[asterisk-users] better handling of calls forwarded by SIP phones
Hello,
When a user forwards his SIP phone to another extension (say an absent
boss to his secretary) I'd like the unanswsered forwarded call to end up
in the new destination's voicemail. With my current diaplan the call is
handled by the original recipient's voicemail:
[macro-stdexten]
exten = a,1,VoicemailMain(${MACRO_EXTEN})
exten = s,1,Dial(SIP/014647${MACRO_EXTEN}|${RINGTIME}|t|)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(su${MACRO_EXTEN})
exten = s-NOANSWER,n,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(sb${MACRO_EXTEN})
exten = s-BUSY,n,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
Ideally the dialplan would need to detect that the call was forwarded
and not Goto voicemail.
Any idea?
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[asterisk-users] IAX connection to FWD not working
Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 814179 Unregistered 60 Timeout 192.246.69.186:4569 805208 Unregistered 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web interfaces are available today for debian based Asterisk installation?
Lee wrote: Maxim Veksler wrote: I am aware of both of these tools, I don't like them! They make absolute changes in your /etc/asterisk/* files, they assume that they are the only thing you will be using for managing your asterisk pbx and they are both totally unfriendly to 3rd party changes. Yup, which is precisely why the webtools we built (see post from Michiel, thanks!) will only write into separate files that can be #included. Florian ___ --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] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13 [ Virusgeprüft]
Hi, sure in an small office you can use iaxmodem/hylafax to receive faxes - we use it for sending faxes, but would you try to set up about 100 iaxmodems inside hylafax if you can handle it directly inside asterisk with rx_fax and a small script ? [EMAIL PROTECTED] schrieb am 20.12.2006 02:17:22: Hi, No IaxModem is only a modem simulator. Let asterisk do the difference, and send it to you iax extension... @++. Jean-Yves Avenard a écrit : Hi On 12/20/06, Lee Howard [EMAIL PROTECTED] wrote: This thread seems like an awfully crazy amount of work to get fax working when using IAXmodem and HylaFAX would do it without the headache, most likely. Does IAXmodem allows you to receive faxes with any extensions (auto-detecting incoming faxes). JY ___ --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 Diese E-Mail und alle Anhänge enthalten vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese E-Mail und ihren Inhalt. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser E-Mail ist nicht gestattet. This e-mail and any attached files may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail by mistake) please notify the sender immediately and delete this e-mail. Any unauthorised duplication, disclosure or distribution of this e-mail and content is strictly forbidden. ___ --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] Day/night service and indications on the phone
Olivier ha scritto:
I'm happy to report that with a very litte change to app_devstate.c
(just in the way ast_device_state_changed_literal() is called)
that module just compiles and works fine even without bristuffing
anything.
BTW I'm using a Thomson ST2030S phone with a status key subscribed
to a DS/xxx hint.
Thanks again for your precious help!
Could you elaborate ?
How is it working now ?
How you extensions.conf file looks like ?
Regards
Here's what I've got:
Configuration file for operator's phone:
...
[sys]
...
FeatureKeyExt10=S/sip:700
...
extensions.conf (within phone sip account's context):
...
;day-night service
exten = 700,hint,DS/night
exten = 700,1,DBGet(night=DEVSTATES/night)
exten = 700,n,GotoIf($[ ${night} = 2 ]?disable)
exten = 700,n,Devstate(night,2)
exten = 700,n,Playback(custom/night-service-on)
exten = 700,n,Hangup()
exten = 700,n(disable),Devstate(night,1)
exten = 700,n,Playback(custom/night-service-off)
exten = 700,n,Hangup()
(I have of course my own audio files that prompt
the operator about night service status)
The operator turns on/off the night service by just
pressing the F10 key on the phone, and its led
adjusts accordingly.
As to app_devstate.c, I've replaced any occurence of
ast_device_state_changed_literal(), which in bristuffed asterisk
takes 3 parameters (devname, cid, cidname) with
ast_device_state_changed_literal(devName)
as the original asterisk prototype requires (I don't care about
cid and cidname for this specific function).
To compile it outside bristuffed asterisk, just copy app_devstate.c to
the apps directory then edit the Makefile in it, adding
APPS+=app_devstate.so
after the first APPS= assignment.
I suggest you to do make and copy manually the resulting app_devstate.so
to your asterisk modules directory, instead of doing make install,
then issue a load app_devstate.so on the asterisk cli without restarting
it.
Thanks for the tip about thomson blf and firmware.
I'll try to trace sip dialog between thomson and chan_sip, although I'm
not very much into development. With some amount of luck I can try to
change the behavior of chan_sip code
Alberto.
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[asterisk-users] asterisk run on vxworks for hardware pbx
Hi
My hardware PBX run asterisk on vxworks,Because the vxworks not support
perl.
Now I want to add a callback function to my pbx.
now it can store Caller and Called party numbers in queue when Called party
is busy
Then I malloc a new ast_channel to call.It is should use
ast_get_channel_by_exten_locked() or ast_channel_alloc() ,
my program as follow,But it isn't work, anyone know how to do this.
{
struct ast_channel *callbk;
char *callbk_real_context;
char xferto[256],dialstr[265];
char *cid_num;
char *cid_name;
int outstate=0;
char *exten = NULL ,*context = NULL;
pu = head; //pu is a queue hav dst and src number
printfl(\n\n\n\n %s time is over,pu-dst);
show_queue(head);
memset(xferto, 0, sizeof(xferto));
//callbk = ast_channel_alloc(0);
callbk = ast_get_channel_by_exten_locked(pu-dst,
context);
if (!ast_strlen_zero(callbk-macrocontext))
callbk_real_context =
callbk-macrocontext;
else
callbk_real_context = callbk-context;
ast_copy_string(xferto,pu-dst,sizeof(xferto));
cid_num = callbk-cid.cid_num;
cid_name = callbk-cid.cid_name;
if (ast_exists_extension(callbk,
callbk_real_context,xferto, 1, cid_num)) {
snprintf(dialstr, sizeof(dialstr),
[EMAIL PROTECTED]/n, xferto, callbk_real_context);
}
callback_request_and_dial(callbk, Local,
ast_best_codec(callbk-nativeformats), dialstr, 15000, outstate, cid_num,
cid_name);
}
static struct ast_channel *callback_request_and_dial(struct ast_channel
*caller, const char *type, int format, void *data, int timeout, int
*outstate, const char *cid_num, const char *cid_name)
{
int cause = 0;
struct ast_channel *chan;
if ((chan = ast_request(type, format, data, cause))) {
ast_set_callerid(chan, cid_num, cid_name, cid_num);
ast_channel_inherit_variables(caller, chan);
printfl(\n\n In if ((chan = ast_request(type, format, data, cause;
if (!ast_call(chan, data, timeout)) {
dosomething;
}
dosomething;
}
}
thinks!
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Re: [asterisk-users] Echo problem
On 12/20/06, Jason Bachman [EMAIL PROTECTED] wrote: As I understand it, the echo cancelers in Asterisk only work with the Analog cards (FXS/FXO). Not true, echo is caused by any number of things in the voice network, so Asterisk will echo cancel any Zap device. We use it to cancel ISDN2e and ISDN30 E1 lines very successfully. If you are getting echo on a digital line, there is a problem with either a DAC, the T1 clocking, or you are getting bit errors. Again, not true. The echo is (mostly) not caused in or by asterisk, it is caused out there. Even if a call is digital end-to-end, there is the posibility of acoustic echo in the handsets. Of course the above problems might also cause echo, but I expect they would also cause a log full of errors :) You have a Switch in the middle - perhaps the switch is doing doing digital-analog conversions instead of sending the digital data straight through. The cause of the echo could very well be there, and the echo cancelers (even if they worked on a digital line) would not help because the cause of the echo is somewhere else, not at the Digium card. Check your Tadiran switch for any echo cancel options. I'm not familiar with that switch so I am no help to you on that, but I am pretty sure that its not the Digium card or Asterisk. I agree, that is a very good candidate. AD/DA conversions in this device would IMHO make it responsible for cancelling any resultant echo, and the conversions could indeed add significant delay. Regards, --Jason Bachman Scott Gifford wrote: Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds; none of these adjustments made a difference, except adjusting gain made the echo quieter. 1.5 to 2 seconds. That is a HUGE delay. echo delay is normally measured in tens or perhaps hundreds of milliseconds, and you are unlikely to find a software EC that can deal with a 1.5 to 2 second delay! This sounds as if there is something very broken in the voice network, causing huge amounts of delay. As suggested above, check the intermediate switch. [snip] We have done loopback tests with the Digium card with a loop plug in it. What were the results? Cheers, Steve ___ --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] AstManProxy - Manager
Astmanproxy is just a proxy. It it just taking the load off asterisk for multiplexing multiple Asterisk manager connections, but it does not change the protocol (except to add a couple of features) unless you select one of the non standard plugins. Regards, Steve On 12/19/06, Daniel Gradecak [EMAIL PROTECTED] wrote: Hello, I cannot find documnetation explaining how to access AstManProxy. I am working with Asterisk Java and accessing Asterisk Manager. I wonder if AStManProxy is using the same API as Manager? Can I access it with Asterisk java too ? Regards, Daniel ___ --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] Need Wholesale Termination
Hi Shady, You'll have better luck posting this to the -biz list. This list is for non-commercial discussion only. Alex On 12/20/06, Shady [EMAIL PROTECTED] wrote: Looking for a good termination provider for US/Canada Please contact offlist. Shady ___ --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 -- Alex Robar [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
Re: [asterisk-users] IAX connection to FWD not working
Hi Timothy, Mine seems to be working OK as of a few minutes ago: unlimited*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 727044 216.58.41.183:4569 60 Registered Do you have any other IAX trunks? Are they working for you? Alex On 12/20/06, Timothy Parez [EMAIL PROTECTED] wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 814179 Unregistered 60 Timeout 192.246.69.186:4569 805208 Unregistered 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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 -- Alex Robar [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
Re: [asterisk-users] IAX connection to FWD not working
That's odd :) It's been like this for days I post a message and it's up ? :) They are now registered :) Cool. Alex Robar wrote: Hi Timothy, Mine seems to be working OK as of a few minutes ago: unlimited*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 http://192.246.69.186:4569 727044 216.58.41.183:4569 http://216.58.41.183:4569 60 Registered Do you have any other IAX trunks? Are they working for you? Alex On 12/20/06, *Timothy Parez* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 http://192.246.69.186:4569 814179 Unregistered 60 Timeout 192.246.69.186:4569 http://192.246.69.186:4569 805208 Unregistered 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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 -- Alex Robar [EMAIL PROTECTED] mailto:[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 - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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] IAX connection to FWD not working
However I can call 613 and it works I can be called and it works but when I call any other number I get call ended right away :p Alex Robar wrote: Hi Timothy, Mine seems to be working OK as of a few minutes ago: unlimited*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 http://192.246.69.186:4569 727044 216.58.41.183:4569 http://216.58.41.183:4569 60 Registered Do you have any other IAX trunks? Are they working for you? Alex On 12/20/06, *Timothy Parez* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 http://192.246.69.186:4569 814179 Unregistered 60 Timeout 192.246.69.186:4569 http://192.246.69.186:4569 805208 Unregistered 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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 -- Alex Robar [EMAIL PROTECTED] mailto:[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 - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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] AGI Help Please
Below are a few errors in the script and on a google search, although I found people with the same error, I didn't find a resolution. Any thoughts on what is causing this error? Any thoughts as to why the output is not showing on the CLI without doing a debug? snip Content-type: text/html X-Powered-By: PHP/4.3.9 These 2 lines should not be there. AGI Tx AGI Rx AGI Tx 510 Invalid or unknown command AGI Rx AGI Tx 510 Invalid or unknown command These 2 errors are probably caused by the Content-type and X-Powered-By lines. AGI Rx VERBOSEThere have been AGI Tx 510 Invalid or unknown command AGI Rx VERBOSE125 calls made AGI Tx 510 Invalid or unknown command According to this page http://www.voip-info.org/wiki/view/verbose Usage: Verbose(message [level]) Also, you usually put error_reporting(0); at the top of the script so you won't have warnings and errors confusing Asterisk. I never wrote a PHP AGI without using this : http://phpagi.sourceforge.net/ so I can't help you much You should give it a try, you might like it :) hth ___ --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: Match a Numer - then continue with dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG So, in the event that the logic flows beyond
DG coo1_OnNet, we want to reset the caller id of say, 3254001 Doug,
DG to 3254000 Widgets Inc.
DG exten = 3254101,1,Dial(SIP/3254101,20,tr)
DG exten = 3254102,1,Dial(SIP/3254102,20,tr)
DG exten = 3254103,1,Dial(SIP/3254103,20,tr)
[coo1_CallStart]
include = coo1_OnNet
You want something which executes here, if coo1_OnNet didn't match?
exten = _.,1,Set(CALLERID(all)=Widgets Inc 3254001)
will do that.
If you then want to continue in priority 1 instead of 2, you just do
exten = _.,n,Goto(coo1_CallStart2,${EXTEN},1)
[coo1_CallStart2]
include = syst_OnNet
include = syst_OffNet
/Benny
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[asterisk-users] Need quality toll free 800 number over IAX?
Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd ___ --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] IAX connection to FWD not working
You mean that you can't call other FWD users? Alex On 12/20/06, Timothy Parez [EMAIL PROTECTED] wrote: However I can call 613 and it works I can be called and it works but when I call any other number I get call ended right away :p Alex Robar wrote: Hi Timothy, Mine seems to be working OK as of a few minutes ago: unlimited*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 http://192.246.69.186:4569 727044 216.58.41.183:4569 http://216.58.41.183:4569 60 Registered Do you have any other IAX trunks? Are they working for you? Alex On 12/20/06, *Timothy Parez* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 http://192.246.69.186:4569 814179 Unregistered 60 Timeout 192.246.69.186:4569 http://192.246.69.186:4569 805208 Unregistered 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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 -- Alex Robar [EMAIL PROTECTED] mailto:[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 - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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 -- Alex Robar [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
[asterisk-users] AgentCallbackLogin() deprecated in 1.4
Hello all, I've seen that the application AgentCallbackLogin()has been set to deprecated in version 1.4. So I've done some tests based on the tutorial queues-with-callback-members.txt coming with version 1.4. What's not clear for me is what is happening to agents.conf, it seems that it's no longer needed, and I have to define my agents using variables in extensions.ael. The other thing is, that show agents doesn't show me which agents are logged in and if I use show queue I can see local channels attached to a queue but no agents. For my point of view there is some functionality lost with the new concept. If I want to program a realtime display to show agentstates in queues based on the output from show queue, what's the concept to map agents to the local channels? How can I configure agents in future? Any comments regarding that topic are appreciated. Thanks and Regards Markus ___ --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] IAX connection to FWD not working
The same with our servers. I just deleted the FWD trunk. That took less time and quit using the FWD Account If anyone has any info on why please let me know. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Timothy Parez wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 814179 Unregistered 60 Timeout 192.246.69.186:4569 805208 Unregistered 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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 Inbound (clean). Database: 0660-0, 12/19/2006 - 12/20/2006 3:24:32 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
Re: [asterisk-users] AstManProxy - Manager
Hi, Is AstManProxy an alive project ? It seems to me that no development are ongoing. Will AstManProxy comply with Asterisk 1.4 ? Regards ___ --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] Echo problem
We followed these instructions in trying to eliminate echo: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/doc s-html/x1695.html Our lines come in from the telco in a PRI, then connect to a Tadiran switch which hands the lines off to Asterisk over a T1 card. Checkout the Digium KB: http://kb.digium.com/19/ You will see a suggestion to adjust the gain levels as well. Even though the echo is there, it helps to not make it noticeable to the users. I just found this as well, although they are trying to sell their product at the same time, it helps explain echo and some steps in Asterisk for reducing echo: http://www.xorcom.com/pdfs/AB007_Echo.pdf Michael L. Young ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No music on hold?
I installed the asterisk-addons from source and installed them. It looks like it copied format_mp3.so but I'm not sure if 1.2.14 addons are compatible with asterisk 1.2.10. Also I unpacked the asterisk source for the 3 MOH .mp3's and copied them to the appropriate location. Still MOH is not working. Any other ideas? Thanks again, Phil Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. J. ___ --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] BLF on GXP2000
Use 'show hints' in the CLI to see if they are actually registering
changing status. It sounds like they're registering but not changing
status.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Johnson
Sent: Tuesday, December 19, 2006 6:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] BLF on GXP2000
Rebooted the phone...No luck
On 12/19/06, Ken Williams [EMAIL PROTECTED] wrote:
One thing I've noticed, is any time I make changes to Asterisk I
have to reboot the phones to keep BLF up to date. Have you tried that?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Johnson
Sent: Monday, December 18, 2006 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] BLF on GXP2000
Well, I am making some progress. I have made some changes as
defined below and now have a green line on the BLF, but it still does
not indicate when the extension receives a call or goes off hook.
Here are the changes:
the [ext-local-custom] context no longer exists
the subscribecontext in sip.con no longer exists
[internal]
exten = 101,1,Macro(voicemail,${polycom430})
exten = 101,hint,${polycom430}
Asterisk 1.4.0b3
*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
[EMAIL PROTECTED]: SIP/101
State:Idle Watchers 1
- 1 hints registered
On 12/18/06, Ken Williams [EMAIL PROTECTED]
wrote:
Here's what I have, it's to early for me to think so
hopefully looking at mine helps :D
extensions.conf:
[ext-local]
exten = 701,1,Macro(exten-vm,701,701)
exten = 701,n,Hangup
exten = 701,hint,SIP/701
sip.conf:
[701]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/701
context=from-internal
canreinvite=no
callerid=device 701
mailbox=701
If this doesn't help in some fashion let me know and
I'll think it through a little later...off to get some coffee.
From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] On Behalf Of Chris Johnson
Sent: Sunday, December 17, 2006 4:50 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] BLF on GXP2000
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extensions.conf:
[globals]
polycom430=SIP/101
[internal]
exten = 101,1,Macro(voicemail,${polycom430})
[macro-voicemail]
exten = s,1,Dial(${ARG1},10,tT)
exten = s,2,VoiceMail([EMAIL PROTECTED] )
exten = s,102,VoiceMail([EMAIL PROTECTED])
[ext-local-custom]
exten = 101,hint,${polycom430}
sip.conf:
[general]
subscribecontext=ext-local-custom
And have set up the key to Asterisk BLF with UserID101
When I reload the phone, I get the following error:
[Dec 17 13:34:33] ERROR[2551]: chan_sip.c:14064
handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from
192.168.1.248 http://192.168.1.248/ , but there is no hint for that
extension
Any help is greatly appreciated.
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Re: [asterisk-users] IAX connection to FWD not working
Indeed, they can call me, I can call 613 but not them Their phone rings for like 1 second. I get callended. Alex Robar wrote: You mean that you can't call other FWD users? Alex On 12/20/06, *Timothy Parez* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: However I can call 613 and it works I can be called and it works but when I call any other number I get call ended right away :p Alex Robar wrote: Hi Timothy, Mine seems to be working OK as of a few minutes ago: unlimited*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 http://192.246.69.186:4569 http://192.246.69.186:4569 727044 216.58.41.183:4569 http://216.58.41.183:4569 http://216.58.41.183:4569 60 Registered Do you have any other IAX trunks? Are they working for you? Alex On 12/20/06, *Timothy Parez* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 http://192.246.69.186:4569 http://192.246.69.186:4569 http://192.246.69.186:4569 814179 Unregistered 60 Timeout 192.246.69.186:4569 http://192.246.69.186:4569 http://192.246.69.186:4569 805208 Unregistered 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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 - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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 -- Alex Robar [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments
Re: [asterisk-users] AgentCallbackLogin() deprecated in 1.4
On Wed, 20 Dec 2006 14:39:42 +0100
Markus Bönke [EMAIL PROTECTED] wrote:
Hello all,
The other thing is, that show agents
doesn't show me which agents are logged in and if I use show queue
I can see local channels attached to a queue but no agents. For my
point of view there is some functionality lost with the new concept.
Snap! I've been designing an * system for our call centre and fallen
into exactly the same trap. I ended up coding my own agent login/logout
procedures using astdb functions to store the extension at which an
agent is sitting...
However what I'm missing most is a 'wallboard' for 'number of agents
on Do-Not-Disturb' / number of waiting calls / average wait time, so I'm
considering QueueMetrics, but E 2500 is a lot of cash for that one
feature. I'll probably get one of the codies here to knock something
together.
If I want to program a realtime display to show agentstates in queues
based on the output from show queue, what's the concept to map
agents to the local channels? How can I configure agents in future?
Well, you might want to make use of the 'pre-queue AGI' facility, and
use that to set that agent as 'on call' (Postgres/MySQL or just AstDB)
in that.. then after the Queue application exits, use ${UNIQUEID} to
change the state of the agent to 'free' via func_odbc or another AGI.
If you specify setinterfacevar=yes in queues.conf, then
you can do 'GET VARIABLE MEMBERINTERFACE' in the AGI to find the name
of the Local/ channel that the caller is about to be connected to.
Then look that up in AstDB... here's what I do - it's heavily based on
the agi-test.agi that comes with Asterisk.
# Which queue member was this incoming caller about to speak to?
print GET VARIABLE MEMBERINTERFACE\n;
my $result = STDIN;
checkresult($result);
# Incoming string is 200 result=1 (Local/[EMAIL PROTECTED]) so we need to
# trim the fat
$aid=$result;
$aid =~ s/.*Local\///;
$aid =~ s/[EMAIL PROTECTED]//;
chop $aid; # drop the end carriage return
# This /has/ to work because this is the same logic that the 'agents'
# context uses in the dialplan!
print DATABASE GET LRCC $aid\n;
my $result = STDIN;
checkresult($result);
# More trimmings.
$ext=$result;
$ext =~ s/.*\(//;
$ext =~ s/\).*//;
chop $ext;
You can then go on and do...
$sql=UPDATE agent_status SET status = 'on call', uniqueid='.$AGI
{'uniqueid'}.', extension='.$AGI{'ext'}.' WHERE agentid='$aid';
$dbh-do($sql);
.. then just view the contents of the agent_status table. I hope that
makes sense - it was a bit of a ramble :) 1.4 has been a lot of fun so
far - I'm using a lot of the new features and doing stuff that I
couldn't have thought of with 1.2 :)
Cheers,
Gavin.
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Re: [asterisk-users] Billing solution
Giedrius, did you read my post? Doesn't seem so as I ask for solution that does NOT require to modify my dialplan. On 12/20/06, Giedrius Augys [EMAIL PROTECTED] wrote: 2006/12/20, C F [EMAIL PROTECTED]: Well I did: astpp http://www.astpp.org/ On 12/20/06, Vicky [EMAIL PROTECTED] wrote: I am looking for exactly same kind of billing stuff but i dont think you will get it without letting ur billing program make some changes in asterisk . On 20/12/06, Carlos Rojas [EMAIL PROTECTED] wrote: a2billing Is very good On 12/19/06, Giedrius Augys [EMAIL PROTECTED] wrote: 2006/12/19, C F [EMAIL PROTECTED]: Can anyone recommend a call accounting solution with rating for post paid billing that works well with asterisk using the account code or any other info from the CDR? I don't want the billing software to any phone calls for me, therefore any solution that modifies my extensions.conf is out, nor does it have to allow for customers the ability to log in to check their usage/balances. I have looked at astbill but it looks to be way overcomplicated for what I want, as well as it requires realtime. Thank you ___ --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 Mor and Mcc ___ --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 ___ --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 As I said , MCC would the best solution for you ( http://www.kolmisoft.com/ ). You will compile app mcc2 , and you use this app as Dial command in extensions.conf . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] AstManProxy - Manager
On Wed, Dec 20, 2006 at 02:57:17PM +0100, Olivier wrote: Hi, Is AstManProxy an alive project ? It seems to me that no development are ongoing. Will AstManProxy comply with Asterisk 1.4 ? Last release seems to be from 3 monthes ago. 1.4 has not been released yet, as you recall. Anyway, latest astmanproxy seems to have a basic support for the manager over HTTP protocol of 1.4. But maybe this is just me reading the docs wrong. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Thomson ST2030S and BLF
Olivier ha scritto: Alberto, Call pickup is not implemented yet within Thomson ST2030 (1.50 firmware). More precisely, call pickup current implementation is not Asterisk compliant. A new release is scheduled for February (I've got this confirmed by Thomson 10 minutes ago) but we don't know if call pickup will be included. Regards I'd like to know what kind of compliance is required. I've tried to track what happens when a subscriber line key is pushed: - The Thomson phone sends an initial SUBSCRIBE message to Asterisk (each message is actually send twice, the first as anonymous, NACKed, the second with md5 digest auth, ACKed by asterisk) - The Thomson phone sends subsequent periodic SUBSCRIBE refreshing messages to Asterisk - When the SIP channel whose extension is hint-ed in extensions.conf gets busy/ringing/etc., Asterisk sends a NOTIFY message with a xml body containing the updated status on the line - The Thomson ACKs the NOTIFY and updates the LED status accordingly These steps work regularly. Now, when a line is ringing, if I press the flashing line key, the Thomson sends a SUBSCRIBE message to Asterisk instead of an INVITE (which is sent, on the opposite, when the line key is not flashing). Asterisk replies (I guess) correctly by ACKing and sending a NOTIFY (which is also ACKed by Thomson). Then nothing happens, the phone gives an error and...voilà, the key keeps flashing fast until next reboot. I wonder why the ST2030 sends a SUBSCRIBE upon key press when the key is flashing, while it sends an INVITE when the key is lit or off. Any clue on that? What is the ST2030 expecting back from Asterisk in order to proceed with call pickup?? It looks like the phone is NOT willing to send any pickup request... ___ --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] AGI Help Please
Try running it as
./test.php
the hash-bang should take care of the php-location.
The first two lines are one cause of your problem. Could be the lack of
the -q param for php. However, I would expect the script to not show
anything, as it should be reading params from asterisk first. Iirc,
asterisk is picky about the order in which commands are accepted --
sending commands (such as VERBOSE) before pulling its output buffer,
could be problematic.
William Piper wrote:
Jay,
I just tried the suggested changes... same response.
I tested the script via command-line it works fine.
[EMAIL PROTECTED] agi-bin]# php test.php
Content-type: text/html
X-Powered-By: PHP/4.3.9
VERBOSEThere have been
VERBOSE1 calls made
[EMAIL PROTECTED] agi-bin]#
The permissions are correct:
-rwxr-xr-x 1 root root 1004 Dec 19 23:42 test.php
Any other thoughts?
Thanks,
bp
On 12/19/06, *Jay Milk* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Does the script run from command-line? Without taking a close look at
this, the include statements in the function body of connect_db look
potentially messy.
Also, any output to stdout is interpreted by asterisk as a command, so
those fputs statements would be a problem -- do
fputs($stdout,VERBOSE \There have been\\n);
fputs($stdout,VERBOSE \$row_count calls made\\n);
instead.
William Piper wrote:
List,
I finally decided to break down start playing with AGI
scripts, but
for the life of me, I can't figure out what I am doing wrong.
Below is a super simple script to run a query in mysql to see
how many
call records there are for the extension calling in, then print the
total in the CLI.
This is all I get on the CLI:
-- Executing AGI(SIP/216-0baa, test.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
-- AGI Script test.php completed, returning 0
-- Executing Hangup(SIP/216-0baa, ) in new stack
Here is the script:
#!/usr/bin/php -q
?php
ob_implicit_flush(false);
set_time_limit(6);
$stdin = fopen(php://stdin,r);
$stdout = fopen('php://stdout', 'w');
function read() {
global $stdin, $debug;
$input = str_replace(\n, , fgets($stdin, 4096));
return $input;
}
function connect_db() {
$database=asteriskcdrdb;
include(./common.php);
include(./dbconnect.php);
}
// parse agi headers into array
while ($env=read()) {
$env = str_replace(\,,$env);
$s = split(: ,$env);
$agi[str_replace(agi_,,$s[0])] = trim($s[1]);
if (($env == ) || ($env == \n)) {
break;
}
}
// main program
$clid = $agi[callerid];
connect_db();
$query1 = SELECT * FROM cdr WHERE dst = '$clid' ;
$query_result1 = @mysql_query($query1);
$row_count = mysql_num_rows($query_result1);
$row1 = @mysql_fetch_array ($query_result1);
fputs($stdout,There have been\n);
fputs($stdout,$row_count calls made\n);
fflush($stdout);
fclose($stdin);
fclose($stdout);
exit;
?
There are no debug errors and the query is going through just
fine...
and yes, I chmod 755.
Does anyone have a clue what I am doing wrong?
Thanks,
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[asterisk-users] Dial 9 to get out?
Hi all, Can someone point me in the right direction here. What I'd like to do with Asterisk is a) dial a 3 digit extention (i.e. 100) on my polycom phones and after the 3rd digit is entered, it dials that extension and b) dial 9 to get out like older PBX systems. Since my internal extensions start with a 1 I think what happens is I enter extension 100 for example, and the phone sits there. If I enter 1, areacode, number the moment I enter the last digit of the number it dials the number. ALSO I'd like to be able to dial local numbers without using 1+areacode. Note that I'm using voicepulse so it makes sense that it isn't intelligent enough to know when a number is a local one or not. Thanks! Phil ___ --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] Dial 9 to get out?
Look at the digit map in your Polycom configuration files. I had the same problem and had to chage the digit map to support an extra digit when dialing 9. On 12/20/06, Phil Finkler [EMAIL PROTECTED] wrote: Hi all, Can someone point me in the right direction here. What I'd like to do with Asterisk is a) dial a 3 digit extention (i.e. 100) on my polycom phones and after the 3rd digit is entered, it dials that extension and b) dial 9 to get out like older PBX systems. Since my internal extensions start with a 1 I think what happens is I enter extension 100 for example, and the phone sits there. If I enter 1, areacode, number the moment I enter the last digit of the number it dials the number. ALSO I'd like to be able to dial local numbers without using 1+areacode. Note that I'm using voicepulse so it makes sense that it isn't intelligent enough to know when a number is a local one or not. Thanks! Phil ___ --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 -- Bruce Nortex Networks ___ --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-LOW: Re: [asterisk-users] What web interfaces are available today for debian based Asterisk installation?
Florian Overkamp wrote: Lee wrote: Maxim Veksler wrote: I am aware of both of these tools, I don't like them! They make absolute changes in your /etc/asterisk/* files, they assume that they are the only thing you will be using for managing your asterisk pbx and they are both totally unfriendly to 3rd party changes. Yup, which is precisely why the webtools we built (see post from Michiel, thanks!) will only write into separate files that can be #included. Florian See, I like that. I may have to place a flag in compilation process of my software to do a full update (replacement) or go into separate files that can be linked to the original extensions.conf, etc. If only to have more flexibility. -- 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
RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message-
From: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 10:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
On Tue, Dec 19, 2006 at 05:19:57PM -0700, Douglas Garstang wrote:
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 4:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
Please correct me if I'm misunderstanding your
requirements, but see
below (inline) for what I would do:
-Original Message-
[snip]
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = syst_OffNet
Instead of including your system-wide logic for offnet calling,
introduce a per-company offnet and include that instead:
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = coo1_OffNet
[coo1_OffNet]
exten = _X.,1,Set(CALLERID(NUM)=3254000)
exten = _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
exten = _X.,3,Goto(syst_OffNet,${EXTEN},1)
Bradley, If I do this, then I can no longer continue with further
extensions in my dialplan as Asterisk has already matched a number.
An explicit WaitExten?
No I don't want the user to have to enter another number. Processing should
continue with the original number dialled.
*sigh*
Doug.
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Re: [asterisk-users] No music on hold?
Phil, did you add letter 'm' to your dial options?? exten = _XXX,1,Dial(SIP/XXX,60,m) Regards Arlen Nascimento On 12/20/06, Phil Finkler [EMAIL PROTECTED] wrote: I installed the asterisk-addons from source and installed them. It looks like it copied format_mp3.so but I'm not sure if 1.2.14 addons are compatible with asterisk 1.2.10. Also I unpacked the asterisk source for the 3 MOH .mp3's and copied them to the appropriate location. Still MOH is not working. Any other ideas? Thanks again, Phil Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. J. ___ --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 -- Arlen Nascimento ___ --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: Match a Numer - then continue with dialplan
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 6:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG So, in the event that the logic flows beyond
DG coo1_OnNet, we want to reset the caller id of say, 3254001 Doug,
DG to 3254000 Widgets Inc.
DG exten = 3254101,1,Dial(SIP/3254101,20,tr)
DG exten = 3254102,1,Dial(SIP/3254102,20,tr)
DG exten = 3254103,1,Dial(SIP/3254103,20,tr)
[coo1_CallStart]
include = coo1_OnNet
You want something which executes here, if coo1_OnNet didn't match?
exten = _.,1,Set(CALLERID(all)=Widgets Inc 3254001)
will do that.
If you then want to continue in priority 1 instead of 2, you just do
exten = _.,n,Goto(coo1_CallStart2,${EXTEN},1)
[coo1_CallStart2]
include = syst_OnNet
include = syst_OffNet
That won't do it. Processing will continue in the current extension priority. I
need it to continue looking for an extension to match against. Once Asterisk
has matched the dialled number against an extension in the dialplan, your stuck
in an extension you can never get out and get Asterisk to go back to
looking for extensions to match against.
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RE: [asterisk-users] AgentCallbackLogin() deprecated in 1.4
Yes, we have issues with this application being removed as well. In my opinion, it's a loss of functionality. -Original Message- From: Markus Bönke [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 6:40 AM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: [asterisk-users] AgentCallbackLogin() deprecated in 1.4 Hello all, I've seen that the application AgentCallbackLogin()has been set to deprecated in version 1.4. So I've done some tests based on the tutorial queues-with-callback-members.txt coming with version 1.4. What's not clear for me is what is happening to agents.conf, it seems that it's no longer needed, and I have to define my agents using variables in extensions.ael. The other thing is, that show agents doesn't show me which agents are logged in and if I use show queue I can see local channels attached to a queue but no agents. For my point of view there is some functionality lost with the new concept. If I want to program a realtime display to show agentstates in queues based on the output from show queue, what's the concept to map agents to the local channels? How can I configure agents in future? Any comments regarding that topic are appreciated. Thanks and Regards Markus ___ --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] Re: Match a Numer - then continue with dialplan
I haven't really been following this thread but doesn't the following
snipet kinda do this
[out-international]
exten = _011,1,goto(process-international,s,1)
[process-international]
exten = s,1,playback(international-call)
exten = s,n,playback(please-enter-the)
exten = s,n,read(number,number)
exten = s,n,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = s,n,Macro(failann,${DIALSTATUS})
This matches 011 then could do any number of things. Here I just goto,
then it looks for more numbers (the announcement is optional) and then
dials them.
Maybe not what you are looking for but it is an example of Asterisk
matching an extension and then going on to take more digits that then
branch based on other digits. Here the 011 is prepended to the final
number.
BTW - what is a numer?
Doug
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Re: [asterisk-users] Thomson ST2030S and BLF
Hi Albertore, As you can guess, my previous reply was mostly based on a general discussion with Thomson marketing and support teams. They developped an Asterisk patch to support one key call pickup but never reached a decision about the way to have this patch maintained as this patch modifiez chan_sip.c and chan_sip.c is updated every day. AFAIK, as call pickup related standards are not stabilised yet (it seems so), I asked them if it could be possible to include an Asterisk independant workaround. I didn't get any usable reply yet, beside usual maybe with next release. From http://bugs.digium.com/view.php?id=5014, I don't think one key call pickup is going to appear anytime soon with Asterisk. Regards ___ --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: Match a Numer - then continue with dialplan
-Original Message-
From: Doug Crompton [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 8:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: Match a Numer - then continue with
dialplan
I haven't really been following this thread but doesn't the following
snipet kinda do this
[out-international]
exten = _011,1,goto(process-international,s,1)
[process-international]
exten = s,1,playback(international-call)
exten = s,n,playback(please-enter-the)
exten = s,n,read(number,number)
exten = s,n,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = s,n,Macro(failann,${DIALSTATUS})
This matches 011 then could do any number of things. Here I just goto,
then it looks for more numbers (the announcement is optional) and then
dials them.
Maybe not what you are looking for but it is an example of Asterisk
matching an extension and then going on to take more digits that then
branch based on other digits. Here the 011 is prepended to the final
number.
Don't get offended Doug, but I get really frustrated when I try to explain what
I am trying to do with Asterisk, and people don't seem to quite get it. Your
about the 4th person who's replied to this post, and hasn't quite grasped my
question. :) --- smiley.. see...we're all cool.
I don't want Asterisk to go on to ask for more digits. I want to do a very
simple thing. I want to set a variable when call flow continues beyond a
certain point (without asking the user for more digits), and then continue on,
and use that variable later. It's a very simple thing, I can't work out why
Asterisk doesn't let me do that.
Surely other people have hit the situation where they first check extensions
within a company, and then if there's no match, you glue all the other
companies dialplans together with this one. At that point, when one company
dials another, the caller id that's sent should be the company caller id, not
the caller id of the individual extension. It's a very common business
requirement... at least that's what my boss, who has spend many years
installing TDM pbx's tells me.
BTW - what is a numer?
A numer is a spelling mistake. I was going to change the title, but it would
have broken the thread.
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Re: [asterisk-users] Polycom ring backs and CID
Hi - We'll still need to see more of your dialplan. By your description, it looks like the call is failing because the Dial() times out. Take two... My calls are NOT FAILING. Never have so let me restate... Call comes in receptionist answers. For some ungodly reason this client does not want voicemail, so when a call is xferred, the call goes through fine, if no one answers it rings back to the receptionist *SUCCESSFULLY*. However, what the client is complaining about is, it sounds idiotic to repeat the company mantra Thank you for calling Foobar Co. how can I xfer your call to a caller they just answered but failed to be xferred successfully. Before someone asks why identify the caller ID this customer also (for some ungodly reason) only wants his CID showing up in and out. (Don't ask) So again: Call comes in -- Receptionist (How can I direct your call) Receptionist -- Transfers to extension Extension -- No answer -- Back to receptionist Receptionist (same call) -- Thank you for calling Foobar Easier to comprehend? Please, just show me your dialplan. It is extensions.conf. I can't help you without seeing that. - Noah ___ --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 sip account format
I have 2 sip accounts with name 1234 and abcd respectively. Account abcd can make call to 1234 but not visa versa. When I change account abcd to 1abcd, both of them can make call to each others. In the case, the format of sip account should be start with number. I wonder whether we can use a sip account using only characters. Anyone can tell me how? Is it possible? ___ --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: TR: TR:
?
KOUCH RACHID a crit:
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, December 20, 2006 6:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
"DG" == Douglas Garstang [EMAIL PROTECTED] writes:
DG So, in the event that the logic flows beyond
DG coo1_OnNet, we want to reset the caller id of say, 3254001 Doug,
DG to 3254000 Widgets Inc.
DG exten = 3254101,1,Dial(SIP/3254101,20,tr)
DG exten = 3254102,1,Dial(SIP/3254102,20,tr)
DG exten = 3254103,1,Dial(SIP/3254103,20,tr)
[coo1_CallStart]
include = coo1_OnNet
You want something which executes here, if coo1_OnNet didn't match?
exten = _.,1,Set(CALLERID(all)=Widgets Inc 3254001)
will do that.
If you then want to continue in priority 1 instead of 2, you just do
exten = _.,n,Goto(coo1_CallStart2,${EXTEN},1)
[coo1_CallStart2]
include = syst_OnNet
include = syst_OffNet
That won't do it. Processing will continue in the current extension priority. I need it to continue looking for an extension to match against. Once Asterisk has matched the dialled number against an extension in the dialplan, your stuck in an extension you can never get out and get Asterisk to go back to looking for extensions to match against.
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RE: [asterisk-users] Match a Numer - then continue with dialplan
[snip]
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = syst_OffNet
Instead of including your system-wide logic for offnet calling,
introduce a per-company offnet and include that instead:
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = coo1_OffNet
[coo1_OffNet]
exten = _X.,1,Set(CALLERID(NUM)=3254000)
exten = _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
exten = _X.,3,Goto(syst_OffNet,${EXTEN},1)
Bradley, If I do this, then I can no longer continue with
further extensions in my dialplan as Asterisk has already
matched a number. I still need to check black/white lists,
set pic codes and rate centers, 4 digit extensions etc within
the company context. I just need to set the caller id and
then move on. If I goto over to ${EXTEN} within syst_OffNet,
I'd have to put ALL this logic within that extension, which
would mean potentiall several hundred priorities. Asterisk
really does need a way to match a number, execute some code,
and then go back to looking for extensions.
Why not do something like this (in pseudo dialplan):
matching and initial dialplan stuff
decide the outgoing callerid should change
SetVar(outgoing_callerid=1234567)
continue with dialplan and do all kinds of weird things
Set(CALLERID(NUM)=${outgoing_callerid})
Dial(outgoing destination)
This will not screw up your extesnions matching, but you will
need to check that outgoing_callerid has been filled before setting
callerid (or make sure it is always filled with something sensible).
Check the variables page in the wiki on exact syntax ;-)
--
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+31 (0)23 70743422132 HZ Hoofddorp
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Re: [asterisk-users] No music on hold?
On Tue, Dec 19, 2006 at 10:04:23PM -0500, Jerry wrote: Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. Why? The OP is looking to play MP3s, and unless I misunderstood the instructions on the Wiki, addons is required (format_mp3) to play MP3's on 1.2.x. Is that not the case? J. ___ --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] spandsp 0.0. 3 RxFax fax =?ISO-8859-1?Q?_reception c rashes bristuffed_asterisk_1=2E2=2E13_[?= Virusgeprüft]
[EMAIL PROTECTED] wrote: sure in an small office you can use iaxmodem/hylafax to receive faxes - we use it for sending faxes, but would you try to set up about 100 iaxmodems inside hylafax if you can handle it directly inside asterisk with rx_fax and a small script ? Yes, I would, actually use 100 iaxmodems instead. 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
RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message-
From: Andreas Sikkema [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
[snip]
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = syst_OffNet
Instead of including your system-wide logic for offnet calling,
introduce a per-company offnet and include that instead:
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = coo1_OffNet
[coo1_OffNet]
exten = _X.,1,Set(CALLERID(NUM)=3254000)
exten = _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
exten = _X.,3,Goto(syst_OffNet,${EXTEN},1)
Bradley, If I do this, then I can no longer continue with
further extensions in my dialplan as Asterisk has already
matched a number. I still need to check black/white lists,
set pic codes and rate centers, 4 digit extensions etc within
the company context. I just need to set the caller id and
then move on. If I goto over to ${EXTEN} within syst_OffNet,
I'd have to put ALL this logic within that extension, which
would mean potentiall several hundred priorities. Asterisk
really does need a way to match a number, execute some code,
and then go back to looking for extensions.
Why not do something like this (in pseudo dialplan):
matching and initial dialplan stuff
decide the outgoing callerid should change
Ok...
SetVar(outgoing_callerid=1234567)
Bzzt. In order to call SetVar, I have to match the extension dialled. When that
happens, there is NO WAY to continue searching the dialplan after that point
for another extension to match.
continue with dialplan and do all kinds of weird things
Can only continue within the current proirity... which means that at this
point, all my further logic has to be coded as priorities in the extension that
called SetVar. Seeing as though I have several dozen more contexts to include,
this isn't feesible.
Set(CALLERID(NUM)=${outgoing_callerid})
Dial(outgoing destination)
This will not screw up your extesnions matching, but you will
need to check that outgoing_callerid has been filled before setting
callerid (or make sure it is always filled with something sensible).
Thanks for trying.
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Re: [asterisk-users] sip help for newbie
I'm not sure. I'm a linux newb and this is just running on a server I have hosted somewhere. I do have control of the box, just not sure what's open or how to open them. On 12/13/06, Dovid B [EMAIL PROTECTED] wrote: You need port 5060 as well as 1-2 UDP open to the server. Also is the server behind NAT at all ? - Original Message - *From:* blackwater dev [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, December 13, 2006 5:14 AM *Subject:* Re: [asterisk-users] sip help for newbie Thanks for the info, I've gone through the tutorial and followed it and asterisk is running but I just can't seem to log in. The xten phone just tells me connection timed out. I'm simply running asterisk on a webserver that is also running apache and service content. I simply pinged the box to get the ip to plug into the softphone. Do I need to open a port or something? On 12/12/06, Forrest Beck [EMAIL PROTECTED] wrote: www.asteriskguru.com On 12/12/06, blackwater dev [EMAIL PROTECTED] wrote: Does anyone know of any good step by step tutorials on getting sip set up? I have asterisk installed but can't seem to figure out how to get an account set up and connect from my xTen phone so I can try the demo. The tutorials I read online seem to go into voicepulse stuff and all and I don't have an account there so am a bit lost. 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 ___ --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 ___ --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] AgentCallbackLogin() deprecated in 1.4
-Original Message- From: Gavin Hamill [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 7:10 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AgentCallbackLogin() deprecated in 1.4 On Wed, 20 Dec 2006 14:39:42 +0100 Markus Bönke [EMAIL PROTECTED] wrote: Hello all, The other thing is, that show agents doesn't show me which agents are logged in and if I use show queue I can see local channels attached to a queue but no agents. For my point of view there is some functionality lost with the new concept. Funny. I said the same thing in this list about 2 months ago and I got told I was nuts. ___ --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] No music on hold?
I already posted about this, but contrary to what is stated on the Wiki, mpg123 is required (at least in 1.2.x) if you wish to use mp3's for your MoH. I decided to go this route: http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it -- Kevin Trumbull -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Sent: Wednesday, December 20, 2006 8:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No music on hold? On Tue, Dec 19, 2006 at 10:04:23PM -0500, Jerry wrote: Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. Why? The OP is looking to play MP3s, and unless I misunderstood the instructions on the Wiki, addons is required (format_mp3) to play MP3's on 1.2.x. Is that not the case? J. ___ --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] question about sip account format
On 12/20/06, Rilawich Ango [EMAIL PROTECTED] wrote: I have 2 sip accounts with name 1234 and abcd respectively. Account abcd can make call to 1234 but not visa versa. When I change account abcd to 1abcd, both of them can make call to each others. In the case, the format of sip account should be start with number. I wonder whether we can use a sip account using only characters. Anyone can tell me how? Is it possible? Yes, as I recall you cen use alpha, numeric or both. I am guessing you do not have the proper character matching in whatever context the calls are going to in extensions.conf. I think something like this should work: exten = _[A-Za-z0-9].,1,Answer Regards, David ___ --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] Polycom ring backs and CID
(FYI client did not want VM... Don't ask...)
[general]
static=yes
writeprotect=no
[incoming]
exten = s,1,NoOP(${EXTEN})
exten = s,2,Goto(main-aa,s,1)
exten = 13015550835,1,Goto(main-aa,s,1)
exten = 3015550835,1,Goto(main-aa,s,1)
exten = 5550835,1,Goto(main-aa,s,1)
exten = 0835,1,Goto(main-aa,s,1)
exten = 13015551293,1,Goto(main-aa,s,1)
exten = 3015551293,1,Goto(main-aa,s,1)
exten = 5551293,1,Goto(main-aa,s,1)
exten = 1293,1,Goto(main-aa,s,1)
exten = 13015551470,1,Goto(main-aa,s,1)
exten = 3015551470,1,Goto(main-aa,s,1)
exten = 5551470,1,Goto(main-aa,s,1)
exten = 1470,1,Goto(main-aa,s,1)
exten = 13015551743,1,Goto(main-aa,s,1)
exten = 3015551743,1,Goto(main-aa,s,1)
exten = 5551743,1,Goto(main-aa,s,1)
exten = 1743,1,Goto(main-aa,s,1)
exten = 13015552196,1,Goto(main-aa,s,1)
exten = 3015552196,1,Goto(main-aa,s,1)
exten = 5552196,1,Goto(main-aa,s,1)
exten = 2196,1,Goto(main-aa,s,1)
exten = 13015558549,1,Goto(main-aa,s,1)
exten = 8605558549,1,Goto(main-aa,s,1)
exten = 5558549,1,Goto(main-aa,s,1)
exten = 8549,1,Goto(main-aa,s,1)
exten = 1301001,1,Goto(main-aa,s,1)
exten = 301001,1,Goto(main-aa,s,1)
exten = 001,1,Goto(main-aa,s,1)
exten = 5001,1,Goto(main-aa,s,1)
exten = 1301002,1,Goto(main-aa,s,1)
exten = 301002,1,Goto(main-aa,s,1)
exten = 795002,1,Goto(main-aa,s,1)
exten = 5002,1,Goto(main-aa,s,1)
exten = 1301003,1,Goto(main-aa,s,1)
exten = 301003,1,Goto(main-aa,s,1)
exten = 003,1,Goto(main-aa,s,1)
exten = 5003,1,Goto(main-aa,s,1)
exten = 1301004,1,Goto(main-aa,s,1)
exten = 301004,1,Goto(main-aa,s,1)
exten = 004,1,Goto(main-aa,s,1)
exten = 5004,1,Goto(main-aa,s,1)
exten = 1301005,1,Goto(main-aa,s,1)
exten = 301005,1,Goto(main-aa,s,1)
exten = 005,1,Goto(main-aa,s,1)
exten = 5005,1,Goto(main-aa,s,1)
exten = 1301006,1,Goto(main-aa,s,1)
exten = 301006,1,Goto(main-aa,s,1)
exten = 006,1,Goto(main-aa,s,1)
exten = 5006,1,Goto(main-aa,s,1)
exten = 1301010,1,Goto(main-aa,s,1)
exten = 301010,1,Goto(main-aa,s,1)
exten = 010,1,Goto(main-aa,s,1)
exten = 5010,1,Goto(main-aa,s,1)
exten = 1301012,1,Goto(main-aa,s,1)
exten = 301012,1,Goto(main-aa,s,1)
exten = 012,1,Goto(main-aa,s,1)
exten = 5012,1,Goto(main-aa,s,1)
exten = 1301032,1,Goto(main-aa,s,1)
exten = 301032,1,Goto(main-aa,s,1)
exten = 032,1,Goto(main-aa,s,1)
exten = 5032,1,Goto(main-aa,s,1)
[main-aa]
exten = s,1,GotoIfTime(18:00-7:00|mon-fri|*|*|*?main-night-aa,s,1)
exten = s,2,GotoIfTime(*|sat-sun|*|*|*?main-night-aa,s,1)
exten = s,3,Dial(SIP/100SIP/101|30|tr)
exten = s,4,Goto(main-night-aa,s,1)
exten = s,104,Goto(main-night-aa,s,1)
[main-night-aa]
exten = s,1,Voicemail([EMAIL PROTECTED])
exten = s,2,Goto(main-aa,s,1)
[outbound]
exten = 00,1,VoicemailMain([EMAIL PROTECTED])
exten = a,1,VoicemailMain([EMAIL PROTECTED])
include = incoming
include = internal
include = parkedcalls
exten = 911,1,Set(CALLERID(number)=13015558549)
exten = 911,2,Goto(to_the_netherworld,${EXTEN},1)
exten = 411,1,Set(CALLERID(number)=13015558549)
exten = 411,2,Goto(to_the_netherworld,${EXTEN},1)
exten = _011.,1,Set(CALLERID(number)=13015558549)
exten = _011.,2,Goto(to_the_netherworld,${EXTEN},1)
exten = _1NXXNXX,1,Set(CALLERID(number)=13015558549)
exten = _1NXXNXX,2,Goto(to_the_netherworld,${EXTEN},1)
exten = _NXXNXX,1,Set(CALLERID(number)=13015558549)
exten = _NXXNXX,2,Goto(to_the_netherworld,${EXTEN},1)
exten = _NXX,1,Set(CALLERID(number)=13015558549)
exten = _NXX,2,Goto(to_the_netherworld,${EXTEN},1)
[to_the_netherworld]
exten = _X.,1,ChanIsAvail(Zap/1Zap/2Zap/3Zap/4)
exten = _X.,2,Cut(thechannel=AVAILCHAN,,1)
exten = _X.,3,Dial(${thechannel}/${EXTEN})
exten = _X.,4,Hangup
;exten = 911,1,Dial(Zap/g1/911)
;exten = 911,2,Dial(Zap/g1/911)
;exten = 411,1,Dial(SIP/[EMAIL PROTECTED])
;exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
;exten = _1800.,1,Dial(Zap/g1/${EXTEN})
;exten = _1866.,1,Dial(Zap/g1/${EXTEN})
;exten = _1877.,1,Dial(Zap/g1/${EXTEN})
;exten = _1888.,1,Dial(Zap/g1/${EXTEN})
;exten = _1301NXX,1,Dial(Zap/g1/${EXTEN},40,tr)
;exten = _1301NXX,2,Dial(SIP/[EMAIL PROTECTED],40,tr)
;exten = _860NXX,1,Dial(Zap/g1/${EXTEN},40,tr)
;exten = _860NXX,2,Dial(SIP/[EMAIL PROTECTED],40,tr)
;exten = _1NXXNXX,2,Dial(SIP/[EMAIL PROTECTED],40,tr)
;exten = _1NXXNXX,1,Dial(Zap/g1/${EXTEN},30,tr)
;exten = _1NXXNXX,3,Hangup
;exten = _NXXNXX,2,Dial(SIP/[EMAIL PROTECTED],40,tr)
;exten = _NXXNXX,1,Dial(Zap/g1/1${EXTEN},30,tr)
;exten = _NXXNXX,3,Hangup
;exten = _NXX,1,Dial(Zap/g1/1301${EXTEN},30,tr)
;exten = _NXX,2,Dial(SIP/[EMAIL PROTECTED],30,tr)
;exten = _NXX,3,Hangup
[parkedcalls]
; Parking
exten = 700,1,NoOp()
exten =
700,n,ParkAndAnnounce(call:ha/on:PARKED|105|SIP/7${BLINDTRANSFER:7:2}|default,71${BLINDTRANSFER:5:2},1)
exten = 700,hint,Local/7
[internal]
exten = 100,1,Dial(SIP/100|30|tr)
exten = 100,3,Voicemail([EMAIL PROTECTED])
exten = 100,2,
exten = 100,102,Goto(main-aa,s,1)
exten =
RE: [asterisk-users] Need quality toll free 800 number over IAX?
www.Kall8.com Arick Davis _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Blunt Sent: Wednesday, December 20, 2006 5:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Need quality toll free 800 number over IAX? Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd ___ --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] Match a Numer - then continue with dialplan
Bzzt. In order to call SetVar, I have to match the extension dialled. When that happens, there is NO WAY to continue searching the dialplan after that point for another extension to match. You can't use a generic extension and search a database table for $EXTEN - callerid relation and then set it? Your diallingplan is _so_ different to what we do, yet what you want to do is pretty much the same to what we do all the time. But our Asterisk boxes have _no_ sip CPE's registered to them and our diallingplan is littered with database lookups. We have no static stuff in our dialingplan. And we have quite a number of users. But no queues etc. -- Andreas SikkemaBBeyond Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] Match a Numer - then continue with dialplan
On 20/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: Bzzt. In order to call SetVar, I have to match the extension dialled. When that happens, there is NO WAY to continue searching the dialplan after that point for another extension to match. Can you not use either Goto or the Local channel, maybe a combination, to restart the dialplan with your variable set? (Might need a _ or two on the variable name to get it to survive) Peter -- Peter Bowyer Email: [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
Re: [asterisk-users] Echo problem
Steve Davies [EMAIL PROTECTED] writes: Scott Gifford [EMAIL PROTECTED] writes: [...] 1.5 to 2 seconds. That is a HUGE delay. echo delay is normally measured in tens or perhaps hundreds of milliseconds, and you are unlikely to find a software EC that can deal with a 1.5 to 2 second delay! This sounds as if there is something very broken in the voice network, causing huge amounts of delay. As suggested above, check the intermediate switch. What's interesting is the lines come in via 2 PRI lines, and most calls go out via analog lines to people's desks and a voicemail system. These lines all work fine. So the problem likely isn't in the PSTN and isn't an inherent flaw with the switch, though it could be the T1 card connected to our Asterisk server or its configuration. It seems the problem is either on the Tadiran switch or the Asterisk server. Unfortunately we don't have a good way to determine which, since we don't have another switch to try, or another device to replace the Digium server. [snip] We have done loopback tests with the Digium card with a loop plug in it. What were the results? Oh, sorry, I should have said: These tests were successful. Scott. ___ --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] No music on hold?
On Wed, Dec 20, 2006 at 08:30:27AM -0800, Kevin Trumbull wrote: I already posted about this, but contrary to what is stated on the Wiki, mpg123 is required (at least in 1.2.x) if you wish to use mp3's for your MoH. I decided to go this route: http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it If you're not streaming the MP# from an external source, converting it off-line will always be cheaper. And it may even actually save you disk space, because mp3 files have a much higher quality than Asterisk requires. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] spandsp 0.0.3 Rx Fax fax =?ISO-8859-1?Q?_reception crashes bri stuffed_asterisk_1=2E2=2E13_[?= Virusge prüft]
Does IAXmodem allows you to receive faxes with any extensions (auto-detecting incoming faxes). You just let Asterisk do the fax detection for you, and when it hears CNG, send it to the fax extension, and your fax extension would just Dial() one of the IAXmodems (using IAX) [EMAIL PROTECTED] wrote: sure in an small office you can use iaxmodem/hylafax to receive faxes - we use it for sending faxes, but would you try to set up about 100 iaxmodems inside hylafax if you can handle it directly inside asterisk with rx_fax and a small script ? Yes, I would, actually use 100 iaxmodems instead. Lee. I second that. After struggling with rxfax (which was total cake to set up, but reception reliability in my specific installation was poor) I bit the bullet and put in a separate Hylafax server connected to my Asterisk box with a crossover cable, rolled up my sleeves, and stated making IAXmodems - 1 per user. I am at over 200 IAXmodem's, and my failure rate on faxes plummeted to about .8 % - more than comparable to a regular fax machine. AFAIC, Hylafax + IAXmodem is the way to go for anything serious, unless we are talking about thousands of users and thousands of faxes per day. I don't even know what could be scaled to that scenario and not be unmanageable. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming Lines Confusion
First off, please, for the love of God, don't cremate me, if I should already know the answer to this! I've installed a small setup for an office who wanted to be able to talk to each other instead of having to rely on MSN to communicate. Weird request, I know, but hey, we do what we need to do to get paid. I installed soft phones, gave everyone an extension, and bingo, they can call and talk over their PCs happy as hell. Which brings me to my problem - they loved the system so much, that now, they want it for ALL their calls, that is their calls that involve the real world. ATM, they all have separate independent land lines, which is why they had a problem in the first place, large bills for calling each other, now they want a VoIP solution, that would have calls coming in over their broadband connection, and automatically route to each of their phones, depending on which line has called them. I just got out of a meeting with them, and what they want, goes as such... Bob has a VoIP number 020-xxx-xxx - when this number rings, the box answers the call, plays some music, while it waits for Bob to answer. This call should only go to Bobs extension. If he's not there, it routes the call to his voicemail Mary has a different number. When this number rings, it gets routed straight to her extension, in the same manner as Bobs, but if she's not available, it looks for who is, and rings their phones, and if no one answers, then goes to voice mail. Basically, there are 2 types of behaviours that they would like on their lines. My problem, is how to implement it! I'm an asterisk virgin, and getting them to be able to talk to each other across their office network and 12 extensions, took the best part of 2 hours - I don't want to have to spend a whole day working on this one. The VoIP numbers have already been purchased, and are ready to go - i just need to configure it all - Can it be done!? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need quality toll free 800 number over IAX?
I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Never used them but the rates seems ok : http://www.les.net/ ___ --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: Newbie Questions - Grandstorm phones?
I have been using an approach such as this but am looking for something
else because of some limitations it has. The phone thinks it dialed,
and was connected to 011 (which it was)
As such, that will be stored in the phones dial history (redial if
nothing else).
I'm not even certain what I want is possible, which is why I'm asking
the list.
Thank you for your help once again though.
- Anthony Kepler
[EMAIL PROTECTED] | SIP/EMail
Doug Crompton wrote:
Well that is certainly an option but not all phones would have a send key
especially if you are using analog phones. I guess the # keys
functions in
that way on many of those.
I still like my wired phones to work like they use to. You dial a
number
and it executes the call immediately.
Ok I came up with one that I think would work, maybe needs some
refinement
[out-international]
exten = _011,1,goto(process-international,s,1)
[process-international]
exten = s,1,read(number)
exten = s,2,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = s,3,Macro(failann,${DIALSTATUS})
This accepts the 011 prefix and then any number of following digits.
Terminator is timeout period OR # key to send. Change obviously for your
provider.
The read command has many options including saying a file. You could for
instance hear Country Code after dialing 011. This would clue you into
the fact that you were dialing and international call. There are also
digit limits and timeouts that can be set.
So if you use early dial this would be the only rule that would require a
wait or # key to send. I could certainly live with that.
Can anyone supply some international test numbers??? Say in the UK or
Germany or wherever outside the US.
Doug
___
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RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message- From: Andreas Sikkema [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Match a Numer - then continue with dialplan Bzzt. In order to call SetVar, I have to match the extension dialled. When that happens, there is NO WAY to continue searching the dialplan after that point for another extension to match. You can't use a generic extension and search a database table for $EXTEN - callerid relation and then set it? Yes, I can do that. However, in order to do all that, I have to match an extension first. Same problem as before. Your diallingplan is _so_ different to what we do, yet what you want to do is pretty much the same to what we do all the time. I dunno about that. I think we're the only crazy ones offering company masked caller id, or else there'd be lots of people asking how to do it. But our Asterisk boxes have _no_ sip CPE's registered to them and our diallingplan is littered with database lookups. We have no static stuff in our dialingplan. And we have quite a number of users. If you have no statuc stuff in your dialplan, how do you use the 'include =' statement? We don't have users... we have companies. It's a hosted IPT service... and to make the problem even more insane, each company has multiple levels of organisational structure. ___ --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] Match a Numer - then continue with dialplan
Douglas Garstang wrote: Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Match dialed digits of 668 exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,More Stuff Here/ ___ --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] Match a Numer - then continue with dialplan
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
On 20/12/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Bzzt. In order to call SetVar, I have to match the
extension dialled. When that happens, there is NO WAY to
continue searching the dialplan after that point for another
extension to match.
Can you not use either Goto or the Local channel, maybe a combination,
to restart the dialplan with your variable set? (Might need a _ or two
on the variable name to get it to survive)
The Goto() command requires priority (extension, context). I'd need to jump to
a context, without supplying an extension, which it won't accept. If I pass a
priority, we're right back at square one, we're I'm stuck in a priority and
can't get back to an extension. I tried putting a Dial(Local/${EXTEN}), but the
problem was that Asterisk then went into an infinite when I tried to include
all the company contexts together (because it was matching the Dial/Local
again).
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Re: [asterisk-users] Match a Numer - then continue with dialplan
Douglas Garstang wrote: -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 19, 2006 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Match a Numer - then continue with dialplan On 12/19/06, Douglas Garstang [EMAIL PROTECTED] wrote: Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug. Have you tried using the SetVar cmd? I haven't tried it but it sounds like it might work for this. http://www.voip-info.org/wiki/view/Asterisk+variables Regards, David David, If I call setvar, my variable will be set, but dialplan processing will stop... Then something else is wrong. SetVar will not stop dialplan processing. In 1.4, I believe SetVar() was removed. Check upgrade.txt. Use Set in 1.4 instead. ___ --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: Match a Numer - then continue with, dialplan
I think you're making it far too difficult.
What I do is something like this:
[outgoing]
include = internal
include = longdistance
;Always include internal first, as matches from the first include
;will be used first. This allows you to make sure your internal
;extensions don't go out your trunks.
[longdistance]
ignorepat = 9;
include = default; already included from local, but putting here for
clarity
include = local;
exten = _91XXX,1,Macro(trunkout,${EXTEN}) ;Medium Distance
exten = _91XX,1,Macro(trunkout,${EXTEN}) ;Long Distance
Then, I have:
[macro-trunkout]
exten = s,1,Set(cname=${DB(showname/${CALLERIDNUM})});
exten = s,n,Set(cnum=${DB(shownum/${CALLERIDNUM})});
exten = s,n,GotoIf($[foo${cnum} = foo]?6); //if calling from ZAP
channel that set caller ID already
exten = s,n,Set(CALLERID(name)=${cname}|a);
exten = s,n,Set(CALLERID(number)=${cnum}|a);
exten = s,n,Dial(${TRUNK}/${ARG1:${TRUNKMSD}});
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CONGESTION,1,Congestion(30)
exten = s-CONGESTION,2,Hangup
exten = s-CANCEL,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-BUSY,2,Hangup
Why is this important? It's not. But it is fundamentally different from
what you're asking. You want to match a partial extension dialed and
then continue appending digits. What you really need to do is wait for
the whole number, then decide what kind of number it is, do the
processing, and send it on its way. It's just a slight change in the way
you're thinking, because you understand that there's a class of numbers
to treat differently. And that's OK. Just don't do anything with it
until the whole extension has been entered!
You'll notice that, anything not going through the trunkout macro
doesn't get tweaked, and anything that goes through there will read from
the database. I could just as easily set a single value, but I have some
users that I want to go out as themselves, and different departments
that have a general number, etc. I found the Asterisk Database to be the
easiest to tweak, as I have some scripts to allow admins to change the
effective CallerID on the fly.
I hope this helps! Asterisk can do what you're asking, and it does every
day.
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[asterisk-users] Can't make outgoing calls (T100P)
Hi there, I have a new box setup using the latest version of FreePBX and the latest SVN of Asterisk 1.2 as of yesterday. Incoming calls from our PRI work fine. However, outgoing calls gives me the operator saying The call cannot be completed as dialed after two rings. Here's an outgoing call from extension 271: -- Executing Set(SIP/271-09f61dc0, OUTNUM=7883229) in new stack -- Executing Set(SIP/271-09f61dc0, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/271-09f61dc0, 0?customtrunk) in new stack -- Executing Dial(SIP/271-09f61dc0, ZAP/g0/7883229|300|) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/7883229 -- Zap/1-1 is proceeding passing it to SIP/271-09f61dc0 -- PROGRESS with cause code 28 received -- Zap/1-1 is making progress passing it to SIP/271-09f61dc0 -- Hungup 'Zap/1-1' I've tried to find out what cause code 28 is with no luck. zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us zapata.conf: [channels] language=en #include zapata_additional.conf context=from-pstn switchtype=national pridialplan=national signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes callerid=asreceived echocancelwhenbridged=no echotraining=800 group=0 channel=1-10 We have 10 enabled lines from this PRI. Any help/suggestions are appreciated. Regards, -Darren ___ --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] spandsp 0.0. 3 RxFax fax =?ISO-8859-1?Q?_reception c rashes bristuffed_asterisk_1=2E2=2E13_[?= Virusgeprüft]
Colin Anderson wrote: AFAIC, Hylafax + IAXmodem is the way to go for anything serious, unless we are talking about thousands of users and thousands of faxes per day. I don't even know what could be scaled to that scenario and not be unmanageable. For the thousands and thousands scenario you could very well also still use HylaFAX (well, HylaFAX+ would probably suit it better) and IAXmodem. However, you'd certainly take advantage of DID and Caller*ID support available in both HylaFAX and IAXmodem instead of using a one-to-one modem-to-user mapping such as you appear to be doing. 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
RE: [asterisk-users] AstManProxy - Manager
I don't use many of the features of astmanproxy but it does work. I use it to capture events from several servers. Some of these are running the 1.4 beta releases. -Jonahtan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, December 20, 2006 9:21 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AstManProxy - Manager On Wed, Dec 20, 2006 at 02:57:17PM +0100, Olivier wrote: Hi, Is AstManProxy an alive project ? It seems to me that no development are ongoing. Will AstManProxy comply with Asterisk 1.4 ? Last release seems to be from 3 monthes ago. 1.4 has not been released yet, as you recall. Anyway, latest astmanproxy seems to have a basic support for the manager over HTTP protocol of 1.4. But maybe this is just me reading the docs wrong. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Match a Numer - then continue with dialplan
-Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Match a Numer - then continue with dialplan Douglas Garstang wrote: Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Match dialed digits of 668 exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,More Stuff Here/ Ugh. 'More Stuff Here' isn't what I need Eric. I need to continue the dialplan. I need do be able to continue to search for extensions. All I want to do is set the callerid, so that later on, when we find a match, the extension can be dialled with the new caller id already set. This ain't gonna work... exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,include = blacklist exten = 669,3,include = blacklist exten = 669,3,include = blacklist exten = 669,3,include = blacklist ___ --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] Match a Numer - then continue with dialplan
-Original Message- From: Douglas Garstang Sent: Wednesday, December 20, 2006 10:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Match a Numer - then continue with dialplan -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Match a Numer - then continue with dialplan Douglas Garstang wrote: Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Match dialed digits of 668 exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,More Stuff Here/ Ugh. 'More Stuff Here' isn't what I need Eric. I need to continue the dialplan. I need do be able to continue to search for extensions. All I want to do is set the callerid, so that later on, when we find a match, the extension can be dialled with the new caller id already set. This ain't gonna work... exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,include = blacklist exten = 669,3,include = blacklist exten = 669,3,include = blacklist exten = 669,3,include = blacklist Dang it. My fat fingers posted too soon by mistake. As I was trying to say, This obviously won't work... exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,include = blacklist exten = 669,4,include = whitelist exten = 669,5,include = PIC_Code_Insertion exten = 669,6,include = Rate_Center_Insertion exten = 669,7,include = Findme/Followme ___ --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] Match a Numer - then continue with dialplan
-Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Match a Numer - then continue with dialplan Douglas Garstang wrote: -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 19, 2006 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Match a Numer - then continue with dialplan On 12/19/06, Douglas Garstang [EMAIL PROTECTED] wrote: Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug. Have you tried using the SetVar cmd? I haven't tried it but it sounds like it might work for this. http://www.voip-info.org/wiki/view/Asterisk+variables Regards, David David, If I call setvar, my variable will be set, but dialplan processing will stop... Then something else is wrong. SetVar will not stop dialplan processing. In 1.4, I believe SetVar() was removed. Check upgrade.txt. Use Set in 1.4 instead. I was not clear. EXTENSION processing will stop. Once you've matched an extension, and your logic is running through priorities in an extension, you no longer have the ability to search for another extension to match against. That's what I need to do. Again, when control flows beyond a certain point, ie when all calls are now known to be extra-company, we need to set the callerid to the external company id... so that later on when we dial, the caller id presented to person in the other company is correct. ___ --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] Polycom ring backs and CID
Change step 2 on your internal extensions to do whatever you want to do
(change the ringer, callID, whatever) then go to main-aa,s,1. Or, change
step 2 to go someplace else, at somplace else, do whatever you want to do,
and then go to main-aa,s,1. The second method is easier to change if, later
on, you want to change whatever it is that you want to do.
On 12/20/06, J. Oquendo [EMAIL PROTECTED] wrote:
(FYI client did not want VM... Don't ask...)
[general]
static=yes
writeprotect=no
[incoming]
exten = s,1,NoOP(${EXTEN})
exten = s,2,Goto(main-aa,s,1)
exten = 13015550835,1,Goto(main-aa,s,1)
exten = 3015550835,1,Goto(main-aa,s,1)
exten = 5550835,1,Goto(main-aa,s,1)
exten = 0835,1,Goto(main-aa,s,1)
exten = 13015551293,1,Goto(main-aa,s,1)
exten = 3015551293,1,Goto(main-aa,s,1)
exten = 5551293,1,Goto(main-aa,s,1)
exten = 1293,1,Goto(main-aa,s,1)
exten = 13015551470,1,Goto(main-aa,s,1)
exten = 3015551470,1,Goto(main-aa,s,1)
exten = 5551470,1,Goto(main-aa,s,1)
exten = 1470,1,Goto(main-aa,s,1)
exten = 13015551743,1,Goto(main-aa,s,1)
exten = 3015551743,1,Goto(main-aa,s,1)
exten = 5551743,1,Goto(main-aa,s,1)
exten = 1743,1,Goto(main-aa,s,1)
exten = 13015552196,1,Goto(main-aa,s,1)
exten = 3015552196,1,Goto(main-aa,s,1)
exten = 5552196,1,Goto(main-aa,s,1)
exten = 2196,1,Goto(main-aa,s,1)
exten = 13015558549,1,Goto(main-aa,s,1)
exten = 8605558549,1,Goto(main-aa,s,1)
exten = 5558549,1,Goto(main-aa,s,1)
exten = 8549,1,Goto(main-aa,s,1)
exten = 1301001,1,Goto(main-aa,s,1)
exten = 301001,1,Goto(main-aa,s,1)
exten = 001,1,Goto(main-aa,s,1)
exten = 5001,1,Goto(main-aa,s,1)
exten = 1301002,1,Goto(main-aa,s,1)
exten = 301002,1,Goto(main-aa,s,1)
exten = 795002,1,Goto(main-aa,s,1)
exten = 5002,1,Goto(main-aa,s,1)
exten = 1301003,1,Goto(main-aa,s,1)
exten = 301003,1,Goto(main-aa,s,1)
exten = 003,1,Goto(main-aa,s,1)
exten = 5003,1,Goto(main-aa,s,1)
exten = 1301004,1,Goto(main-aa,s,1)
exten = 301004,1,Goto(main-aa,s,1)
exten = 004,1,Goto(main-aa,s,1)
exten = 5004,1,Goto(main-aa,s,1)
exten = 1301005,1,Goto(main-aa,s,1)
exten = 301005,1,Goto(main-aa,s,1)
exten = 005,1,Goto(main-aa,s,1)
exten = 5005,1,Goto(main-aa,s,1)
exten = 1301006,1,Goto(main-aa,s,1)
exten = 301006,1,Goto(main-aa,s,1)
exten = 006,1,Goto(main-aa,s,1)
exten = 5006,1,Goto(main-aa,s,1)
exten = 1301010,1,Goto(main-aa,s,1)
exten = 301010,1,Goto(main-aa,s,1)
exten = 010,1,Goto(main-aa,s,1)
exten = 5010,1,Goto(main-aa,s,1)
exten = 1301012,1,Goto(main-aa,s,1)
exten = 301012,1,Goto(main-aa,s,1)
exten = 012,1,Goto(main-aa,s,1)
exten = 5012,1,Goto(main-aa,s,1)
exten = 1301032,1,Goto(main-aa,s,1)
exten = 301032,1,Goto(main-aa,s,1)
exten = 032,1,Goto(main-aa,s,1)
exten = 5032,1,Goto(main-aa,s,1)
[main-aa]
exten = s,1,GotoIfTime(18:00-7:00|mon-fri|*|*|*?main-night-aa,s,1)
exten = s,2,GotoIfTime(*|sat-sun|*|*|*?main-night-aa,s,1)
exten = s,3,Dial(SIP/100SIP/101|30|tr)
exten = s,4,Goto(main-night-aa,s,1)
exten = s,104,Goto(main-night-aa,s,1)
[main-night-aa]
exten = s,1,Voicemail([EMAIL PROTECTED])
exten = s,2,Goto(main-aa,s,1)
[outbound]
exten = 00,1,VoicemailMain([EMAIL PROTECTED])
exten = a,1,VoicemailMain([EMAIL PROTECTED])
include = incoming
include = internal
include = parkedcalls
exten = 911,1,Set(CALLERID(number)=13015558549)
exten = 911,2,Goto(to_the_netherworld,${EXTEN},1)
exten = 411,1,Set(CALLERID(number)=13015558549)
exten = 411,2,Goto(to_the_netherworld,${EXTEN},1)
exten = _011.,1,Set(CALLERID(number)=13015558549)
exten = _011.,2,Goto(to_the_netherworld,${EXTEN},1)
exten = _1NXXNXX,1,Set(CALLERID(number)=13015558549)
exten = _1NXXNXX,2,Goto(to_the_netherworld,${EXTEN},1)
exten = _NXXNXX,1,Set(CALLERID(number)=13015558549)
exten = _NXXNXX,2,Goto(to_the_netherworld,${EXTEN},1)
exten = _NXX,1,Set(CALLERID(number)=13015558549)
exten = _NXX,2,Goto(to_the_netherworld,${EXTEN},1)
[to_the_netherworld]
exten = _X.,1,ChanIsAvail(Zap/1Zap/2Zap/3Zap/4)
exten = _X.,2,Cut(thechannel=AVAILCHAN,,1)
exten = _X.,3,Dial(${thechannel}/${EXTEN})
exten = _X.,4,Hangup
;exten = 911,1,Dial(Zap/g1/911)
;exten = 911,2,Dial(Zap/g1/911)
;exten = 411,1,Dial(SIP/[EMAIL PROTECTED])
;exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
;exten = _1800.,1,Dial(Zap/g1/${EXTEN})
;exten = _1866.,1,Dial(Zap/g1/${EXTEN})
;exten = _1877.,1,Dial(Zap/g1/${EXTEN})
;exten = _1888.,1,Dial(Zap/g1/${EXTEN})
;exten = _1301NXX,1,Dial(Zap/g1/${EXTEN},40,tr)
;exten = _1301NXX,2,Dial(SIP/[EMAIL PROTECTED],40,tr)
;exten = _860NXX,1,Dial(Zap/g1/${EXTEN},40,tr)
;exten = _860NXX,2,Dial(SIP/[EMAIL PROTECTED],40,tr)
;exten = _1NXXNXX,2,Dial(SIP/[EMAIL PROTECTED],40,tr)
;exten = _1NXXNXX,1,Dial(Zap/g1/${EXTEN},30,tr)
;exten = _1NXXNXX,3,Hangup
;exten = _NXXNXX,2,Dial(SIP/[EMAIL PROTECTED],40,tr)
;exten = _NXXNXX,1,Dial(Zap/g1/1${EXTEN},30,tr)
;exten = _NXXNXX,3,Hangup
;exten = _NXX,1,Dial(Zap/g1/1301${EXTEN},30,tr)
;exten =
[asterisk-users] Asterisk Now
I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual processor machine. The install lookups on the search for the Sata drive, since however it loads the sata_sil driver it doesn't work. Did someone knows what version of Linux is using on Asterisk Now? Thanks, Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No music on hold?
No, I didn't have m added. Should I have it added? I know I've ran Asterisk with mp3123 in the past and music worked ok. It seems when I hit the hold button on the phones, it does trigger the message saying music on hold is starting but it INSTANTLY stops. I wish it gave some details as to WHY it stops suddenly! This is driving me nuts. Phil Phil, did you add letter 'm' to your dial options?? exten = _XXX,1,Dial(SIP/XXX,60,m) Regards Arlen Nascimento On 12/20/06, Phil Finkler PhilF at iqconsultinginc.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: I installed the asterisk-addons from source and installed them. It looks like it copied format_mp3.so but I'm not sure if 1.2.14 addons are compatible with asterisk 1.2.10. Also I unpacked the asterisk source for the 3 MOH .mp3's and copied them to the appropriate location. Still MOH is not working. Any other ideas? Thanks again, Phil Heya, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see the following: -- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3' -- Stopped music on hold on IAX2/voicepulse01-3 It says music starts and then it instantly stops. Any ideas? Do you have asterisk-addons installed? That could be the issue. J. ___ --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 -- Arlen Nascimento ___ --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: Match a Numer - then continue with dialplan
In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: Don't get offended Doug, but I get really frustrated when I try to explain what I am trying to do with Asterisk, and people don't seem to quite get it. Your about the 4th person who's replied to this post, and hasn't quite grasped my question. :) --- smiley.. see...we're all cool. Perhaps its the terminology you used that is confusing people. See below: I don't want Asterisk to go on to ask for more digits. I want to do a very simple thing. I want to set a variable when call flow continues beyond a certain point (without asking the user for more digits), and then continue on, and use that variable later. It's a very simple thing, I can't work out why Asterisk doesn't let me do that. To almost all people call flow would mean executing one priority after another for a given extension. After reading and re-reading your posts trying to work out what you are trying to do, it seems to me that when *you* say call flow, you mean the act of trying to find an extension. And what your looking for is a way to do things a different points in the *search*, while it is still trying to decide on a statement to land on. Is that correct? If so, I think you need to re-think the strategy a bit. The only way a command gets executed in a dialplan is when Asterisk has matched an extension and a priority. Then once it has executed that command, it increments the priority (unless it was a Goto or something) and starts searching again. However, don't forget that it searches for matching extensions every time the priority changes. You are not locked into a particular pattern or extension number from priority 1 onwards. You can mix and match patterns with literal extensions, even across includes, e.g. [example] include = ctx31X include = ctx3XX exten = _X.,1,NoOp(this gets executed first for everything) exten = _X.,2,NoOp(this gets executed second only if ctx31X or ctx3XX didnt match) exten = _X.,3,NoOp(this gets executed third for everything) [ctx31X] exten = _31X,2,NoOp(this gets executed second for 310-319) [ctx3XX] exten = _3XX,2,NoOp(this gets executed second for 300-309 and 320-399) So you might be able to do something along these lines by being creative with priorities and includes, and setting or testing variables. Something along these lines: 1. Each company starts off in its own context, and at priority 1 in _X. it sets a variable like SRCCOMPANY to something specific to it. It includes all the destination contexts. 2. Each destination context starts at priority 2 and sets a variable like DESTCOMPANY to something specific to that destination. 3. At priority 3 in each source context, SRCCOMPANY and DESTCOMPANY are compared, in order to decide whether to override the CallerID with the source company's generic callerID. Let's say this uses priorities 3, 4 and 5 (for the GotoIf doing the compare, then the SetCallerID, and the NoOp target for the GotoIf when the callerID doesn't need rewriting). The destination contexts do not have priorities 3, 4 and 5. 4. The destination contexts continue at priority 6 to route the call. I think by interleaving priorities between contxts like this you should be able to achieve what you are looking for. Please let us know on the list if you are successful - it encourages us to keep helping in the future! Hope this helps Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 10:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
I think you're making it far too difficult.
What I do is something like this:
[outgoing]
include = internal
include = longdistance
;Always include internal first, as matches from the first include
;will be used first. This allows you to make sure your internal
;extensions don't go out your trunks.
[longdistance]
ignorepat = 9;
include = default; already included from local, but putting here for
clarity
include = local;
exten = _91XXX,1,Macro(trunkout,${EXTEN}) ;Medium Distance
exten = _91XX,1,Macro(trunkout,${EXTEN}) ;Long Distance
Then, I have:
[macro-trunkout]
exten = s,1,Set(cname=${DB(showname/${CALLERIDNUM})});
exten = s,n,Set(cnum=${DB(shownum/${CALLERIDNUM})});
exten = s,n,GotoIf($[foo${cnum} = foo]?6); //if
calling from ZAP
channel that set caller ID already
exten = s,n,Set(CALLERID(name)=${cname}|a);
exten = s,n,Set(CALLERID(number)=${cnum}|a);
exten = s,n,Dial(${TRUNK}/${ARG1:${TRUNKMSD}});
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CONGESTION,1,Congestion(30)
exten = s-CONGESTION,2,Hangup
exten = s-CANCEL,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-BUSY,2,Hangup
Why is this important? It's not. But it is fundamentally
different from
what you're asking. You want to match a partial extension dialed and
then continue appending digits. What you really need to do is
wait for
the whole number, then decide what kind of number it is, do the
processing, and send it on its way. It's just a slight change
in the way
you're thinking, because you understand that there's a class
of numbers
to treat differently. And that's OK. Just don't do anything with it
until the whole extension has been entered!
Uhm, No. I'm not trying to partially match extensions and then continue
appending digits. What makes you think that?
You'll notice that, anything not going through the trunkout macro
doesn't get tweaked, and anything that goes through there
will read from
the database. I could just as easily set a single value, but
I have some
users that I want to go out as themselves, and different departments
that have a general number, etc. I found the Asterisk
Database to be the
easiest to tweak, as I have some scripts to allow admins to
change the
effective CallerID on the fly.
David, this is completely different from what I am trying to do.
Let's try this a different way. Let's say you have two companies. When someone
calls a number in their own company, we use their INTERNAL caller id. When they
call someone in another company, we want to send their EXTERNAL caller id. How
would you do this?
Doug.
I hope this helps! Asterisk can do what you're asking, and it
does every
day.
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RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message- From: Tony Mountifield [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 11:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Match a Numer - then continue with dialplan In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: Don't get offended Doug, but I get really frustrated when I try to explain what I am trying to do with Asterisk, and people don't seem to quite get it. Your about the 4th person who's replied to this post, and hasn't quite grasped my question. :) --- smiley.. see...we're all cool. Perhaps its the terminology you used that is confusing people. See below: I don't want Asterisk to go on to ask for more digits. I want to do a very simple thing. I want to set a variable when call flow continues beyond a certain point (without asking the user for more digits), and then continue on, and use that variable later. It's a very simple thing, I can't work out why Asterisk doesn't let me do that. To almost all people call flow would mean executing one priority after another for a given extension. After reading and re-reading your posts trying to work out what you are trying to do, it seems to me that when *you* say call flow, you mean the act of trying to find an extension. And what your looking for is a way to do things a different points in the *search*, while it is still trying to decide on a statement to land on. Is that correct? Yes to the first sentence. Not quite sure what you mean after that. If so, I think you need to re-think the strategy a bit. The only way a command gets executed in a dialplan is when Asterisk has matched an extension and a priority. Then once it has executed that command, it increments the priority (unless it was a Goto or something) and starts searching again. That was my original question. I was asking if there was a way to set a variable and the continue, which doesn't seem like too strange a thing to have Asterisk support. However, don't forget that it searches for matching extensions every time the priority changes. You are not locked into a particular pattern or extension number from priority 1 onwards. You can mix and match patterns with literal extensions, even across includes, e.g. Don't follow you. When asterisk matches an extension, it starts interating through the priorities until there's none left, or you Goto() somewhere else. [example] include = ctx31X include = ctx3XX exten = _X.,1,NoOp(this gets executed first for everything) exten = _X.,2,NoOp(this gets executed second only if ctx31X or ctx3XX didnt match) exten = _X.,3,NoOp(this gets executed third for everything) You lost me here. [ctx31X] exten = _31X,2,NoOp(this gets executed second for 310-319) [ctx3XX] exten = _3XX,2,NoOp(this gets executed second for 300-309 and 320-399) So you might be able to do something along these lines by being creative with priorities and includes, and setting or testing variables. Something along these lines: 1. Each company starts off in its own context, and at Can't do that. The point at which a phone enters the dial plan needs to start with rather a long list of include= statements, to grant/deny access to certain features. priority 1 in _X. it sets a variable like SRCCOMPANY to something specific to it. It includes all the destination contexts. 2. Each destination context starts at priority 2 and sets a variable like DESTCOMPANY to something specific to that destination. 3. At priority 3 in each source context, SRCCOMPANY and DESTCOMPANY are compared, in order to decide whether to override the CallerID with the source company's generic callerID. Let's say this uses priorities 3, 4 and 5 (for the GotoIf doing the compare, then the SetCallerID, and the NoOp target for the GotoIf when the callerID doesn't need rewriting). The destination contexts do not have priorities 3, 4 and 5. 4. The destination contexts continue at priority 6 to route the call. I think by interleaving priorities between contxts like this you should be able to achieve what you are looking for. Please let us know on the list if you are successful - it encourages us to keep helping in the future! I tried your example, which I completely don't follow, and it didn't seem to execute as you expected. Dialling 311 yields: *CLI -- Executing NoOp(SIP/3254101-d10e, this gets executed first for everything) in new stack -- Executing NoOp(SIP/3254101-d10e, this gets executed second only if ctx31X or ctx3XX didnt match) in new stack -- Executing NoOp(SIP/3254101-d10e, this gets executed third for everything) in new stack I need to make extensive use of the include= directive, and I just can't see how getting stuck in priorities within an extension is going to allow me to do that. Doug. ___
Re: [asterisk-users] Need quality toll free 800 number over IAX?
Used les.net for outgoing for a while, seems to have some bandwidth problems -- call quality is low. Time Bandit wrote: I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Never used them but the rates seems ok : http://www.les.net/ ___ --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: Match a Numer - then continue with dialplan
-Original Message- From: Tony Mountifield [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 11:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Match a Numer - then continue with dialplan In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: Don't get offended Doug, but I get really frustrated when I try to explain what I am trying to do with Asterisk, and people don't seem to quite get it. Your about the 4th person who's replied to this post, and hasn't quite grasped my question. :) --- smiley.. see...we're all cool. Perhaps its the terminology you used that is confusing people. See below: I don't want Asterisk to go on to ask for more digits. I want to do a very simple thing. I want to set a variable when call flow continues beyond a certain point (without asking the user for more digits), and then continue on, and use that variable later. It's a very simple thing, I can't work out why Asterisk doesn't let me do that. To almost all people call flow would mean executing one priority after another for a given extension. After reading and re-reading your posts trying to work out what you are trying to do, it seems to me that when *you* say call flow, you mean the act of trying to find an extension. And what your looking for is a way to do things a different points in the *search*, while it is still trying to decide on a statement to land on. Is that correct? If so, I think you need to re-think the strategy a bit. The only way a command gets executed in a dialplan is when Asterisk has matched an extension and a priority. Then once it has executed that command, it increments the priority (unless it was a Goto or something) and starts searching again. However, don't forget that it searches for matching extensions every time the priority changes. You are not locked into a particular pattern or extension number from priority 1 onwards. You can mix and match patterns with literal extensions, even across includes, e.g. [example] include = ctx31X include = ctx3XX exten = _X.,1,NoOp(this gets executed first for everything) exten = _X.,2,NoOp(this gets executed second only if ctx31X or ctx3XX didnt match) exten = _X.,3,NoOp(this gets executed third for everything) [ctx31X] exten = _31X,2,NoOp(this gets executed second for 310-319) [ctx3XX] exten = _3XX,2,NoOp(this gets executed second for 300-309 and 320-399) So you might be able to do something along these lines by being creative with priorities and includes, and setting or testing variables. Something along these lines: 1. Each company starts off in its own context, and at priority 1 in _X. it sets a variable like SRCCOMPANY to something specific to it. It includes all the destination contexts. I think that's the deal breaker right there. I can't start a company within an extension. The starting point for each phone within a company needs to make extensive use of the include= directive. Features will be disabled by default, so there will be a list of includes to block unpurchased features. Then we'll include contexts for 911, voicemail retrieval and general numbers, ie: [coo1_CallStart] include = syst_FeaturePersonalMeetmeBlock include = syst_FeatureIntercomBlock include = syst_FeatureIDDBlock include = syst_Emergency include = syst_VMRetrieve include = coo1_General include = syst_GeneralInternal include = syst_ExportedApps include = syst_Route Finally, when we're finished scanning for blocked services, and asterisk terminated extensions, we try to route the call from this phone to the destination number, either OnNet or OffNet. That's where syst_Route comes in. For managability, we have to use lots of includes. We can't have our entire dialplan as one big _X. extension match. ___ --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] No music on hold?
Phil Finkler wrote: No, I didn’t have m added. Should I have it added? I know I’ve ran Asterisk with mp3123 in the past and music worked ok. It seems when I hit the hold button on the phones, it does trigger the message saying music on hold is starting but it INSTANTLY stops. I wish it gave some details as to WHY it stops suddenly! This is driving me nuts. Phil I was wondering the same thing as my MOH isn't working either in a 1.2.14 installation so I'm recompiling mpg123 as per: http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+redhat We know you obviously need to use the m flag for the caller to hear MOH when dialing an extension, but I wonder if it's required AFTER the call has been answered and then put on hold. -- 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
Re: [asterisk-users] No music on hold?
Lee Jenkins wrote: I was wondering the same thing as my MOH isn't working either in a 1.2.14 installation so I'm recompiling mpg123 as per: http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+redhat We know you obviously need to use the m flag for the caller to hear MOH when dialing an extension, but I wonder if it's required AFTER the call has been answered and then put on hold. OK, asterisk just finished compiling and my MOH is working correctly. I have also verified that you do *not* have to have m in the Dial command in order for MOH to play when placed on hold. Note that I have a command in the initial context of my dialplan that set music on hold: exten=s,1,SetMusicOnHold(default) -- 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
[asterisk-users] Dial own extension to get to voicemail.
I've gotten this Polycom 501 pretty much licked, but I need to know if there's a way in a dialplan to say if someone dials their own extension it goes straight to voicemail and asks them for their password. I thought I saw an example of this on the web but I can't seem to find it. Any advice appreciated! Phil ___ --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: Newbie Questions - Grandstorm phones?
Anthony,
Ok I understand. The 011 is unique though and I guess the problem is
the length of the remaining digits. This could vary based on country?? and
I suspect there is no unique rule that could be applied??? I have not
studied this but is there any uniqness to the remaining digits?
Doug
On Wed, 20 Dec 2006, Anthony Kepler wrote:
I have been using an approach such as this but am looking for something
else because of some limitations it has. The phone thinks it dialed,
and was connected to 011 (which it was)
As such, that will be stored in the phones dial history (redial if
nothing else).
I'm not even certain what I want is possible, which is why I'm asking
the list.
Thank you for your help once again though.
- Anthony Kepler
[EMAIL PROTECTED] | SIP/EMail
Doug Crompton wrote:
Well that is certainly an option but not all phones would have a send key
especially if you are using analog phones. I guess the # keys
functions in
that way on many of those.
I still like my wired phones to work like they use to. You dial a
number
and it executes the call immediately.
Ok I came up with one that I think would work, maybe needs some
refinement
[out-international]
exten = _011,1,goto(process-international,s,1)
[process-international]
exten = s,1,read(number)
exten = s,2,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = s,3,Macro(failann,${DIALSTATUS})
This accepts the 011 prefix and then any number of following digits.
Terminator is timeout period OR # key to send. Change obviously for your
provider.
The read command has many options including saying a file. You could for
instance hear Country Code after dialing 011. This would clue you into
the fact that you were dialing and international call. There are also
digit limits and timeouts that can be set.
So if you use early dial this would be the only rule that would require a
wait or # key to send. I could certainly live with that.
Can anyone supply some international test numbers??? Say in the UK or
Germany or wherever outside the US.
Doug
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Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
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RE: [asterisk-users] Dial own extension to get to voicemail.
What about comparing the caller id to the dialled number, and if they match, then call Voicemail() ? -Original Message- From: Phil Finkler [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dial own extension to get to voicemail. I've gotten this Polycom 501 pretty much licked, but I need to know if there's a way in a dialplan to say if someone dials their own extension it goes straight to voicemail and asks them for their password. I thought I saw an example of this on the web but I can't seem to find it. Any advice appreciated! Phil ___ --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: Match a Numer - then continue with dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes: DG If I pass a priority, we're right back at square one, we're I'm DG stuck in a priority and can't get back to an extension. You ALWAYS have both a priority and an extension. There is no such thing as being stuck in a priority. /Benny ___ --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] Need quality toll free 800 number over IAX?
www.IPKall.com [EMAIL PROTECTED] wrote: I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Anyone except NuFone. Their customer service is non-existant - you have to email every day for a couple of months before you'll be privileged enough to get a one-line response to a service outage issue. If you dare to point out that the response didn't address the issue then you'll unleash the combined wrath of both of the brain cells in residence at NuFone's support department. Not immediately, of course - you'll have to wait another couple of months for a reply. If you give up on them and decide to go elsewhere, they will pocket any outstanding funds you have pre-paid into your account. Existing NuFone customers are advised to not pre-pay too much to these yokels, and to jump ship as soon as possible. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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
RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 1:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Match a Numer - then continue with dialplan DG == Douglas Garstang [EMAIL PROTECTED] writes: DG If I pass a priority, we're right back at square one, we're I'm DG stuck in a priority and can't get back to an extension. You ALWAYS have both a priority and an extension. There is no such thing as being stuck in a priority. Benny, lets say I have this... exten = _X.,1,NoOp(1) exten = _X.,2,NoOp(2) exten = _X.,3,NoOp(3) - Current code execution location exten = 555,1,NoOp(1) exten = 555,2,NoOp(2) exten = 555,3,NoOp(3) How would I jump back into the dialplan from the current execution location and continue to search for matches? Doug ___ --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: Match a Numer - then continue with dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
-Original Message- From: Benny Amorsen
[mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006
6:16 AM To: asterisk-users@lists.digium.com Subject:
[asterisk-users] Re: Match a Numer - then continue with dialplan
exten = _.,n,Goto(coo1_CallStart2,${EXTEN},1)
[coo1_CallStart2] include = syst_OnNet include = syst_OffNet
DG That won't do it. Processing will continue in the current
DG extension priority. I need it to continue looking for an extension
DG to match against. Once Asterisk has matched the dialled number
DG against an extension in the dialplan, your stuck in an
DG extension you can never get out and get Asterisk to go back to
DG looking for extensions to match against.
It looks for extensions to match against all the time. What you say
makes no sense.
E.g. this code works, with EXTEN being 321 and starting in incoming.
[incoming]
exten = _3XX,1,NoOp(We get to this place)
exten = _X2X,2,Goto(incoming,${EXTEN},700)
exten = _XX1,700,NoOp(We end up here)
If EXTEN was 301, only priority 1 would run. If it was 320, priority 1
and 2 would run.
/Benny
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RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
[example] include = ctx31X include = ctx3XX exten = _X.,1,NoOp(this gets executed first for everything) exten = _X.,2,NoOp(this gets executed second only if ctx31X or ctx3XX didnt match) exten = _X.,3,NoOp(this gets executed third for everything) [ctx31X] exten = _31X,2,NoOp(this gets executed second for 310-319) [ctx3XX] exten = _3XX,2,NoOp(this gets executed second for 300-309 and 320-399) Does this really work? I've never seen this behavior documented anywhere. Asterisk always searches the current context before looking in included ones for a start. Second, I don't see how it can just jump out of [example] into [ctx31X] and back again without being told to do so ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Match a Numer - then continue with dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG Surely other people have hit the situation where they first check
DG extensions within a company, and then if there's no match, you
DG glue all the other companies dialplans together with this one.
Of course we have. Just Goto(gluedtogethercontext,${EXTEN},1)
/Benny
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Re: [asterisk-users] Need quality toll free 800 number over IAX?
Kevin Walsh wrote: www.IPKall.com [EMAIL PROTECTED] wrote: I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Anyone except NuFone. Their customer service is non-existant - you have to email every day for a couple of months before you'll be privileged enough to get a one-line response to a service outage issue. If you dare to point out that the response didn't address the issue then you'll unleash the combined wrath of both of the brain cells in residence at NuFone's support department. Not immediately, of course - you'll have to wait another couple of months for a reply. If you give up on them and decide to go elsewhere, they will pocket any outstanding funds you have pre-paid into your account. Existing NuFone customers are advised to not pre-pay too much to these yokels, and to jump ship as soon as possible There's is positively nothing at all incorrect about the above post. Might just want to add that if you do get an 800# with nufone, it may not be yours next time they fail to pay a supplier. If you use nufone and use your DID for anything more than sporadic personal calls, port it now. ___ --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] Dial own extension to get to voicemail.
On Wed, Dec 20, 2006 at 02:34:36PM -0500, Phil Finkler wrote:
I've gotten this Polycom 501 pretty much licked, but I need to know if
there's a way in a dialplan to say if someone dials their own extension
it goes straight to voicemail and asks them for their password. I
thought I saw an example of this on the web but I can't seem to find it.
Any advice appreciated!
You can do it, but it's more work than having an extension (the standard one
seems to be 86 now) that goes to:
VoicemailMain(s${CALLERID(num)[EMAIL PROTECTED]);
(But only in a context where the callerid can be trusted.)
To do what you want, you would need to have your extension processing
macro test if CALLERID(num) = ${EXTEN}, and then invoke the above
expression instead of dialing the extension.
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RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG Surely other people have hit the situation where they first check
DG extensions within a company, and then if there's no match, you
DG glue all the other companies dialplans together with this one.
Of course we have. Just Goto(gluedtogethercontext,${EXTEN},1)
After doing which, you can no longer use the include = directive.
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RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
-Original Message- From: Benny Amorsen
[mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006
6:16 AM To: asterisk-users@lists.digium.com Subject:
[asterisk-users] Re: Match a Numer - then continue with dialplan
exten = _.,n,Goto(coo1_CallStart2,${EXTEN},1)
[coo1_CallStart2] include = syst_OnNet include = syst_OffNet
DG That won't do it. Processing will continue in the current
DG extension priority. I need it to continue looking for an extension
DG to match against. Once Asterisk has matched the dialled number
DG against an extension in the dialplan, your stuck in an
DG extension you can never get out and get Asterisk to go back to
DG looking for extensions to match against.
It looks for extensions to match against all the time. What you say
makes no sense.
E.g. this code works, with EXTEN being 321 and starting in incoming.
[incoming]
exten = _3XX,1,NoOp(We get to this place)
exten = _X2X,2,Goto(incoming,${EXTEN},700)
exten = _XX1,700,NoOp(We end up here)
If EXTEN was 301, only priority 1 would run. If it was 320, priority 1
and 2 would run.
Ok, but how does that help me? All I want to do is set a variable to be used
later on in the dialplan.
Eg, if someone dialls 2944000, which is in a different company...:
[co1_phone-start]
include = co1_did
include = sys_glue
[co1_did]
exten = 3254101,1,Dial(SIP/3254101,18,tr)
exten = 3254102,1,Dial(SIP/3254102,18,tr)
exten = 3254103,1,Dial(SIP/3254103,18,tr)
; No match, so now we want to use the external caller id variable for use later
on, when
; we finally dial the dest number after performing all restriction and feature
checks.
; Actually I just realised we want to SET the caller id.
[sys-glue]
include co1_did
include co2_did
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[asterisk-users] Call Routing
HI I am able to setup the Dundi and works fine in locating the phone number's and extensions in branch office's. Only problem is unable to route the call if we receive it on serverA from PSTN and some one enter the extension number which reside in ServerB, it doesn't route the call. But if I dial the extension on ServerB from phone on serverA it works fine. Ali Arshad ___ --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: Match a Numer - then continue with dialplan
DG Surely other people have hit the situation where they first check
DG extensions within a company, and then if there's no match, you
DG glue all the other companies dialplans together with this one.
Of course we have. Just Goto(gluedtogethercontext,${EXTEN},1)
After doing which, you can no longer use the include = directive.
Perhaps I can get a clarification before proceeding further...
In reading the thread the situation seems to be: Company A users has a
user with extension/callerid XXX, he calls someone in company B and you
want to set the callerid to company A's main number rather than the
userr's default callerid?
Is this correct?
Mike
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RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message-
From: Mike [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG Surely other people have hit the situation where they
first check
DG extensions within a company, and then if there's no match, you
DG glue all the other companies dialplans together with this one.
Of course we have. Just Goto(gluedtogethercontext,${EXTEN},1)
After doing which, you can no longer use the include = directive.
Perhaps I can get a clarification before proceeding further...
In reading the thread the situation seems to be: Company A
users has a
user with extension/callerid XXX, he calls someone in company
B and you
want to set the callerid to company A's main number rather than the
userr's default callerid?
Is this correct?
Mike,
Exactamundo.
Doug.
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Re: [asterisk-users] Re: Match a Numer - then continue with dialplan
Perhaps I can get a clarification before proceeding further...
In reading the thread the situation seems to be: Company A
users has a
user with extension/callerid XXX, he calls someone in company
B and you
want to set the callerid to company A's main number rather than the
userr's default callerid?
Is this correct?
Mike,
Exactamundo.
Doug.
Ok.
How about:
;outgoing context for company A
[companyA]
;Various include statements
include = foo
.
.
.
;Handle calls from A - B
;Here will match company B numbers
exten = , 1, Set(CALLERID=CompanyAMain)
exten = , 1, Dial(${EXTEN}
You can do the inverse for companyB, or you could l have a single macro
that deals with calls to/from each company and decides what do to based
on the callerid making the call.
Mike.
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Re: [asterisk-users] Re: Match a Numer - then continue with dialplan
Typo, sorry. Should be:
Here will match company B numbers
exten = , 1, Set(CALLERID=CompanyAMain)
exten = , 2, Dial(${EXTEN})
;Handle calls from A - B
;Here will match company B numbers
exten = , 1, Set(CALLERID=CompanyAMain)
exten = , 1, Dial(${EXTEN}
You can do the inverse for companyB, or you could l have a single
macro that deals with calls to/from each company and decides what do
to based on the callerid making the call.
Mike.
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Re: [asterisk-users] AgentCallbackLogin() deprecated in 1.4
I have been speaking privately to a number of CC integrators and resellers about the AgentCallbackLogin() deprecation issue, and I'd dare say nobody is enthusiastic about it. With all its problems, AgentCallBackLogin is the workhorse of most of today's Asterisk CCs, and my impression is that the proposed solution meets a very lukewarm reception at the moment. Just my euro 0.02 l. On Wed, 20 Dec 2006 17:26:51 +0100, Douglas Garstang [EMAIL PROTECTED] wrote: The other thing is, that show agents doesn't show me which agents are logged in and if I use show queue I can see local channels attached to a queue but no agents. For my point of view there is some functionality lost with the new concept. Funny. I said the same thing in this list about 2 months ago and I got told I was nuts. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message-
From: Mike [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with
dialplan
Perhaps I can get a clarification before proceeding further...
In reading the thread the situation seems to be: Company A
users has a
user with extension/callerid XXX, he calls someone in company
B and you
want to set the callerid to company A's main number
rather than the
userr's default callerid?
Is this correct?
Mike,
Exactamundo.
Doug.
Ok.
How about:
;outgoing context for company A
[companyA]
;Various include statements
include = foo
.
.
.
;Handle calls from A - B
;Here will match company B numbers
exten = , 1, Set(CALLERID=CompanyAMain)
exten = , 1, Dial(${EXTEN}
You can do the inverse for companyB, or you could l have a
single macro
that deals with calls to/from each company and decides what
do to based
on the callerid making the call.
Mike.
Mike, this is a hosted IPT solution. There's potentially going to be hundreds
(we hope) of companies hosted and configured on this box. I'd have to write
static code to compare every number in every company to every number in every
other company, and that's just not feesible.
Doug.
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