Re: [asterisk-users] Problem with SPA3000 -- dropping calls
Most of the times it works, other times phone on FXS rings, I pick it up and all I get is a dial tone. this might also be MWI in SPA-style. regards adam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] buying cards from pakistan
Tzafrir Cohen wrote: On Fri, Apr 18, 2008 at 04:15:32PM +0200, giuliano curti wrote: On Thu, 17 Apr 2008 13:25:19 +0100 Alan Lord [EMAIL PROTECTED] wrote: [cut] I bought an X100p card from .. I have a similar card (X101P Tiger Jet) but seems does not recognize dmtf: external pstn callers cannot select from menu options, that is asterisk doesn't route the ext user selection coming from the channel Zap/1; While I do agree that those cards are not the best, I believe that DTMF issues may not be directly their problem. snip / I can confirm that the card I purchased works fine with DTMF tones :-) I have a simple IVR setup and can access voicemail boxes etc with no trouble at all. I am using the card with zaptel 1.4.9.2 (but a much older 1.4.5 or similar was fine too) and the OSLEC echo canceller. My zaptel.conf is this: loadzone=uk defaultzone=uk fxsks=1 My zapata.conf is this: ; Zapata telephony interface ; ; Configuration file [channels] ;Hardware defaults for the x100p card usecallerid=yes hidecallerid=no threewaycalling=yes transfer=yes echocancel=yes echotrainingwhenbridged=no cidsignalling=v23 ; Added for UK CLI detection ;cidstart=usehist ; After patching the driver from here : ; http://www.lusyn.com/resources/asterisk/usehist.htm callerid=asreceived ; propagate the CID received from BT rxgain=6.0 txgain=6.0 ;define channel context=main_menu language=en signalling=fxs_ks channel = 1 ;Our x100p FYI - I have not got CLI detection to work (yet) but I'm not sure if I have it enabled on my line and it wasn't a priority for me ;-) But I must say, since I discovered OSLEC, the card has worked a treat! Not bad for £17.00. Before I tried OSLEC echo was pretty bad. HTH Alan -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP calls
Chris Mason wrote: QOS can only be on outgoing, you can't set the priority of a packet after you receive it. The only other solution would be the cooperation of the ISP to provide QOS upstream of you. Good luck. QOS is probably not the most precise term as it's normally associated with RSVP, MPLS, packet headers, etc. But you can, in Netscreens at least, define a Guaranteed Bandwidth. We do this for SIP/IAX IPs, in both outgoing and incoming policies, and it works both ways. Audio quality is good and there are no chan_sip.c: Peer is now (UNREACHABLE|Lagged) messages even during long DVD or Bitorrent xfers. The reason it works outbound is a no-brainer, but inbound bandwidth is also effectively guaranteed. Sure there's no way to control external devices that ignore ICMP source-quench or break TCP congestion control but those flows are typically limited to nefarious sources which would not be responsive to other types of QOS anyhow (BGP being one potential exception). Roger Marquis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] func_curl.so Error on load
Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start up Asterisk (with -cvvv) I get an error regarding func_curl.so (lines omitted) ... == Registered custom function STRFTIME == Registered custom function STRPTIME == Registered custom function EVAL == Registered custom function KEYPADHASH == Registered custom function SPRINTF func_strings.so = (String handling dialplan functions) == Registered application 'ADSIProg' app_adsiprog.so = (Asterisk ADSI Programming Application) asterisk: symbol lookup error: /usr/lib/asterisk/modules/func_curl.so: undefined symbol: curl_global_init This is on CentOS 5.1, kernel 2.6.18-53.1.14 on an i686. Both curl and curl-devel 7.15.5-2.el5 are installed. Is there a symlink somewhere that Asterisk is looking for? Or does it need a newer version of curl? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_curl.so Error on load
Nevermind, I found the problem. Chris Brentano wrote: Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start up Asterisk (with -cvvv) I get an error regarding func_curl.so (lines omitted) ... == Registered custom function STRFTIME == Registered custom function STRPTIME == Registered custom function EVAL == Registered custom function KEYPADHASH == Registered custom function SPRINTF func_strings.so = (String handling dialplan functions) == Registered application 'ADSIProg' app_adsiprog.so = (Asterisk ADSI Programming Application) asterisk: symbol lookup error: /usr/lib/asterisk/modules/func_curl.so: undefined symbol: curl_global_init This is on CentOS 5.1, kernel 2.6.18-53.1.14 on an i686. Both curl and curl-devel 7.15.5-2.el5 are installed. Is there a symlink somewhere that Asterisk is looking for? Or does it need a newer version of curl? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme with time condition
Hi All, How can i enable time condition on meetme? below i would like to deny callers if the time is not yet the scheduled time of the conference, but it seems like its still goes to 600,2, hope anyone can help. [meet-me-test] exten = 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3) exten = 600,2,Playback(vm-goodbye) exten = 600,3,Hangup exten = 600,4,MeetMe(600||600600) regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_curl.so Error on load
On Sat, Apr 19, 2008 at 11:11:53AM -0700, Chris Brentano wrote: Nevermind, I found the problem. And for the benefit of the readers of the archives: what was it? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme with time condition
Hi All, How can i enable time condition on meetme? below i would like to deny callers if the time is not yet the scheduled time of the conference, but it seems like its still goes to 600,2, hope anyone can help. [meet-me-test] exten = 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3) exten = 600,2,Playback(vm-goodbye) exten = 600,3,Hangup exten = 600,4,MeetMe(600||600600) regards, nhadie - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme with time condition
On Sun, 20 Apr 2008, nhadie ramos wrote: How can i enable time condition on meetme? below i would like to deny callers if the time is not yet the scheduled time of the conference, but it seems like its still goes to 600,2, hope anyone can help. [meet-me-test] exten = 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3) exten = 600,2,Playback(vm-goodbye) exten = 600,3,Hangup exten = 600,4,MeetMe(600||600600) 1) I read this as If the time is between 10 and 11 in the morning on the 19th of April, hangup. Else, play goodbye and hangup. Neither case will get you to the conference. 2) Is it still the 19'th in your part of the world? 3) Try replacing each element with * to identify the source of your problem. And a few suggestions: 1) Since the target of your gotoiftime is in the same context and the same extension, you only need to specify the priority. 2) You can use the n priority. Both will make your code more robust, maintainable, and reusable. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme with time condition
Hi Steve,
Thanks for the reply. I made it look like this:
[meet-me-test]
exten = 600,1,GotoIfTime(11:00-12:00|*|20|Apr?meetnow)
exten = 600,n,Answer
exten = 600,n,Festival('Conference is not yet active, please dial in on the
assigned time');
exten = 600,n,Hangup
exten = 600,n(meetnow),MeetMe(600||600600)
exten = 600,n,Hangup
and it's kind of working now, except for the Festival I added, i dont hear any
audio but it's executing it
Executing [EMAIL PROTECTED]:3] Festival(SIP/1100-b691a738, Conference is
not yet active, please dial in on the assigned time) in new stack.
also, how can i limit the time of the conference? on this scenario how would it
get cut on 12:00? is it also possible to extend the time limit even if the
conference is already ongoing? thanks alot.
regards,
nhadie
=
1) I read this as If the time is between 10 and 11 in the morning on the
19th of April, hangup. Else, play goodbye and hangup. Neither case will
get you to the conference.
2) Is it still the 19'th in your part of the world?
3) Try replacing each element with * to identify the source of your
problem.
And a few suggestions:
1) Since the target of your gotoiftime is in the same context and the same
extension, you only need to specify the priority.
2) You can use the n priority.
Both will make your code more robust, maintainable, and reusable.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
Nhadie Ramos [EMAIL PROTECTED] wrote: Hi All,
How can i enable time condition on meetme? below i would like to deny
callers if the time is not yet the scheduled time of the conference, but
it seems like its still goes to 600,2, hope anyone can help.
[meet-me-test]
exten = 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3)
exten = 600,2,Playback(vm-goodbye)
exten = 600,3,Hangup
exten = 600,4,MeetMe(600||600600)
regards,
nhadie
-
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now.___
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Re: [asterisk-users] func_curl.so Error on load
Tzafrir Cohen schrieb: On Sat, Apr 19, 2008 at 11:11:53AM -0700, Chris Brentano wrote: Nevermind, I found the problem. And for the benefit of the readers of the archives: what was it? Sometimes I get the impression that it's an illusion to think someone would actually care to read the archives. (btw: How many concurrent calls can Asterisk handle, rougly? ;-) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
