online , 13 offline] > > voipgw*CLI> show agents > 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is > 'default') > 56420(Ran Dodds) not logged in (musiconhold is 'default') > 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold > is 'default') > 56423(Manager) not logged in (musiconhold is 'default') > 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') > 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') > 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') > 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is
RE: [asterisk-users] Asterisk queue and agents
Sorry Forgot to tell you I was on exten 56405 called to my cell. I then called into the Queue with another cell and this is the output. Also forgot to include the show queue voipgw*CLI> show queue dayton has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: agent/56432 (Unavailable) has taken no calls yet agent/56422 (Unavailable) has taken no calls yet agent/56426 (Unavailable) has taken no calls yet agent/56424 (Unavailable) has taken no calls yet agent/56429 (Unavailable) has taken no calls yet agent/56427 (Unavailable) has taken no calls yet agent/56425 (Unavailable) has taken no calls yet agent/56411 (Unavailable) has taken no calls yet agent/56428 (Unavailable) has taken no calls yet No Callers masion has 1 calls (max unlimited) in 'fewestcalls' strategy (0s holdtime), W:0, C:0, A:2, SL:0.0% within 0s Members: agent/564321 (Unavailable) has taken no calls yet agent/564221 (Unavailable) has taken no calls yet agent/56405 (paused) (Not in use) has taken no calls yet agent/56423 (Unavailable) has taken no calls yet agent/56421 (paused) (Not in use) has taken no calls yet agent/56420 (Unavailable) has taken no calls yet agent/56416 (paused) (Not in use) has taken no calls yet Callers: 1. SIP/208.70.216.73-09780030 (wait: 0:12, prio: 0) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Thursday, March 08, 2007 7:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk queue and agents Asterisk SVN-branch-1.4-r58243 Voipgw*CLI> show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] voipgw*CLI> show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] If you tell me how to do a "full" DEBUG/VERBOSE I will be happy to send you one. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Thursday, March 08, 2007 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents
RE: [asterisk-users] Asterisk queue and agents
Asterisk SVN-branch-1.4-r58243 Voipgw*CLI> show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] voipgw*CLI> show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] If you tell me how to do a "full" DEBUG/VERBOSE I will be happy to send you one. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Thursday, March 08, 2007 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents What version of Asterisk is this "the r number" on the 1.4 branch? I'll try and reproduce the condition here. Also - if you could post into that bug on Mantis a "full" DEBUG/VERBOSE log and what it looks like when you do "show queues" when one of these agents is on the phone, that'd be real helpful. Thanks. On 3/7/07, Hall, Eric M. <[EMAIL PROTECTED]> wrote: > BJ > Here is the sip.conf file. Hints work great. The only problem is the queue > is sending calls to an agent that's on the phone. > > > [general] > rtcachefriends=yes > videosupport=yes > port=5060 ; Port to bind to (SIP is 5060) > bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) > context=sip ; Send unknown SIP callers to this context > allow=g729 > allow=h263 ; H.263 is our video codec > allow=h263p ; H.263p is the enhanced video codec > ;allow=g711 > ;allow=all > ;allow=ulaw > ;allow=gsm > nat=1 > host=dynamic > type=peer > qualify=yes > notifyringing=yes > Subscribecontext=sip > call-limit=300 > notifyhold = yes > limitonpeer = yes > notifyringing = yes; Notify subscriptions on RINGING state > (default: no) > notifyhold = yes > > > [56405] ;Eric Test > type=friend ; "friend" means this device takes and makes calls > username=1 ; Username on device > callerid=Eric Test Phone <56405> > secret=56405; Password for device > host=dynamic ; This host is not on the same IP addr every time > context=sip ; Inbound calls from this host go here > [EMAIL
RE: [asterisk-users] Asterisk queue and agents
BJ Here is the sip.conf file. Hints work great. The only problem is the queue is sending calls to an agent that's on the phone. [general] rtcachefriends=yes videosupport=yes port=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) context=sip ; Send unknown SIP callers to this context allow=g729 allow=h263 ; H.263 is our video codec allow=h263p ; H.263p is the enhanced video codec ;allow=g711 ;allow=all ;allow=ulaw ;allow=gsm nat=1 host=dynamic type=peer qualify=yes notifyringing=yes Subscribecontext=sip call-limit=300 notifyhold = yes limitonpeer = yes notifyringing = yes; Notify subscriptions on RINGING state (default: no) notifyhold = yes [56405] ;Eric Test type=friend ; "friend" means this device takes and makes calls username=1 ; Username on device callerid=Eric Test Phone <56405> secret=56405; Password for device host=dynamic ; This host is not on the same IP addr every time context=sip ; Inbound calls from this host go here [EMAIL PROTECTED]; Activate the message waiting light if this canreinvite=no; Leave this alone for now; see archives for details nat=1 qualify=yes Subscribecontext=sip notifyringing=yes call-limit=300 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, March 07, 2007 10:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents I don't think this is a bug. From UPGRADE.txt: * Queues depend on the channel driver reporting the proper state for each member of the queue. To get proper signalling on queue members that use the SIP channel driver, you need to enable a call limit (could be set to a high value so it is not put into action) and also make sure that both inbound and outbound calls are accounted for. Example: [general] limitonpeer = yes [peername] type=friend call-limit=10 Please test with that and report your findings, and if it's still not working find us on IRC as we'd like to take a further look and see what might be wrong. BJ On 3/7/07, Hall, Eric M. <[EMAIL PROTECTED]> wrote: > Looks like it's a bug > > http://bugs.digium.com/view.php?id=9172&nbn=3 > > I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and > report back to the list. > > > > Eric Hall > > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz > (Ta^3) > Sent: Wednesday, March 07, 2007 1:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk queue and agents > > >Have a question for the group > >If I have an agent is on the phone outside of the queue should that > > person > >still get queue calls ? > >Doing a show agents online I see Available however show hints I see > > inuse. > > There is a ringinuse feature for SIP devices on 1.4.X which is what you are > looking for. > > -- > Octavio Ruiz Cervera > Neocenter, SA. de CV. > http://www.neocenter.com/ > Soluciones para Centros de Contacto y Telefonía IP > Tel.: (+52 55) 8590-9000 Ext. 9016 > Cel.: (+55 55) 5514-087790 > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk queue and agents
Looks like it's a bug http://bugs.digium.com/view.php?id=9172&nbn=3 I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and report back to the list. Eric Hall -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz (Ta^3) Sent: Wednesday, March 07, 2007 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents >Have a question for the group >If I have an agent is on the phone outside of the queue should that person >still get queue calls ? >Doing a show agents online I see Available however show hints I see inuse. There is a ringinuse feature for SIP devices on 1.4.X which is what you are looking for. -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] auto dialer
OK now I fell like a a$$... Thanks for that kick in the butt !! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Melcon Moraes Sent: Wednesday, March 07, 2007 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] auto dialer WaitTime stands for how long to wait until the call is considered "NO ANSWERED" Who can pickup a phone in 2 seconds, if not a robot? Try switch values between Retrytime and WaitTime. []'s MM -Original Message- From: "Hall, Eric M." <[EMAIL PROTECTED]> To: Cc: Sent: Wed, 7 Mar 2007 15:53:23 -0500 Delivered: Wed, 07 Mar 2007 17:45:35 Subject:[asterisk-users] auto dialer Not able to get the auto dialer part of asterisk to workwith the zap channel. It works great with the sip channel. Here is the callfile and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currentlyrunning on VoIP-PBX (pid = 8002) Verbosity is at least 3 -- Attempting call on ZAP/G1/6144994925 [EMAIL PROTECTED]:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- Hungup 'Zap/23-1' [Mar 7 15:46:29] NOTICE[10159]: pbx_spool.c:341attempt_thread: Call failed to go through, reason 0 VoIP-PBX*CLI> E-mail classificado pelo Identificador de Spam Inteligente. Para alterar a categoria classificada, visite o <http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l= 1,1173300915.746475.15282.aldavila.hst.terra.com.br,8031,Des15,Des15>Ter ra Mail --Original Message Ends-- -- Melcon Moraes <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] auto dialer
Not able to get the auto dialer part of asterisk to work with the zap channel. It works great with the sip channel. Here is the call file and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid = 8002) Verbosity is at least 3 -- Attempting call on ZAP/G1/6144994925 for [EMAIL PROTECTED]:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- Hungup 'Zap/23-1' [Mar 7 15:46:29] NOTICE[10159]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason 0 VoIP-PBX*CLI> ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk queue and agents
I think that is already set. Here is my queue.conf [general] persistentmembers = yes autofill = yes monitor-type = MixMonitor [support] musicclass = default strategy = fewestcalls timeout = 10 retry = 5 autofill=yes autopause=yes setinterfacevar=no announce-frequency = 90 periodic-announce-frequency=60 announce-holdtime = yes announce-round-seconds = 10 queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-holdtime = queue-holdtime queue-minutes = queue-minutes queue-seconds = queue-seconds queue-thankyou = queue-thankyou queue-lessthan = queue-less-than queue-reporthold = queue-reporthold ;periodic-announce = queue-periodic-announce joinempty = yes leavewhenempty = no eventwhencalled = vars QueueMemberStatus=yes eventmemberstatus = yes reportholdtime = no ringinuse = no memberdelay = 1 member => agent/56416 member => agent/56420 member => agent/56421 member => agent/56423 member => agent/56405 member => agent/564221 member => agent/564321 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz (Ta^3) Sent: Wednesday, March 07, 2007 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents >Have a question for the group >If I have an agent is on the phone outside of the queue should that person >still get queue calls ? >Doing a show agents online I see Available however show hints I see inuse. There is a ringinuse feature for SIP devices on 1.4.X which is what you are looking for. -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk queue and agents
Have a question for the group If I have an agent is on the phone outside of the queue should that person still get queue calls ? Doing a show agents online I see Available however show hints I see inuse. Any ideas Eric Hall Vice-president Amaxx, Inc. "Customized IT Solutions" 5925B Wilcox Place Dublin OH 43016 614.923.6652 - Direct 614.486.3481 - Office 614.923.6652 - eFax ___ The information contained in this message and any attachment may be proprietary, confidential, and privileged or subject to the work product doctrine and thus protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail question
Group In voicemail.conf I would like to having the following setup per context not per-mailbox settings serveremail userscontext fromstring usedirectory emailbody pagerfromstring dialout sendvoicemail callback review operator volgain nextaftercmd forcename forcegreetings tempgreetwarn Can this be done? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dial question
D
Not sure why this works
exten => _3665[0-9],1,goto(test|${EXTEN}|1)
but this does not.
exten => _366[50-59],1,goto(test|${EXTEN}|1)
I would like to route 36650 - 36700 to a Context 'test' however I'm only
able to get 10 to work at a time. Any ideas?
Any help would be great!
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RE: Spam? Re: [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207
Did that. No change -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Friday, March 02, 2007 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207 Hall, Eric M. wrote: > Group > > I'm having some trouble with asterisk and the page cmd. > Any help would be great! > > This is what's in my extensions.conf > > exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0) > > exten => _**2,2,Page(SIP/36651)|d > > exten => _**2,3,Hangup > Looks like you have at least a syntax error. You have: _**2,2,Page(SIP/36651)|d And it should be _**2,2,Page(SIP/36651|d) Try fixing the "d" option by placing it within the right parenthesis and try it again. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207
Group
I'm having some trouble with asterisk and the page cmd.
Any help would be great!
This is what's in my extensions.conf
exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0)
exten => _**2,2,Page(SIP/36651)|d
exten => _**2,3,Hangup
CLI output
Connected to Asterisk SVN-branch-1.4-r57207 currently running on
VoIP-PBX (pid = 11317)
-- Remote UNIX connection
Verbosity is at least 3
Extension Changed 36652 new state InUse for Notify User 36653
-- Executing [EMAIL PROTECTED]:1] SIPAddHeader("SIP/36652-b7d0c1f0",
"Call-Info: answer-after=0") in new stack
-- Executing [EMAIL PROTECTED]:2] Page("SIP/36652-b7d0c1f0", "SIP/36651")
in new stack
-- Called 36651
-- Playing 'beep' (language 'en')
Extension Changed 36651 new state Ringing for Notify User 36653
-- SIP/36651-09eb3648 is ringing
-- SIP/36651-09eb3648 answered
Extension Changed 36651 new state InUse for Notify User 36653
-- Created MeetMe conference 1023 for conference '10382980d'
[Mar 2 09:14:58] WARNING[11449]: channel.c:1686 ast_hangup: Hard hangup
called by thread 29141936 on SIP/36651-09eb3648, while fd is blocked by
thread 20036528 in procedure ast_waitfor_nandfds! Expect a failure
== Spawn extension (amaxx, **2, 2) exited non-zero on
'SIP/36652-b7d0c1f0'
Extension Changed 36651 new state Idle for Notify User 36653
Extension Changed 36652 new state Idle for Notify User 36653
VoIP-PBX*CLI>
Disconnected from Asterisk server
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RE: Spam? Re: [asterisk-users] Asterisk and IM
Has anyone got Asterisk IM to work Using this link http://www.sipalive.com/dev/asterisk/ And a clean install of asteris 1.4.0-Beta3 I get the following error Any ideas? I have no idea what the .rej file is telling me so it maybe easy to see it here but I'm a little out of my strike zone her! patch -p0 --- 90,99 #include "asterisk.h" + /* Include this for message queuing support. Comment out if not wanted. + * You will need to link with sqlite */ + /* #include "queue_chan_sip.h" + ASTERISK_FILE_VERSION(__FILE__, "$Revision: 48487 $") #include -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kenneth Padgett Sent: Friday, January 05, 2007 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [asterisk-users] Asterisk and IM > I have been asked to get IM via the X-Ten softphone to work with Asterisk. > Anyone have any ideas? I have looked on google and other places with > no luck. > > Our system is as followed > > Linux CentOS 4.4 > Asterisk 1.4.0-beta3 > X-Lite v3.0 for Windows If by IM, you mean the built-in Jabber stuff in v1.4... I am having trouble with that and CentOS 4.4 myself, can't get the required libs or some such non-sense. -Kenneth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [asterisk-users] Asterisk and IM
Kenneth Thanks for the reply. What I'm looking to do is listed here http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging However the patch does not work on the system listed below. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kenneth Padgett Sent: Friday, January 05, 2007 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [asterisk-users] Asterisk and IM > I have been asked to get IM via the X-Ten softphone to work with Asterisk. > Anyone have any ideas? I have looked on google and other places with > no luck. > > Our system is as followed > > Linux CentOS 4.4 > Asterisk 1.4.0-beta3 > X-Lite v3.0 for Windows If by IM, you mean the built-in Jabber stuff in v1.4... I am having trouble with that and CentOS 4.4 myself, can't get the required libs or some such non-sense. -Kenneth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and IM
Hello group I have been asked to get IM via the X-Ten softphone to work with Asterisk. Anyone have any ideas? I have looked on google and other places with no luck. Our system is as followed Linux CentOS 4.4 Asterisk 1.4.0-beta3 X-Lite v3.0 for Windows Thanks! Eric Hall ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] App_Swift
Group
I have app_swift working on our asterisk server running 1.4-Beta3.
My question is can you read variables with it? Like reading back
callerid number ${CALLERID(number)
Eric Hall
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RE: [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto
Fixed my problem! Note to self... READ EVERYTHING in the instructions! Again thanks for the information! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Thursday, November 30, 2006 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto Great link. After I all you said I get this error loading the module in asterisk via load app_swift The 'load' command is deprecated and will be removed in a future release. Please use 'module load' instead. [Nov 30 13:54:08] WARNING[7825]: loader.c:362 load_dynamic_module: Error loading module 'app_swift': libswift.so.4: cannot open shared object file: No such file or directory [Nov 30 13:54:08] WARNING[7825]: loader.c:607 load_resource: Module 'app_swift' could not be loaded. Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: Thursday, November 30, 2006 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto Hi All, Recent discussions on app_cepstral on the list have led me to believe there's some issues with Asterisk 1.4 I set about creating a small howto for people to get cepstral, with app_swift working. Check it out: http://www.voipphreak.ca/archives/354-Asterisk-1.4-Gentoo-CepstralSwift- Howto-using-App_Swift.html Thanks, Diwelf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto
Great link. After I all you said I get this error loading the module in asterisk via load app_swift The 'load' command is deprecated and will be removed in a future release. Please use 'module load' instead. [Nov 30 13:54:08] WARNING[7825]: loader.c:362 load_dynamic_module: Error loading module 'app_swift': libswift.so.4: cannot open shared object file: No such file or directory [Nov 30 13:54:08] WARNING[7825]: loader.c:607 load_resource: Module 'app_swift' could not be loaded. Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: Thursday, November 30, 2006 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.4 : App_Swift (Cepstral) Howto Hi All, Recent discussions on app_cepstral on the list have led me to believe there's some issues with Asterisk 1.4 I set about creating a small howto for people to get cepstral, with app_swift working. Check it out: http://www.voipphreak.ca/archives/354-Asterisk-1.4-Gentoo-CepstralSwift- Howto-using-App_Swift.html Thanks, Diwelf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [asterisk-users] Getting app_cepstral to work withAsterisk 1.4.0-beta3
I get an error when I do a make install [EMAIL PROTECTED] app_swift-0.9.5]# make install gcc -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC -DCHANNEL_HAS_CID -DNEW_CONFIG -I/opt/swift/include -c -o app_swift.o app_swift.c app_swift.c:49: warning: type defaults to `int' in declaration of `STANDARD_LOCAL_USER' app_swift.c:49: warning: data definition has no type or storage class app_swift.c:50: warning: type defaults to `int' in declaration of `LOCAL_USER_DECL' app_swift.c:50: warning: data definition has no type or storage class app_swift.c: In function `swift_exec': app_swift.c:158: warning: implicit declaration of function `LOCAL_USER_ADD' app_swift.c:162: warning: assignment discards qualifiers from pointer target type app_swift.c:275: warning: implicit declaration of function `LOCAL_USER_REMOVE' app_swift.c: In function `unload_module': app_swift.c:305: error: `STANDARD_HANGUP_LOCALUSERS' undeclared (first use in this function) app_swift.c:305: error: (Each undeclared identifier is reported only once app_swift.c:305: error: for each function it appears in.) app_swift.c: In function `usecount': app_swift.c:327: warning: implicit declaration of function `STANDARD_USECOUNT' make: *** [app_swift.o] Error 1 [EMAIL PROTECTED] app_swift-0.9.5]# Eric Hall Vice-president Amaxx, Inc. "Customized IT Solutions" 5925B Wilcox Place Dublin OH 43016 614.923.6652 - Direct 614.486.3481 - Office 614.923.6652 - eFax Try our off site backup service free for 30 days. http://www.nationalbackup.com/> http://www.nationalbackup.com http://www.nationalbackup.com/> ___ The information contained in this message and any attachment may be proprietary, confidential, and privileged or subject to the work product doctrine and thus protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Earle Clubb Sent: Wednesday, November 29, 2006 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [asterisk-users] Getting app_cepstral to work withAsterisk 1.4.0-beta3 Hall, Eric M. wrote: Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks Eric, I had similar compilation issues when trying to use app_cepstral. This doesn't answer your question, but I've had good success using app_swift. http://www.loopfree.net/app_swift/ Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3
Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks [CC] app_cepstral.c -> app_cepstral.o In file included from /usr/src/asterisk/include/asterisk/linkedlists.h:23, from /usr/src/asterisk/include/asterisk/frame.h:37, from /usr/src/asterisk/include/asterisk/channel.h:110, from app_cepstral.c:33: /usr/src/asterisk/include/asterisk/lock.h: In function `ast_mutex_init': /usr/src/asterisk/include/asterisk/lock.h:513: warning: implicit declaration of function `pthread_mutexattr_settype' /usr/src/asterisk/include/asterisk/lock.h:513: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/src/asterisk/include/asterisk/lock.h:513: error: (Each undeclared identifier is reported only once /usr/src/asterisk/include/asterisk/lock.h:513: error: for each function it appears in.) In file included from /usr/src/asterisk/include/asterisk/cdr.h:48, from /usr/src/asterisk/include/asterisk/channel.h:115, from app_cepstral.c:33: /usr/src/asterisk/include/asterisk/utils.h: In function `_ast_strndup': /usr/src/asterisk/include/asterisk/utils.h:421: warning: implicit declaration of function `strndup' /usr/src/asterisk/include/asterisk/utils.h:421: warning: assignment makes pointer from integer without a cast /usr/src/asterisk/include/asterisk/utils.h: In function `_ast_asprintf': /usr/src/asterisk/include/asterisk/utils.h:446: warning: implicit declaration of function `vasprintf' In file included from app_cepstral.c:36: /opt/swift/include/swift.h: At top level: /opt/swift/include/swift.h:765: warning: function declaration isn't a prototype app_cepstral.c:43: warning: type defaults to `int' in declaration of `STANDARD_LOCAL_USER' app_cepstral.c:43: warning: data definition has no type or storage class app_cepstral.c:44: warning: type defaults to `int' in declaration of `LOCAL_USER_DECL' app_cepstral.c:44: warning: data definition has no type or storage class app_cepstral.c: In function `cepstral_exec': app_cepstral.c:225: warning: implicit declaration of function `LOCAL_USER_ADD' app_cepstral.c:233: warning: implicit declaration of function `LOCAL_USER_REMOVE' app_cepstral.c: At top level: app_cepstral.c:252: warning: no previous prototype for 'unload_module' app_cepstral.c: In function `unload_module': app_cepstral.c:253: error: `STANDARD_HANGUP_LOCALUSERS' undeclared (first use in this function) app_cepstral.c: At top level: app_cepstral.c:258: warning: no previous prototype for 'load_module' app_cepstral.c:263: warning: no previous prototype for 'description' app_cepstral.c:268: warning: no previous prototype for 'usecount' app_cepstral.c: In function `usecount': app_cepstral.c:270: warning: implicit declaration of function `STANDARD_USECOUNT' app_cepstral.c: At top level: app_cepstral.c:305: warning: function declaration isn't a prototype make[1]: *** [app_cepstral.o] Error 1 make: *** [apps] Error 2 Eric Hall Vice-president Amaxx, Inc. "Customized IT Solutions" 5925B Wilcox Place Dublin OH 43016 614.923.6652 - Direct 614.486.3481 - Office 614.923.6652 - eFax Try our off site backup service free for 30 days. http://www.nationalbackup.com/> http://www.nationalbackup.com http://www.nationalbackup.com/> ___ The information contained in this message and any attachment may be proprietary, confidential, and privileged or subject to the work product doctrine and thus protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best text to speech program
I'm looking to set up asterisk to call customer 3 days before the app and remind them we will be out to see them. I'm looking for any ideas on good ways to do this. Also I think it would be best to do some type of text to speech however I do not like the sound of the free one . Any ideas? Thanks!!! Eric Hall ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] whisper paging
Does anyone have a quick howto and a sample to get whisper paging to work? I'm running sterisk Asterisk 1.4.0-beta2 Thanks for your help! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Set hint status from dialplan?
Here is an output from a 1.4.0-Beta2 voipgw*CLI> show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- Agent Call Agent Proxy Channel yes yes no Console OSS Console Channel Driver no yes no Zap Zapata Telephony Driver w/PRIno yes no Skinny Skinny Client Control Protocol (Skinny) no yes no Phone Standard Linux Telephony API Driver no yes no Feature Feature Proxy Channel Driver no yes no SIP Session Initiation Protocol (SIP)yes yes yes Local Local Proxy Channel Driver yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes MGCPMedia Gateway Control Protocol (MGCP)yes yes no -- 10 channel drivers registered. voipgw*CLI> show version Asterisk 1.4.0-beta2 built by root @ voipgw on a i686 running Linux on 2006-09-25 00:49:44 UTC voipgw*CLI> -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, September 26, 2006 2:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: Set hint status from dialplan? On Tuesday 26 September 2006 13:57, C F wrote: > Andrew what does "show channeltypes" give you? *CLI> show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- Zap Zapata Telephony Driver w/PRIno yes no SIP Session Initiation Protocol (SIP)yes yes yes Local Local Proxy Channel Driver yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no -- 6 channel drivers registered. *CLI> show version Asterisk SVN-trunk-r41990 built by root @ asterisk on a i686 running Linux on 2006-09-12 03:02:05 UTC Curious... I see Local/ has a devicestate, and I've never heard of a "Feature/" channel type before... :-) So I imagine I could use Local/[EMAIL PROTECTED] to see parking slot state, but nothing for arbitrary channels such as what Lacy is showing. Is that correct? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? [asterisk-users] OT: Opinions on Aastra 480i CT?
I have this phone on my desk. It works very very well! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Monday, September 25, 2006 10:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Spam? [asterisk-users] OT: Opinions on Aastra 480i CT? Looks good, great price: http://www.aastratelecom.com/ipphones/pro_243.asp Anybody using these? How's the cordless? Does it play nice with * ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about SVN-trunk-r43322 and Asterisk Recording Interface
Group Any known problems with Asterisk SVN-trunk-r43322 and Asterisk Recording Interface or the vmail.cgi script? I'm unable to see voicemails via the web even though the MWI is flashing and if I look in /var/spool/asterisk/voicemail/default/100/INBOX I do see msg files in that folder. Have not built a system in a while so I must be rusty. Never had problems with install of asterisk and the ARI or vmail.cgi. Thanks again for all the help I have been given over that last few days. Its been a BIG time saver!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about SVN-trunk-r43322 and Asterisk Recording Interface
Group Any known problems with Asterisk SVN-trunk-r43322 and Asterisk Recording Interface or the vmail.cgi script? I'm unable to see voicemails via the web even though the MWI is flashing and if I look in /var/spool/asterisk/voicemail/default/100/INBOX I do see msg files in that folder. Have not built a system in a while so I must be rusty. Never had problems with install of asterisk and the ARI or vmail.cgi. Thanks again for all the help I have been given over that last few days. Its been a BIG time saver!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HINT problems with SVN-trunk-r43322
Brad Thanks for your insight. The info I used to set this up before was from Grandstream http://www.grandstream.com/FAQ/FAQ_and_Example_for_Asterisk_Configuratio n_for_GXP-2000.pdf I will also notify them about the error in the above document. Thanks again!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Watkins, Bradley Sent: Thursday, September 21, 2006 6:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] HINT problems with SVN-trunk-r43322 The reason is that, at least in the SIP channel in trunk, the structure that keeps track of device state for hinting only gets allocated on peer objects and then only if call-limit is configured to some value. It's been a long time since I've done any development with 1.2 (all my 1.2 systems are waiting for 1.4 to come out so we can add a bunch of features), so I forget how that works there. Rumor has it these restrictions aren't necessary, but I forget. If by '6 months' you mean trunk from that long ago, it's entirely plausible that you got a snapshot during the evolution from where it was in 1.2 to where it is today. Regards, - Brad > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Hall, > Eric M. > Sent: Wednesday, September 20, 2006 10:07 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] HINT problems with SVN-trunk-r43322 > > Group > Looks like the > > type=peer > call-limit=2 > > Works. Now the question is why? The sample I sent is working on a > system build 6 months ago. > Will do some more checking and will report to the list on anything I > find... > > Thanks Bradley for this bit of info you gave!! > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Andrew > Kohlsmith > Sent: Wednesday, September 20, 2006 1:36 PM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] HINT problems with SVN-trunk-r43322 > > On Wednesday 20 September 2006 12:31, Watkins, Bradley wrote: > > You will need to change the type=friend to type=peer and > also define > > call-limit to some value (it can be large if you don't care > about the > > actual limit). That should fix hints for you. > > But if you have it set to >1 then busy status won't work, isn't that > the case? > > -A. > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HINT problems with SVN-trunk-r43322
Group Looks like the type=peer call-limit=2 Works. Now the question is why? The sample I sent is working on a system build 6 months ago. Will do some more checking and will report to the list on anything I find... Thanks Bradley for this bit of info you gave!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, September 20, 2006 1:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] HINT problems with SVN-trunk-r43322 On Wednesday 20 September 2006 12:31, Watkins, Bradley wrote: > You will need to change the type=friend to type=peer and also define > call-limit to some value (it can be large if you don't care about the > actual limit). That should fix hints for you. But if you have it set to >1 then busy status won't work, isn't that the case? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HINT problems with SVN-trunk-r43322
Just found out this may only been a sip problem. State work with zap and SCCP when checking status via cli -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Wednesday, September 20, 2006 12:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HINT problems with SVN-trunk-r43322 On Wed, 2006-09-20 at 11:39 -0400, Hall, Eric M. wrote: > I’m unable to get HINTS working with the new SVN-Trunk > > State never changed when ringing or on the phone. Confirmed here, I only noticed because of this message. -- Dave Cotton <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HINT problems with SVN-trunk-r43322
I’m unable to get HINTS working with the new SVN-Trunk State never changed when ringing or on the phone. Below is my configs (Maybe I missed something) Thanks for any help you could give!! ##sip.conf## [general] callerid=unavailable context=default ; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) ;allow=all allow=ulaw allow=g729 ;allow=gsm ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY videosupport=yes allow=h263 ; H.263 is our video codec allow=h263p ; H.263p is the enhanced video codec qualify=yes notifyringing=yes [101] type=friend ; "friend" means this device takes and makes calls username=101 ; Username on device callerid=Eric <102> secret=101 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=default ; Inbound calls from this host go here [EMAIL PROTECTED]; Activate the message waiting light if this canreinvite=no ; Leave this alone for now; see archives for details nat=1 qualify=yes Subscribecontext=default notifyringing=yes ##extensions.conf## [general] static=yes writeprotect=no autofallthrough=yes priorityjumping=yes [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2 [default] exten => 101,hint,SIP/101 exten => 102,hint,SIP/102 exten => 101,1,dial(sip/101,20,tw) exten => 101,n,voicemail(101) exten => 101,n,hanup() exten => 102,1,dial(sip/102,20,tw) exten => 102,n,voicemail(102) exten => 102,n,hanup() Commands from the CLI CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 102/102 206.173.108.30 D N 5060 OK (5 ms) 101/101 206.173.108.25 D N 5060 OK (5 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] CLI> show hints -= Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED] : SIP/102 State:Idle Watchers 1 [EMAIL PROTECTED] : SIP/101 State:Idle Watchers 1 - 2 hints registered CLI> sip show subscriptions Peer User Call ID Extension Last state Type Mailbox 206.173.108.30 102 fb84429adb2 [EMAIL PROTECTED] Idle dialog-info+xml 206.173.108.25 101 499798bcfa4 [EMAIL PROTECTED] Idle dialog-info+xml 2 active SIP subscriptions ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: FollowMe question
I got the config working. Not sure if someone has pre-recorded sounds for this app or not. Looked all over for them and I'm unable to locate them.If anyone has sound file they would like to share that would help me greatly. Thanks Sent: Friday, September 15, 2006 5:23 PMTo: 'asterisk-users@lists.digium.com'Subject: FollowMe question Group Does anyone have the FollowMe sound files? Do I need to record them? Also does anyone have a working followme.conf file that they would share? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FollowMe question
Group Does anyone have the FollowMe sound files? Do I need to record them? Also does anyone have a working followme.conf file that they would share? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto Dialer question
Hello group I have a customer that has asked me to build an auto dialer that will call customer a few day before an appt and remind them of the time and date of the appt. Does anyone have any good links for apps that could do this type of auto calling? They also request that information be pulled from a database and be able to pull reports on who was called and if they call was picked up. Thanks for any help the group could give me! Eric ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [Asterisk-Users] Cisco 7970 problems
I don't see that anywhere. Here is my zapata.conf This is only happing on my 7970 all other phone are working without trouble. [channels] context=pri signalling=pri_cpe switchtype=dms100 group=1 usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=no useincomingcalleridonzaptransfer=yes callerid=asreceived faxdetect=incoming musiconhold=default echocancel=yes echocancelwhenbridged=yes channel => 1-23 context=Fax switchtype=national signalling=pri_net group=2 overlapdial=yes usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=no useincomingcalleridonzaptransfer=yes callerid=asreceived faxdetect=no musiconhold=default channel => 25-47 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling Sent: Saturday, May 13, 2006 8:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Cisco 7970 problems Hall, Eric M. wrote: > I did not get this back from the list so I'm not sure if this hit the > list last week or not so I'm sending it again. Sorry if this is a > duplicate post! > > > -- > -- > --- > > > Has anyone had problems with a Cisco 7970 running sip image > SIP70.8.0-2SR1S hanging up zap channels? > > Calls to SIP and IAX are fine. Just when the call goes out via the zap > channels > > I have some Cisco 7960 running SIP and they work fine. A classic cause of this is callprogress=yes or busydetect=yes in zapata.conf -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 problems
I did not get this back from the list so I'm not sure if this hit the list last week or not so I'm sending it again. Sorry if this is a duplicate post! --- Has anyone had problems with a Cisco 7970 running sip image SIP70.8.0-2SR1S hanging up zap channels? Calls to SIP and IAX are fine. Just when the call goes out via the zap channels I have some Cisco 7960 running SIP and they work fine. Any ideas? Thanks-Eric Hall ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 problems
Has anyone had problems with a Cisco 7970 running sip image SIP70.8.0-2SR1S hanging up zap channels? Calls to SIP and IAX are fine. Just when the call goes out via the zap channels I have some Cisco 7960 running SIP and they work fine. Any ideas? Thanks-Eric Hall ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
Aaron Any idea how to change it from 24hr to 12hr ? Thanks again! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Friday, May 05, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question Aaron Yes it is very annoying! Thanks for the date time settings. That worked GREAT!!! Thanks - Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Friday, May 05, 2006 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. As for the date time settings... this is what we have in ours: CMLocal M/D/YA Central Standard/Daylight Time I'm guessing you should be able to change it to say Eastern instead of Central On Fri, 5 May 2006, Hall, Eric M. wrote: > > Group > I have a Cisco 7970 Running the newest SIP image. > I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC > > When I get a call the callerid number show something like > [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm > unable to find the correct wording when searching Google to find that > post again. Can anyone help me out here. How can I remove the asterisk > servers IP from the phone number? > > > Also I'm unable to get the time zone correct on the phone. It is in > UTC and I'm in EST I see in the file where it looks like it goes but > what I have tried has not worked as of yet. Here is what it looks like > > > M/D/Y > EST > > > > Thanks again for all your help!!! > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
Aaron Yes it is very annoying! Thanks for the date time settings. That worked GREAT!!! Thanks - Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Friday, May 05, 2006 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. As for the date time settings... this is what we have in ours: CMLocal M/D/YA Central Standard/Daylight Time I'm guessing you should be able to change it to say Eastern instead of Central On Fri, 5 May 2006, Hall, Eric M. wrote: > > Group > I have a Cisco 7970 Running the newest SIP image. > I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC > > When I get a call the callerid number show something like > [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm > unable to find the correct wording when searching Google to find that > post again. Can anyone help me out here. How can I remove the asterisk > servers IP from the phone number? > > > Also I'm unable to get the time zone correct on the phone. It is in > UTC and I'm in EST I see in the file where it looks like it goes but > what I have tried has not worked as of yet. Here is what it looks like > > > M/D/Y > EST > > > > Thanks again for all your help!!! > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 running SIP question
Group I have a Cisco 7970 Running the newest SIP image. I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC When I get a call the callerid number show something like [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm unable to find the correct wording when searching Google to find that post again. Can anyone help me out here. How can I remove the asterisk servers IP from the phone number? Also I'm unable to get the time zone correct on the phone. It is in UTC and I'm in EST I see in the file where it looks like it goes but what I have tried has not worked as of yet. Here is what it looks like M/D/Y EST Thanks again for all your help!!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID Name problem
Title: RE: [Asterisk-Users] CallerID Name problem That worked GREAT Thank you so so MUCH for your help!! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander LopezSent: Monday, May 01, 2006 5:06 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] CallerID Name problem You don't need the answer, But you need the wait. CallerID Name comes over the FACILITY messge many times and it takes a slpit second for it to come in. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.Sent: Monday, May 01, 2006 4:34 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] CallerID Name problem Do you wait before or after the answer? Do you even need the answer? -Original Message-From: Alexander Lopez [mailto:[EMAIL PROTECTED]]Sent: Mon May 01 14:26:49 2006To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] CallerID Name problemHow are the calls coming into the PBX. PRI? If so add a Wait(1) beforeyour try ringing the SIP channel.> -Original Message-> From: [EMAIL PROTECTED] [mailto:asterisk-users-> [EMAIL PROTECTED] On Behalf Of Hall, Eric M.> Sent: Monday, May 01, 2006 12:37 PM> To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: [Asterisk-Users] CallerID Name problem>>> I'm having trouble getting callerid name to show up on my phones(Cisco> 7960 and a few softphones)> When I look in the CDR database I see the name but not on any phonewhen> being called.>> I'm running> Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC>>> Any help would be great !> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [Asterisk-Users] CallerID Name problem
Title: RE: Spam? Re: [Asterisk-Users] CallerID Name problem I'm getting Number but when I look at the CDR database. I do see the name -Original Message- From: Lacy Moore - Aspendora [mailto:[EMAIL PROTECTED]] Sent: Mon May 01 17:10:26 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] CallerID Name problem Do you get caller ID number? If so, WAITing is not going to help, since you already get the info. If you get caller ID number, then your telco is not sending the name. On 5/1/06, Hall, Eric M. <[EMAIL PROTECTED]> wrote: > > Do you wait before or after the answer? Do you even need the answer? > > > > > -Original Message- > From: Alexander Lopez [mailto:[EMAIL PROTECTED]<[EMAIL PROTECTED]> > ] > Sent: Mon May 01 14:26:49 2006 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] CallerID Name problem > > How are the calls coming into the PBX. PRI? If so add a Wait(1) before > your try ringing the SIP channel. > > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Hall, Eric M. > > Sent: Monday, May 01, 2006 12:37 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] CallerID Name problem > > > > > > I'm having trouble getting callerid name to show up on my phones > (Cisco > > 7960 and a few softphones) > > When I look in the CDR database I see the name but not on any phone > when > > being called. > > > > I'm running > > Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC > > > > > > Any help would be great ! > > ___ > > --Bandwidth and Colocation provided by Easynews.com<http://easynews.com/>-- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com <http://easynews.com/>-- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation provided by Easynews.com <http://easynews.com/>-- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID Name problem
Title: RE: [Asterisk-Users] CallerID Name problem Thanks will try that tonight. Thanks again -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED]] Sent: Mon May 01 17:07:43 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] CallerID Name problem You don't need the answer, But you need the wait. CallerID Name comes over the FACILITY messge many times and it takes a slpit second for it to come in. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Hall, Eric M. Sent: Monday, May 01, 2006 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] CallerID Name problem Do you wait before or after the answer? Do you even need the answer? -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED]] Sent: Mon May 01 14:26:49 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] CallerID Name problem How are the calls coming into the PBX. PRI? If so add a Wait(1) before your try ringing the SIP channel. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Hall, Eric M. > Sent: Monday, May 01, 2006 12:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] CallerID Name problem > > > I'm having trouble getting callerid name to show up on my phones (Cisco > 7960 and a few softphones) > When I look in the CDR database I see the name but not on any phone when > being called. > > I'm running > Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC > > > Any help would be great ! > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID Name problem
Title: RE: [Asterisk-Users] CallerID Name problem Do you wait before or after the answer? Do you even need the answer? -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED]] Sent: Mon May 01 14:26:49 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] CallerID Name problem How are the calls coming into the PBX. PRI? If so add a Wait(1) before your try ringing the SIP channel. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Hall, Eric M. > Sent: Monday, May 01, 2006 12:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] CallerID Name problem > > > I'm having trouble getting callerid name to show up on my phones (Cisco > 7960 and a few softphones) > When I look in the CDR database I see the name but not on any phone when > being called. > > I'm running > Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC > > > Any help would be great ! > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID Name problem
Using SIP and SCCP. The softphone uses SIP. Doing a debug I see no name being sent. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kevin ling Sent: Monday, May 01, 2006 2:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] CallerID Name problem Hi, What protocol for your 7960 phone? SCCP or SIP? You can turn on the SIP debug on CLI to make sure the callerid and name pass to your phone. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Tuesday, May 02, 2006 12:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] CallerID Name problem I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great ! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID Name problem
I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great ! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based voicemail client
If your talking about Asterisk Recording Interface this is what I found on the web site Submitted by dan.littlejohn on Wed, 12/28/2005 - 5:34am. ARI does not support realtime yet. It is coming Nice app but just can't do what I need it to. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, March 26, 2006 9:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based voicemail client Ari? Thanks, Steve Totaro http://www.asteriskhelpdesk.com > -Original Message- > From: Hall, Eric M. [mailto:[EMAIL PROTECTED] > Sent: Sunday, March 26, 2006 9:06 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Web based voicemail client > > > I'm looking for a good web based voicemail client that can use mysql or > realtime drivers. I can't seem to get vmail.cgi to work with realtime. > > Thanks for any help you can give. > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web based voicemail client
I'm looking for a good web based voicemail client that can use mysql or realtime drivers. I can't seem to get vmail.cgi to work with realtime. Thanks for any help you can give. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [Asterisk-Users] Failed installing zaptel
Chuck, Thank You I'm also going to try CentOS 3 The problem is I have SATA HDD and running in to trouble getting Linux installed. Will update after I test Ver 3 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 13, 2006 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Failed installing zaptel Hi, I am having the same exact problem. I am assuming that it was a problem with a kernel update I did. I am in the process of rolling back to an older kernel... I will let you let know if this works. There is also a patch for zaptel but I believe this is for going from 1.3 to 1.4? Thanks Hall, Eric M. wrote: >Group > Having trouble installing zaptel. Below is my server specs > >Intel Motherboard D101GGC >TE405P >CentOS-4.2-i386 > > > >Here is the output trying to do a 'make' >=== > >make clean >rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw >ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver >sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo >rm -f *.ko *.mod.c .*o.cmd rm -rf .tmp_versions rm -f gendigits tones.h >rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f >ztcfg-shared fxstest [EMAIL PROTECTED] zaptel]# make >cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA >-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c >cc -o gendigits gendigits.o -lm >./gendigits >cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA >-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c -o makefw >./makefw tormenta2.rbt tor2fw > tor2fw.h Loaded 69900 bytes from file >./makefw pciradio.rbt radfw > radfw.h Loaded 42096 bytes from file >cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA >-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o ztcfg.o ztcfg.c >cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA >-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o zonedata.lo >zonedata.c >cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA >-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o tonezone.lo >tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg >ztcfg.o libtonezone.a -lm >cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA >-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o torisatool.o torisatool.c >cc -o torisatool torisatool.o >cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA >-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o ztmonitor.o ztmonitor.c >cc -o ztmonitor ztmonitor.o >cc -o ztspeed.o -c ztspeed.c >cc -o ztspeed ztspeed.o >cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA >-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o zttool.o zttool.c >cc -o zttool zttool.o -lnewt >cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA >-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c -o zttest >cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA >-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o fxotune.o fxotune.c >cc -o fxotune fxotune.o -lm >/lib/modules/2.6.9-34.ELsmp/build >make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel >modules >make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' > CC [M] /usr/src/zaptel/zaptel.o >/usr/src/zaptel/zaptel.c:372: error: syntax error before "zone_lock" >/usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in >declaration of `zone_lock' >/usr/src/zaptel/zaptel.c:372: error: incompatible types in >initialization >/usr/src/zaptel/zaptel.c:372: error: initializer element is not >constant >/usr/src/zaptel/zaptel.c:372: warning: data definition has no type or >storage class >/usr/src/zaptel/zaptel.c:373: error: syntax error before "chan_lock" >/usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in >declaration of `chan_lock' >/usr/src/zaptel/zaptel.c:373: error: incompatible types in >initialization >/usr/src/zaptel/zaptel.c:373: error: initializer element is not >constant >/usr/src/zaptel/zaptel.c:373: warning: data definition has no type or >storage class >/usr/src/zaptel/zaptel.c: In function `free_tone_zone': >/usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock' >from incompatible pointer type >/usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock' >from incompatible pointer type >/usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone': >/usr/src/zapte
RE: Spam? Re: [Asterisk-Users] Unknown signalling method 'pri_cpe'
Good eye! Its getting late maybe I should just stop now Thank again! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Bockman Sent: Monday, March 13, 2006 8:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Unknown signalling method 'pri_cpe' Hall, Eric M. wrote: > [chan_zap.so] => (Zapata Telephony) > Mar 13 20:44:26 ERROR[10829]: chan_zap.c:10598 setup_zap: Unknown > signalling method 'pri_cpe' Follow the correct order in installing Asterisk as shown on the download page at http://www.asterisk.org zaptel, libpri, asterisk Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unknown signalling method 'pri_cpe'
Anyone have any idea what this is talking about. Here is my zapata.conf [channels] switchtype=5ess signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=default musiconhold=default faxdetect=incoming channel => 1-23 Here is my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for E1 defaultzone=us loadzone=us --- Running asterisk in debug give me this! asterisk -vgcd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf Asterisk SVN-trunk-r7498, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer <[EMAIL PROTECTED]> = Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands Mar 13 20:44:26 NOTICE[10829]: cdr.c:1166 do_reload: CDR simple logging enabled. == RTP Allocating from port range 1 -> 2 Asterisk PBX Core Initializing Registering builtin applications: [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ExecIfTime] == Registered application 'ExecIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [Set] == Registered application 'Set' [ImportVar] == Registered application 'ImportVar' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [res_indications.so] => (Indications Configuration) -- Registered indication country 'at' -- Registered indication country 'au' -- Registered indication country 'br' -- Registered indication country 'be' -- Registered indication country 'ch' -- Registered indication country 'cl' -- Registered indication country 'cn' -- Registered indication country 'cz' -- Registered indication country 'de' -- Registered indication country 'dk' -- Registered indication country 'ee' -- Registered indication country 'es' -- Registered indication country 'fi' -- Registered indication country 'fr' -- Registered indication country 'gr' -- Registered indication country 'hu' -- Registered indication country 'it' -- Registered indication country 'lt' -- Registered indication country 'mx' -- Registered indication country 'nl' -- Registered indication country 'no' -- Registered indication country 'nz' -- Registered indication country 'pl' -- Registered indication country 'pt' -- Registered indication country 'ru' -- Registered indication country 'se' -- Registered indication country 'sg' -- Registered indication country 'uk' -- Registered indication country 'us' -- Registered indication country 'us-o' -- Registered indication country 'tw' -- Registered indication country 'za' -- Setting default indication country to 'us' == Registered application 'PlayTones' == Registered application 'StopPlayTones' [res_agi.so] => (Asterisk Gateway Interface (AGI)) == Registered application 'DeadAGI' == Registered application 'EAGI' == Registered application 'AGI' [res_odbc.so] => (ODBC Resource) Mar 13 20:44:26 NOTICE[10829]: res_odbc.c:265 load_odbc_config: Adding ENV var: INFORMIXSERVER
[Asterisk-Users] Failed installing zaptel
Group Having trouble installing zaptel. Below is my server specs Intel Motherboard D101GGC TE405P CentOS-4.2-i386 Here is the output trying to do a 'make' === make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo rm -f *.ko *.mod.c .*o.cmd rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest [EMAIL PROTECTED] zaptel]# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c -o makefw ./makefw tormenta2.rbt tor2fw > tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw > radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.ELsmp/build make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:372: error: syntax error before "zone_lock" /usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:372: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:372: error: initializer element is not constant /usr/src/zaptel/zaptel.c:372: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:373: error: syntax error before "chan_lock" /usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:373: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:373: error: initializer element is not constant /usr/src/zaptel/zaptel.c:373: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c: In function `free_tone_zone': /usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel/zaptel.c:1035: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1042: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `set_tone_zone': /usr/src/zaptel/zaptel.c:1083: warning: passing arg 1 of `_read_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1095: warning: passing arg 1 of `_read_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_reg': /usr/src/zaptel/zaptel.c:1176: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1199: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1572: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1608: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_ctl_ioctl': /usr/src/zaptel/zaptel.c:3331: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/
RE: [Asterisk-Users] Nat, SIP, Realtime problem
Just wanted to also say this does not happen to all users behind a NAT box on RR or DSL line just a few. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, February 14, 2006 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nat, SIP, Realtime problem Hall, Eric M. wrote: > Asterisk CVS-HEAD dated 2005-08-18 > WhitBox Linux respin 2 > mysql Ver 11.18 Distrib 3.23.58 > Cisco 7960G > > We are using the real-time drivers for sip and everything is working > great. > They have a few employees that use the phones from home on a RR or DSL > line. > The problem is if they make a call everything works great they hang up > and are able to get inbound calls. If they do not make a call for 5 or > 10 mins they are unable to get inbound calls. If they dial out again > its all working for another 5 or 10 mins. This does not happen to all > remote people just a few. Using Realtime SIP peers does not allow for "NAT Keepalive" packets to be sent, so the firewall/NAT devices that those phones are connected to are closing the SIP port hole after an expiration timeout. To fix this, you'll need to upgrade to newer Asterisk (you really should be running 1.2) and use 'realtime caching' for your SIP peers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nat, SIP, Realtime problem
I'm using realtime caching. Here is my sip.conf file [general] callerid=unavailable context=default allowguest=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no rtcachefriends=yes allow=ulaw allow=g729 All other information about the sip clint is keep in the db Thanks again! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, February 14, 2006 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nat, SIP, Realtime problem Hall, Eric M. wrote: > Asterisk CVS-HEAD dated 2005-08-18 > WhitBox Linux respin 2 > mysql Ver 11.18 Distrib 3.23.58 > Cisco 7960G > > We are using the real-time drivers for sip and everything is working > great. > They have a few employees that use the phones from home on a RR or DSL > line. > The problem is if they make a call everything works great they hang up > and are able to get inbound calls. If they do not make a call for 5 or > 10 mins they are unable to get inbound calls. If they dial out again > its all working for another 5 or 10 mins. This does not happen to all > remote people just a few. Using Realtime SIP peers does not allow for "NAT Keepalive" packets to be sent, so the firewall/NAT devices that those phones are connected to are closing the SIP port hole after an expiration timeout. To fix this, you'll need to upgrade to newer Asterisk (you really should be running 1.2) and use 'realtime caching' for your SIP peers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nat, SIP, Realtime problem
Group: I have a customer that is running the following Asterisk CVS-HEAD dated 2005-08-18 WhitBox Linux respin 2 mysql Ver 11.18 Distrib 3.23.58 Cisco 7960G We are using the real-time drivers for sip and everything is working great. They have a few employees that use the phones from home on a RR or DSL line. The problem is if they make a call everything works great they hang up and are able to get inbound calls. If they do not make a call for 5 or 10 mins they are unable to get inbound calls. If they dial out again its all working for another 5 or 10 mins. This does not happen to all remote people just a few. Anyone have any ideas what the heck is going on with this? Thanks for your time. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agent logs
I'm looking for a ay to track when an agent logs in and logs out. Best if it could be put in a mysql db but a text file will be ok for now.. Any help would be great ! Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail crashes asterisk
Thanks I will update via CVS tonight! Thanks again! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Wednesday, August 17, 2005 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voicemail crashes asterisk It was fixed a while ago, download new code. There is a bug in the tracker on it. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Hall, Eric M. > Sent: Wednesday, August 17, 2005 9:23 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Voicemail crashes asterisk > > When a user dial voicemail and just hangs up or enters the wrong > password 3 times asterisk will crash. > > We are using Cisco 7960G with SIP > My asterisk is CVS-HEAD built on 2005-08-02 23:47:59 UTC > > Any help would be great!!! > > > Thanks > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail crashes asterisk
When a user dial voicemail and just hangs up or enters the wrong password 3 times asterisk will crash. We are using Cisco 7960G with SIP My asterisk is CVS-HEAD built on 2005-08-02 23:47:59 UTC Any help would be great!!! Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vmail.cgi and realtime
Has anyone got vmial.cgi to work with realtime drivers? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue/Agents
Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Latest CVS HEAD and the new wct4xxp card
Has anyone used the latest CVS HEAD and the Quad span T1/E1 5 volts card from Digium. I'm not able to get it to load with a modprobe. I have a T100P card and when I install that card it works without any trouble ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] does not implement 'PUBLISH'
Got it! Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Wednesday, July 27, 2005 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] does not implement 'PUBLISH' On Wed, 2005-07-27 at 11:22 -0400, Hall, Eric M. wrote: > Not sure what this is. > When I call my own ext the call will ring for 10 sec and goto the > voicemail. However the phone will keep ringing and I see this on the > asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 > handle_response: Host '192.168.0.200' does not implement 'PUBLISH' > > > Have no idea what this is talking about 192.168.0.200 is a cisco 7960G Have a look at the very long thread yesterday on this very subject. And then update from CVS. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] does not implement 'PUBLISH'
Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about the latest CVS and Zaptel
I'm having trouble with the latest cvs HEAD (7/22/05) and my Wildcard TE405P I just got in from Digium. I'm not able to get podprobe to work with the release. I get an error "unable to install" however when I grab the stable it works great but no realtime drivers for asterisk. I also tried to just get the stable of zaptel and the HEAD of asterisk but asterisk would not load. Any one have any tips? Thanks for taking the time to read this message! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Touch tone problem
Group This is strange. When I call my voice mail extension the system does not pick up my touch tone entries. I have x-lite softphone and a cisco 7960 for my hard phone. When I call from outside I'm able to check my voice mail without any problem. Any help would be great! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about dial out via Zap
Found out something strange.. In zapata.conf if I change the signalling from featd to em_w I'm able to dial out without a problem. But I'm unable to get calls in because of the featd data sent. Change it back to featd and I'm now able to call in but unable to call out. So my question is do I need to do something when calling out for featd? It looks to me like a problem with featd. Below is a copy of my zapata.conf file. zapata.conf [channels] context=from-analog signalling=featd ;signalling=em_w group=1 channel => 1-12 usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes useincomingcalleridonzaptransfer=yes callerid=asreceived echocancel=yes echocancelwhenbridged=yes rxgain=1.0 txgain=1.0 musiconhold=default Thanks Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Monday, August 23, 2004 8:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Question about dial out via Zap Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI> -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1 answered SIP/6149236651-eafb Urgent handler -- Hungup 'Zap/1-1' Urgent handler Urgent handler -- Hungup 'Zap/2-1' Urgent handler Urgent handler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about dial out via Zap
Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI> -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1 answered SIP/6149236651-eafb Urgent handler -- Hungup 'Zap/1-1' Urgent handler Urgent handler -- Hungup 'Zap/2-1' Urgent handler Urgent handler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID problem
My info Asterisk CVS-HEAD-08/04/04 Redhat 9.0 T100P connected to Telco with 12 Digital trunks WINK start. I'm able to dial out and able to get calls coming in but my inbound calls do not display callerid information. Its only shows "asterisk" Telco tells me callerid is turned on and working.. Here is my config files /etc/asterisk/zapata.conf [channels] context=from-analog signalling=em_w group=1 channel => 1-12 usecallerid=yes /etc/zaptel.conf span=1,0,0,esf,b8zs e&m=1-12 loadzone = us defaultzone=us Here is a debug from a call inbound *CLI> -- Saved useragent "CSCO/7" for peer 3000 Urgent handler -- Starting simple switch on 'Zap/1-1' Urgent handler Urgent handler -- Called 3000 Urgent handler -- SIP/3000-ad68 is ringing Urgent handler -- Hungup 'Zap/1-1' Urgent handler Urgent handler Any ideas ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music On Hold - not working for me...
Have you tried to run * in debug mode? I have the same problem and I
found that if I run * in debug (asterisk -vgcd) mode MOH works. I
have no idea why but that is the only way I can get MOH to work for me.
Good luck and please report back to the list if you find a fix!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of avizion
Sent: Wednesday, July 28, 2004 12:58 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Music On Hold - not working for me...
Hi all,
I'm trying to make some simple MOH (Music On Hold) working. So far I've
failed miserably - so I turn here for help.
Basically I've been using the wiki and all the sample confs I could from
there and via google.
The queue system seems to work fine with my limited setup. Just 2 IAX2
clients where I keep Client B busy (by making it listen to mp3 via ext.
777) but logged into the queue. Client A then calls the queue (tried
both ext. 7320 and 6320) and the announcements are fine ("you are next
in line" etc.). When I make Client B not busy - it starts ringing like
it should on the queue. But I never hear the MOH on Client A.
Also - calling 777 does play the mp3 fine - like it should - looped :)
Speaking of 777, I also did: chmod 755 /var/lib/asterisk/mohmp3/*
It's not really stopping me from rolling out this system - but it would
be very nice to have. Any help/pointers appriciated.
Thanks!
Various stuff that might be relevant...
zapata.conf
-SNIP-
musiconhold=default
-SNAP-
musiconhold.conf
-SNIP-
[classes]
default => mp3:/var/lib/asterisk/mohmp3
-SNAP-
extensions.conf
-SNIP-
[macro-queue1]
exten => s,1,Answer
exten => s,2,Queue(${ARG1})
[macro-queue]
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,2
exten => s,4,ResponseTimeout,3
exten => s,5,Background(groovy)
exten => s,6,Queue(${ARG1})
[test]
exten => 6320,1,Macro(queue,Q320)
exten => 6330,1,Macro(queue,Q330)
exten => 6340,1,Macro(queue,Q340)
exten => 6350,1,Macro(queue,Q350)
exten => 6510,1,Macro(queue,Q510)
exten => 69000,1,Macro(queue,Q9000)
exten => 7320,1,Macro(queue1,Q320)
exten => 777,1,Answer
exten => 777,2,MP3Player(/var/lib/asterisk/mohmp3/trickme.mp3)
exten => 777,3,Goto(777,1)
-SNAP-
queues.conf
-SNIP-
[Q320]
announce-frequency = 5
announce-holdtime = yes
strategy = roundrobin
music = default
member => Agent/310,100
member => Agent/312,90
member => Agent/313,10
-SNAP-
outtake from full logfile at
http://relay.dk/~avizion/asterisk/paste1.txt
PS: Should I attach this paste1.txt - or store it elsewhere?
--
avizion on irc.freenode.org #asterisk
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[Asterisk-Users] RE: Install problems
Looks like the 2.6X stuff is not ready yet.. http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation -Original Message- From: Hall, Eric M. Sent: Wednesday, July 21, 2004 6:15 PM To: '[EMAIL PROTECTED]' Subject: Install problems Has anyone install zaptel-1.0-RC1 on Fedora Core 2? First thing I found is I need to have a link to 2.6 from 2.6.5 ln -s /usr/src/linux-2.6.5-1.358/ /usr/src/linux-2.6 fixed this problem. Now I get this. Install gets this error make[2]: *** [/root/asterisk/zaptel-1.0-RC1/zaptel.o] Error 1 make[1]: *** [/root/asterisk/zaptel-1.0-RC1] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' Any ideas? My next step is to try via CVS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Install problems
Has anyone install zaptel-1.0-RC1 on Fedora Core 2? First thing I found is I need to have a link to 2.6 from 2.6.5 ln -s /usr/src/linux-2.6.5-1.358/ /usr/src/linux-2.6 fixed this problem. Now I get this. Install gets this error make[2]: *** [/root/asterisk/zaptel-1.0-RC1/zaptel.o] Error 1 make[1]: *** [/root/asterisk/zaptel-1.0-RC1] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-1.358' Any ideas? My next step is to try via CVS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music on hold
Nothing in the logs about mp3. the startup script is
/etc/rc.d/init.d/asterisk
Here is the file
[EMAIL PROTECTED] root]# cat /etc/rc.d/init.d/asterisk
#!/bin/bash
#
# chkconfig: 2345 99 15
# description: Open source PBX
# processname: asterisk
# source function library
. /etc/rc.d/init.d/functions
RETVAL=0
case "$1" in
start)
echo -n "Starting Asterisk PBX: "
/sbin/modprobe ixj
daemon /usr/sbin/asterisk
RETVAL=$?
echo
[ $RETVAL -eq 0 ] && touch /var/lock/subsys/asterisk
;;
stop)
echo -n "Shutting Asterisk PBX: "
killproc asterisk
/sbin/rmmod -r ixj
RETVAL=$?
echo
[ $RETVAL -eq 0 ] && rm -f /var/lock/subsys/asterisk
;;
restart|reload)
$0 stop
$0 start
RETVAL=$?
;;
status)
status asterisk
RETVAL=$?
;;
*)
echo "Usage: asterisk {start|stop|status|restart|reload}"
exit 1
esac
exit $RETVAL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Maj
Sent: Thursday, July 15, 2004 3:37 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Music on hold
On Wed, 14 Jul 2004, Hall, Eric M. waxed:
> FC1
>
> What I don't understand is why it works using the -vgcd but not
> when just running asterisk ?
Are there any log messages about the mp3 player not being spawned ?
Like "Fork failed" or "unable to spawn mp3player"
?
I am unfamiliar with how FC1 starts a service. Is this something you
added yourself ?
--Chris
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of C. Maj
> Sent: Wednesday, July 14, 2004 5:26 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Music on hold
>
> On Wed, 14 Jul 2004, Hall, Eric M. waxed:
>
> > I have been working on the music on hold part for a few hours today
> > and I found something that just doesn't sound right.
> >
> > If I just run asterisk via service "service asterisk start'
> > everything
>
> > work but MOH If I run it via asterisk -vgcd MOH works...
> >
> >
> > Any idea what the difference is ?
>
> MOH is done via external mpg123 processes, maybe the service stuff
> doesn't like spawning external processes ? What distro are you
> running ?
>
> --Chris
>
>
> --
> Chris Maj, Rochester
> cmaj_at_freedomcorpse_dot_com
> Pronunciation Guide: Maj == May
> ___
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>
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--
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj == May
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RE: [Asterisk-Users] Music on hold
FC1 What I don't understand is why it works using the -vgcd but not when just running asterisk ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Maj Sent: Wednesday, July 14, 2004 5:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold On Wed, 14 Jul 2004, Hall, Eric M. waxed: > I have been working on the music on hold part for a few hours today > and I found something that just doesn't sound right. > > If I just run asterisk via service "service asterisk start' everything > work but MOH If I run it via asterisk -vgcd MOH works... > > > Any idea what the difference is ? MOH is done via external mpg123 processes, maybe the service stuff doesn't like spawning external processes ? What distro are you running ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on hold
I have been working on the music on hold part for a few hours today and I found something that just doesn't sound right. If I just run asterisk via service "service asterisk start' everything work but MOH If I run it via asterisk -vgcd MOH works... Any idea what the difference is ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to place more then 1 call in or out.
Group Everything is working great with my * server. That's to everyone for all your help!!! I have a problem that I can't seem to find a fix for. When I'm on a call and someone calls in the system never picks up. Also I'm unable to place calls out if someone is on the phone. Here is what I have in my system. Please let me know if you need any other information! Let me start by listing my hardware I have 2 X100P cards in the server zaptel.conf # X100P fxsks=1-2 loadzone = us defaultzone=us zapata.conf [channels] group=1 musiconhold=default language=en context=from-analog signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes channel => 1-2 >From the CLI VoIPGW*CLI> zap show channels Chan Extension Context Language MusicOnHold 1from-analog en default 2from-analog en default VoIPGW*CLI> zap show channel 1 Channel: 1> File Descriptor: 22 Span: 1 Extension: Context: from-analog Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No VoIPGW*CLI> zap show channel 2 Channel: 2> File Descriptor: 23 Span: 2CLI> Extension: Context: from-analog Caller ID string: Destroy: 0> Signalling Type: FXS Kewlstart Owner: Real: Callwait: Threeway: Confno: -1> Propagated Conference: -1 Real in conference: 0 DSP: noCLI> Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No VoIPGW*CLI> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for a patch that was post May 1 2004
Hello group I'm working on getting festival installed and working on my FC1. I ran into a problem and after searching Google I found this message talking about a patch for Speech Tools and Festival http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html The above site does not have the files. Does anyone in the group have this patch? Marc Sutter & Reed Wade do you still have it? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about Cisco IP Phone 7960
You the MAN!!! I drop the P of the P0S3-07-1-00 Everything is golden now! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Powell Sent: Thursday, July 08, 2004 9:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Question about Cisco IP Phone 7960 On 08/07/2004 at 08:21 Hall, Eric M. wrote: >I know this is a little off list but I can't think of a better place to >ask this question. > >I upgrade the phone to 7.1 and it installed the Universal Application >Loader. Now I'm getting Protocol Application Invalid after it reads >tftp SIP(MAC).cnf > > >Any ideas? > > >Again sorry this is off topic Make sure you changed ALL the configs to point to the CORRECT image file.. OS79XX.TXT should contain P0S3-07-1-00 and your SIP or SIPDefault.cnf should contain image_version: "P0S3-07-1-00" iirc the default in OS79XX.TXT is the unsigned image... HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about Cisco IP Phone 7960
I know this is a little off list but I can't think of a better place to ask this question. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf Any ideas? Again sorry this is off topic ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 NAT question
I had the same problem. What I found is I needed to set register with proxy to yes in the sip config. Hope this helps -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills Sent: Thursday, July 08, 2004 7:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 NAT question I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The asterisk box is on a WAN connection on the other end of a DS3, the phones connect fine to the Asterisk server as you can see from the output of show sip peers below. tp3/tp3 D N 255.255.255.255 60665 Unmonitored tp2/tp2 D N 255.255.255.255 60646 Unmonitored tp1/tp1 D N 255.255.255.255 60649 Unmonitored Now, the Cisco phones are set to use nat (nat = 1) and in the SIP configuration, the phones are also configured for SIP. [tp1] type=friend secret=tp1 host=dynamic nat=yes callerid="Test Phone 1" I can make calls out over the phones, but can't get anything back in. If I call voicemail say, then that's fine. But if I try and call another phone behind the firewall, it just sits there :/ IS there a specific port range I need to open? Should I be using a different sip config? Cheers for any help, Ben www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 and Voice Mail
I search Google to find how to get the message light to flash on my Cisco 7960 running (Application Load ID POS3-06-3-00) (Boot Load ID PC03M030) (DSP Load ID PS03AT38) All I see is about the sip.conf file witch mine has the mailbox= but still no light. Also the messages button does not work. Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calling an outside phone number as part of a hunt
Thank you! That's what I was thinking but being new I wanted to ask . Thanks again -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Jimenez Sent: Monday, July 05, 2004 11:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Calling an outside phone number as part of a hunt Hall, Eric M. wrote: > I'm trying to see if this is even possible. AFAIK Asterisk has no way of knowing if you do not answer. To Asterisk, the call is complete and "answered" when it starts ringing. A PSTN/POTS call is always going to be the final destination. -- Daniel Jimenez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling an outside phone number as part of a hunt
I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has expired. My hardware is 2 X100P card exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Dial(Zap/1/5551212,10) exten => 2000,3,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about x100P and zap
Ruuing * in debug I get this
*CLI> Jul 5 11:21:02 NOTICE[-1221170256]:
app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' ==
Everyone is busy at this time
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wade J.
WepplerSent: Monday, July 05, 2004 10:54 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Question
about x100P and zap
Your $EXTEN’s need to
be changed to ${EXTEN}. You’ll also need to include any substr #’s within
the brackets (ie. ${EXTEN:1}).
-wade
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.Sent: Monday, July 05, 2004 10:25
AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Question about
x100P and zap
I have 2 X100P card and
configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt
I changed the area codes to match
mine.
When I try to dial out I get
app_dial.c:554 dial_exec: Unable to
create channel of type 'Zap'
A zap show channels gives me
this
Chan Extension
Context Language
MusicOnHold
1
from-analog
en
Any
ideas?
RE: [Asterisk-Users] Question about x100P and zap
I did as you stated however I get the same error. Here is
my config file. Did I miss something?
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wade J.
WepplerSent: Monday, July 05, 2004 10:54 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Question
about x100P and zap
Your $EXTEN’s need to
be changed to ${EXTEN}. You’ll also need to include any substr #’s within
the brackets (ie. ${EXTEN:1}).
-wade
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.Sent: Monday, July 05, 2004 10:25
AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Question about
x100P and zap
I have 2 X100P card and
configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt
I changed the area codes to match
mine.
When I try to dial out I get
app_dial.c:554 dial_exec: Unable to
create channel of type 'Zap'
A zap show channels gives me
this
Chan Extension
Context Language
MusicOnHold
1
from-analog
en
Any
ideas?
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
;
;
;
[from-sip-external]
;
; Take unknown callers that are sending calls to our system,
; and send them to the appropriate extension. It is in this
; area that we do name-to-number mapping for SIP extensions.
;
; This context will allow calls to "[EMAIL PROTECTED]" or calls
; to "[EMAIL PROTECTED]" to be answered on the relevant SIP
; phones. We also do some name-to-number mapping here; see below.
;
; The SIP URI of sip://[EMAIL PROTECTED] will be handled here.
; Note that we assume Sidney is on the SIP phone described as
; extension "2000" in sip.conf, so this short routine just
; re-directs the call flow recursively back into the same
; context, but we change the extension and priority. Since
; we're including [local-extensions], this will get picked up
; by the dialplan contained in local-extensions.
;
; I could be more space-efficient and put all these lines into
; a single regexp, but for clarity I put them each on their
; own lines.
;
; Here are Sidney's aliases
;
exten => sidney,1,Goto(2000,1)
exten => sidney.zweibel,1,Goto(2000,1)
exten => info,1,Goto(2000,1)
;
; ...and John's aliases.
;
exten => john,1,Goto(2001,1)
exten => john.whorfin,1,Goto(2001,1)
exten => sales,1,Goto(2001,1)
;
;
; Include the numbers which we have defined in local-extensions
; and allow them to be accessed from within this context. This
; is how we are able to use the "Goto" commands above, since
; we will be including extensions 2000 and 2001 (and 0 and 2999)
; as available extensions to which we may re-route calls within
; this context.
;
include => local-extensions
;
; If the line hangs up, it's always good to have the "h"
; extension in each context that is the "master" handler
; for calls. This cleanly exits and closes dial path routes.
;
exten => h,1,Hangup
;
; The user has dialed an "i"nvalid number, which means that
; there was no match by any other matching routines. Set an
; absolute timeout on the call (15 seconds), play a Congestion
; tone, and hangup. We set the absolute timeout to prevent easy
; DoS attacks from consuming too much bandwidth. However, it
; is possible that we could still be attacked in some fashion
; by someone making many calls to bogus numbers on our server.
; We could reduce this threat by removing the Congestion
; playback and going straight to hangup, but that is very
; difficult to debug at the remote end, so we are good VoIP
; citizens and we create some audio if the call reaches us.
;
exten => i,1,AbsoluteTimeout(15)
exten => i,2,Congestion
exten => i,3,Hangup
;
;
;
[from-sip-internal]
; Calls that come in from our two SIP phones will land here
; first and match against extensions listed below.
;
; The context [from-sip-internal] is really just a collection
; of include statements that pull the extension matching lists
; in from other contexts. A well-designed dialplan segregates
; extensions with similar functions into contexts, and then
; uses the "include" referencer. This should be a familiar
; concept to anyone who does programming - segmenting a block
; of phone numbers makes them more re-usable in a generic way
; so that the administrator can avoid re-typing the same configs
; over and over.
;
; First, we include [local-extensions], since that's what
; we should try matching on first. If anyone on one of our
; local SIP phones dials an extension that appears in
; [local-extensions], then send the call to whatever priority
; list exists for that number. This is for local-to-local call
; termination.
;
include => local-extensions
;
; Next, we include and try to match against extensions contained
; in [always-out-pots]. These are mostly wildcarded matches,
; so we mak
[Asterisk-Users] Question about x100P and zap
I have 2 X100P card and configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt I changed the area codes to match mine. When I try to dial out I get app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' A zap show channels gives me this Chan Extension Context Language MusicOnHold 1 from-analog en Any ideas?
[Asterisk-Users] Have problem install via cvs
Group Following the information located on http://www.asterisk.org/index.php?menu=download I get the following error installing the zaptel Any help would be great!!! Thanks [EMAIL PROTECTED] zaptel]# make clean; make install rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f *.ko *.mod.c .*o.cmd rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c zaptel.c In file included from /usr/src/linux-2.4/include/asm/semaphore.h:39, from /usr/src/linux-2.4/include/linux/fs.h:202, from /usr/src/linux-2.4/include/linux/capability.h:17, from /usr/src/linux-2.4/include/linux/binfmts.h:4, from /usr/src/linux-2.4/include/linux/sched.h:10, from /usr/src/linux-2.4/include/linux/mm.h:4, from /usr/src/linux-2.4/include/linux/slab.h:14, from /usr/src/linux-2.4/include/linux/proc_fs.h:5, from zaptel.c:45: /usr/src/linux-2.4/include/asm/system.h: In function `__set_64bit_var': /usr/src/linux-2.4/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules /usr/src/linux-2.4/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA makefw.c -o makefw ./makefw tormenta2.rbt tor2fw > tor2fw.h Loaded 69900 bytes from file gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c tor2.c In file included from /usr/src/linux-2.4/include/asm/semaphore.h:39, from /usr/src/linux-2.4/include/linux/fs.h:202, from /usr/src/linux-2.4/include/linux/capability.h:17, from /usr/src/linux-2.4/include/linux/binfmts.h:4, from /usr/src/linux-2.4/include/linux/sched.h:10, from /usr/src/linux-2.4/include/linux/mm.h:4, from /usr/src/linux-2.4/include/linux/slab.h:14, from /usr/src/linux-2.4/include/asm/pci.h:37, from /usr/src/linux-2.4/include/linux/pci.h:658, from tor2.c:33: /usr/src/linux-2.4/include/asm/system.h: In function `__set_64bit_var': /usr/src/linux-2.4/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules /usr/src/linux-2.4/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c torisa.c In file included from /usr/src/linux-2.4/include/asm/semaphore.h:39, from /usr/src/linux-2.4/include/linux/fs.h:202, from /usr/src/linux-2.4/include/linux/capability.h:17, from /usr/src/linux-2.4/include/linux/binfmts.h:4, from /usr/src/linux-2.4/include/linux/sched.h:10, from torisa.c:25: /usr/src/linux-2.4/include/asm/system.h: In function `__set_64bit_var': /usr/src/linux-2.4/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules /usr/src/linux-2.4/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules torisa.c: At top level: torisa.c:1139: warning: `set_tor_base' defined but not used gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c wcusb.c In file included from /usr/src/linux-2.4/include/asm/semaphore.h:39, from /usr/src/linux-2.4/include/linux/fs.h:202, from /usr/src/linux-2.4/include/linux/capability.h:17, from /usr/src/linux-2.4/inc
RE: [Asterisk-Users] All calls go to Voice mail and never ring.
Now its not even going to voice mail.. Here is the output from the debug [EMAIL PROTECTED] asterisk]# asterisk -r == Parsing '/etc/asterisk/asterisk.conf': Found Asterisk CVS-03/31/04-12:57:49, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <[EMAIL PROTECTED]> = Connected to Asterisk CVS-03/31/04-12:57:49 currently running on VoIPGateway (pid = 1748) Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 1ms SCall: 30249 DCall: 0 [24.145.226.226:4569] USERNAME: brett REFRESH : 300 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 3ms SCall: 2 DCall: 30249 [24.145.226.226:4569] AUTHMETHODS : 3 CHALLENGE : 143731950 USERNAME: brett Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00047ms SCall: 30249 DCall: 2 [24.145.226.226:4569] USERNAME: brett REFRESH : 300 MD5 RESULT : ccf4f762dd160c477a78a1fa2f712ad8 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00057ms SCall: 2 DCall: 30249 [24.145.226.226:4569] USERNAME: Brett DATE TIME : 142826960 REFRESH : 60 APPARENT ADDRES : IPV4 24.145.226.226:4569 CALLING NAME: "Eric <111>" Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00047ms SCall: 30249 DCall: 2 [24.145.226.226:4569] USERNAME: brett REFRESH : 300 MD5 RESULT : ccf4f762dd160c477a78a1fa2f712ad8 Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00047ms SCall: 2 DCall: 30249 [24.145.226.226:4569] Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00057ms SCall: 30249 DCall: 2 [24.145.226.226:4569] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 30250 DCall: 0 [24.145.226.226:4569] VERSION : 2 CALLING NUMBER : 7 CALLING NAME: IaxComm User FORMAT : 2 CAPABILITY : 2 USERNAME: brett CALLED NUMBER : 111 DNID: 111 Apr 3 11:46:38 NOTICE[1142106560]: chan_iax2.c:4806 socket_read: Rejected connect attempt from 24.145.226.226 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 9ms SCall: 3 DCall: 30250 [24.145.226.226:4569] CAUSE : No authority found Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 30250 DCall: 0 [24.145.226.226:4569] VERSION : 2 CALLING NUMBER : 7 CALLING NAME: IaxComm User FORMAT : 2 CAPABILITY : 2 USERNAME: brett CALLED NUMBER : 111 DNID: 111 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 3 DCall: 30250 [24.145.226.226:4569] Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 000 Type: VOICE Subclass: 2 Timestamp: 00063ms SCall: 30250 DCall: 0 [24.145.226.226:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00063ms SCall: 3 DCall: 30250 [24.145.226.226:4569] Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 9ms SCall: 30250 DCall: 3 [24.145.226.226:4569] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: Saturday, April 03, 2004 11:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] All calls go to Voice mail and never ring. On Apr 2, 2004, at 7:14 PM, Hall, Eric M. wrote: > I'm starting to get this to work! Well I got Voice Mail to work! > > All calls goes to voice mail without ringing the users phone (iaxComm). > Here is my iax.conf and my extensions.conf > > Any help would be great!! I don't see anything really obviously wrong here, although I didn't spend that much time looking. Try running asterisk with extra debugging (just connect with asterisk -r) and see what it says when you make calls. Odds are, it'll say something that'll be helpful. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] All calls go to Voice mail and never ring.
I'm starting to get this to work! Well I got Voice Mail to work! All calls goes to voice mail without ringing the users phone (iaxComm). Here is my iax.conf and my extensions.conf Any help would be great!! Thanks extensions.conf Description: extensions.conf iax.conf Description: iax.conf
[Asterisk-Users] I'm still a little lost...
I downloaded iaxComm and get up my iax.conf file and the extensions.conf. Here is the out but from CLI in iax debug. What did I forget to do??? Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 1ms SCall: 10489 DCall: 0 [192.168.50.66:4569] USERNAME: 100 REFRESH : 300 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 8ms SCall: 1 DCall: 10489 [192.168.50.66:4569] AUTHMETHODS : 3 CHALLENGE : 455913197 USERNAME: 100 Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00047ms SCall: 10489 DCall: 1 [192.168.50.66:4569] USERNAME: 100 REFRESH : 300 MD5 RESULT : 90dd8ef2853376589a8f9650bf93c034 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00137ms SCall: 1 DCall: 10489 [192.168.50.66:4569] USERNAME: 100 DATE TIME : 142710924 REFRESH : 60 APPARENT ADDRES : IPV4 192.168.50.66:4569 Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00047ms SCall: 10489 DCall: 1 [192.168.50.66:4569] USERNAME: 100 REFRESH : 300 MD5 RESULT : 90dd8ef2853376589a8f9650bf93c034 Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00047ms SCall: 1 DCall: 10489 [192.168.50.66:4569] Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00137ms SCall: 10489 DCall: 1 [192.168.50.66:4569] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 10490 DCall: 0 [192.168.50.66:4569] VERSION : 2 CALLING NUMBER : 7 CALLING NAME: IaxComm User FORMAT : 2 CAPABILITY : 2 USERNAME: 100 CALLED NUMBER : 200 DNID: 200 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00015ms SCall: 2 DCall: 10490 [192.168.50.66:4569] AUTHMETHODS : 3 CHALLENGE : 479986104 USERNAME: 100 Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 10490 DCall: 0 [192.168.50.66:4569] VERSION : 2 CALLING NUMBER : 7 CALLING NAME: IaxComm User FORMAT : 2 CAPABILITY : 2 USERNAME: 100 CALLED NUMBER : 200 DNID: 200 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 2 DCall: 10490 [192.168.50.66:4569] Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00062ms SCall: 10490 DCall: 2 [192.168.50.66:4569] MD5 RESULT : 7ee519af4acc6f18f9dabe631a0e9518 Apr 1 19:04:33 NOTICE[1142106560]: chan_iax2.c:5087 socket_read: Rejected connect attempt from 192.168.50.66, request '[EMAIL PROTECTED]' does not exist Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT Timestamp: 00075ms SCall: 2 DCall: 10490 [192.168.50.66:4569] CAUSE : No such context/extension Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 001 Type: VOICE Subclass: 2 Timestamp: 00094ms SCall: 10490 DCall: 2 [192.168.50.66:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00094ms SCall: 2 DCall: 10490 [192.168.50.66:4569] Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00075ms SCall: 10490 DCall: 2 [192.168.50.66:4569] Rx-Frame Retry[Yes] -- OSeqno: 002 ISeqno: 001 Type: VOICE Subclass: 2 Timestamp: 00094ms SCall: 10490 DCall: 2 [192.168.50.66:4569] VoIPGateway*CLI> ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
I'm trying to use iaxComm and I get the following error. Apr 1 16:18:04 NOTICE[1142106560]: chan_iax2.c:3393 register_verify: No registration for peer 'asterisk' (from x.x.x.x) I'm VERY GREEN with this software so any help on list or off list would be great ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie....
Could I do things like call other ext on the system? Check Voice mail? I would like to test this before I put money in cards I may not need. What Software Phone app is people using? Thanks for all the help so far. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Gudino Sent: Wednesday, March 31, 2004 2:43 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie Hi, On Wed, 2004-03-31 at 16:00, Hall, Eric M. wrote: > I have a question for the group. > To get this running do I need any Digium Cards? I understand I will > need them to connect to the public phone system. I'm looking at just > using IP Phones or IP Softphones just to test this app. You can certainly use Asterisk without Digium hardware. But some applications will not work out of the box, like music on hold and meetme. For them to work you may need to compile ztdummy (uncomment the appropiate line in zaptel Makefile), and make sure that your sip clients transmit silence. If you are running RedHat or Fedora, start asterisk with LD_ASSUME_KERNEL=2.4.1 Good luck, -- Nicolas Gudino <[EMAIL PROTECTED]> House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie....
Now for the next question. I have an old AT&T Merlin Mail system with a Brooktrout comcode series 4 cards in it. Could I use them? Thanks again for your help -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Gudino Sent: Wednesday, March 31, 2004 2:43 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie Hi, On Wed, 2004-03-31 at 16:00, Hall, Eric M. wrote: > I have a question for the group. > To get this running do I need any Digium Cards? I understand I will > need them to connect to the public phone system. I'm looking at just > using IP Phones or IP Softphones just to test this app. You can certainly use Asterisk without Digium hardware. But some applications will not work out of the box, like music on hold and meetme. For them to work you may need to compile ztdummy (uncomment the appropiate line in zaptel Makefile), and make sure that your sip clients transmit silence. If you are running RedHat or Fedora, start asterisk with LD_ASSUME_KERNEL=2.4.1 Good luck, -- Nicolas Gudino <[EMAIL PROTECTED]> House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie....
I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
