Re: [asterisk-users] DECT SIP phones
On Thu, Sep 13, 2007 at 06:05:51PM -0600, Stephen Bosch wrote: I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). Hello! I would reccomend the Kirk DECT gateway. It is SIP capable and avilable for N America. We have a setup with the Skinny ( chan_sccp ) protocol and in Sweden, but I wouldn't expect any problems in NA. Our customer have used it for a while now. Regards:Håkan pgpwx3EBrFqVG.pgp Description: PGP signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX
Thank you, I did what you mentioned below. It seems that I'm getting a hangupcause of 0 which I believe is not defined. Is Alcatel the first party that is trying to disconnect or is it Asterisk? (Not sure how to interpret the debug info I'm posting below) Whether it's Alcatel or Asterisk, what could be the actual cause? (or where should I start looking?) Thanks INF-VOIP*CLI pri debug span 1 Enabled debugging on span 1 -- Executing NoOp(SIP/4053-083189e8, [ALCATEL TEST] Start) in new stack -- Executing Dial(SIP/4053-083189e8, Zap/g1/5900) in new stack 1 -- Making new call for cr 32781 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (8) len=32 1 Call Ref: len= 2 (reference 13/0xD) (Originator) 1 Message type: SETUP (5) 1 [04 03 80 90 a3] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [18 04 e9 81 83 81] 1 Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 1 ChanSel: Reserved 1Ext: 1 DS1 Identifier: 1 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 1 ] 1 [6c 06 21 80 34 30 35 33] 1 Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number not screened (0) '4053' ] 1 [70 05 a1 35 39 30 30] 1 Called Number (len= 7) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5900' ] 1 [a1] 1 Sending Complete (len= 1) -- Called g1/5900 1 Protocol Discriminator: Q.931 (8) len=10 1 Call Ref: len= 2 (reference 13/0xD) (Terminator) 1 Message type: CALL PROCEEDING (2) 1 [18 03 a9 83 81] 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 1 ChanSel: Reserved 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 1 ] 1 -- Processing IE 24 (cs0, Channel Identification) -- Zap/1-1 is proceeding passing it to SIP/4053-083189e8 1 Protocol Discriminator: Q.931 (8) len=5 1 Call Ref: len= 2 (reference 13/0xD) (Terminator) 1 Message type: ALERTING (1) -- Zap/1-1 is ringing -- Zap/1-1 is busy 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered, peerstate Call Received 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 13/0xD) (Originator) 1 Message type: DISCONNECT (69) 1 [08 02 81 90] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:1/0/0) -- Executing NoOp(SIP/4053-083189e8, [ALCATEL TEST] hangupcause: 0) in new stack -- Executing Hangup(SIP/4053-083189e8, ) in new stack == Spawn extension (custom-TEST_ALCATEL, s, 4) exited non-zero on 'SIP/4053-083189e8' 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 13/0xD) (Terminator) 1 Message type: RELEASE (77) 1 [08 02 81 90] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] 1 -- Processing IE 8 (cs0, Cause) 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 13/0xD) (Originator) 1 Message type: RELEASE COMPLETE (90) 1 [08 02 80 90] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) 1 Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Channel 1/1, span 1 received AOC-E charging 0 units INF-VOIP*CLI quit --- Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Looks like the Alcatel is sending back a busy. Check the value of HANGUPCAUSE with a Noop as the priority after the Dial. You may also want to do a pri debug span X to see the actual Q.931 ISDN messages that are exchanged. Vieri wrote: An Asterisk extension calls an Alcatel extension via a PRI link which rings 4 times for about 10-15 seconds and then drops. So if the Alcatel user doesn't answer within 10-15 seconds the call is aborted. (A timeout is *not* specified in the Asterisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk
Re: [asterisk-users] DECT SIP phones
On 08:00, Fri 14 Sep 07, H?kan K?llberg wrote: On Thu, Sep 13, 2007 at 06:05:51PM -0600, Stephen Bosch wrote: I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). Hello! I would reccomend the Kirk DECT gateway. It is SIP capable and avilable for N America. We have a setup with the Skinny ( chan_sccp ) protocol and in Sweden, but I wouldn't expect any problems in NA. Our customer have used it for a while now. We use a setup like that with chan_sccp on 1.2 on one customer location. I dont know if you have NEC-Philips there in NA but they have great dect/sip setups as well. we use them in a couple of installations in medical facilities (man down, assistant call, that kindda stuff) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bug in 1.2.24
On 13:33, Fri 14 Sep 07, Isaac Xiao wrote:
Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten = 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten = 7141,5,MixMonitor(${CALLFILENAME}.gsm|b)
exten = 7141,6,Playback(custom/None)
exten = 7141,7,Queue(7141|t|||7200)
Here is the CLI log.
-- Executing Playback(Zap/9-1, monitoring) in new stack
-- Playing 'monitoring' (language 'md')
-- Executing Playback(Zap/9-1, press-1-to-msg) in new stack
-- Playing 'press-1-to-msg' (language 'md')
-- Executing Goto(Zap/9-1, ext-queues|7141|1) in new stack
-- Goto (ext-queues,7141,1)
-- Executing NoOp(Zap/9-1, do not answer call before entering
queue) in new stack
-- Executing SetCIDName(Zap/9-1, CN) in new stack
-- Executing Set(Zap/9-1,
CALLFILENAME=q7141-20070914-132445-1189740177.10324) in new stack
-- Executing Set(Zap/9-1, __FROM-EXT-QUEUES=ext-queues) in new
stack
-- Executing MixMonitor(Zap/9-1,
q7141-20070914-132445-1189740177.10324.gsm|b) in new stack
-- Executing Playback(Zap/9-1, custom/None) in new stack
-- Executing Queue(Zap/9-1, 7141|t|||7200) in new stack
So Yes. As long as Zap/9-1 channel (customer's channel) not hangs up, it
will be always recorded.
Can you confirm this bug is present in 1.4 as well?
1.2 has gone into security-patches-only mode so no bugfixes
will be made for it.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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Re: [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Looks like the Alcatel is sending back a busy. Note: I'm using libpri patched with BRIstuff. http://ftp.digium.com/pub/libpri/libpri-1.2.4.tar.gz http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1y-d.tar.gz Tonight's top picks. What will you watch tonight? Preview the hottest shows on Yahoo! TV. http://tv.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [SOLVED] fax machine detection for outgoing call on DIVAcard
I was helped by Armin Schindler from the chan_capi user list. So this is my answer and the solution to the chan_capi list : - Make sure you have enabled the onboard DSP by using softdtmf=off relaxdtmf=off in capi.conf. Many thanks, it worked perfectly :-). For information, when the fax is detected, asterisk performs a jump to the fax extension of the context associated to the controller and defined in the capi.conf. _ Ten : Messenger en illimité sur votre mobile ! http://mobile.live.fr/messenger/ten/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff
I don't know about the 1.4 source, but in 1.2 I guess you would have to add some more code to handle_response_peerpoke() to handle the case where you got a 486 response from the peer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vieri Sent: 13 September 2007 18:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff --- Steve Langstaff [EMAIL PROTECTED] wrote: Can you hook into the qualify code somehow? - that uses SIP OPTIONS. I already knew of this wiki page: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify So I did a sip show peer on the asterisk cli which I am supposing is the same as the SIPPEER function. When SIP softphone has DND turned OFF: INF-VOIP*CLI sip show peer 4053 INF-VOIP*CLI * Name : 4053 Secret : Set MD5Secret: Not set Context : from-internal Subscr.Cont. : Not set Language : es AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : device 4053 Expire : 58 Insecure : no Nat : Always ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.215.147.240 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 4053 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw,alaw) Status : OK (127 ms) Useragent: SJphone/1.65.377a (SJ Labs) Reg. Contact : sip:[EMAIL PROTECTED] When SIP softphone has DND turned ON: INF-VOIP*CLI sip show peer 4053 INF-VOIP*CLI * Name : 4053 Secret : Set MD5Secret: Not set Context : from-internal Subscr.Cont. : Not set Language : es AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : device 4053 Expire : 45 Insecure : no Nat : Always ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.215.147.240 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 4053 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw,alaw) Status : OK (127 ms) Useragent: SJphone/1.65.377a (SJ Labs) Reg. Contact : sip:[EMAIL PROTECTED] INF-VOIP*CLI I don't see any difference and SIP Options : (none) doesn't look good. (the SIP extension has qualify=yes) __ __ Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff
--- Steve Langstaff [EMAIL PROTECTED] wrote: I don't know about the 1.4 source, but in 1.2 I guess you would have to add some more code to handle_response_peerpoke() to handle the case where you got a 486 response from the peer. ok thanks, so that just seems to confirm that Asterisk 1.2 DND's behavior can't be modified/customized without patching the source code. I might forward this issue to asterisk-devel. Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk cli
On 9/13/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Sep 13, 2007 at 10:36:56AM -0500, Mark Michelson wrote: Rizwan Hisham wrote: i connect remotely. I have tried both of these cases but no warnings or mesages still. It could be that your logger.conf file doesn't know to send debug messages to the cli. Make sure that the console line in logger.conf includes debug. Mine looks like: console = notice,warning,error,debug Debug messages will just flood your console and make it non-functional. Well, this is interesting to see how to enable debug in console, but i agree - it would make very very much of everything appear there.. I usually have another terminal open, where i do tail -n0 -f /var/log/asterisk/full So, you can look into dialplan execution, and if something goes wrong - switch to second terminal where debug is tailed.. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk voice quality tuning
Satish, Whats your network setup? Do you get bad quality on internal-asterisk calls, or only on external calls? Are you making pure IP calls (sip2sip), or are there E1/T1 cards involved? What codecs are you currently using? What devices are you using? Adrian Marsh From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of satish patel Sent: 14 September 2007 06:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk voice quality tuning Dear all I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE405P intermittent yellow alarm
Thanks to everyone who helped with this. Don Pobanz wrote: On Thursday, September 13, 2007 4:58 AM Richard van der Hoff said Thanks for your help, but again I'd like to ask: what does a yellow alarm actually mean? From the driver source code I can see it is set when the FRS0 register has bit 4 set - but that doesn't help a lot... All of my experience has been with T1s, not E1s but I assume the alarms mean the same even though they are transmitted differently. Suppose that there are three pieces of equipment 'A', 'B', and 'C' and the signal from 'A' to 'B' has been interrupted (designated by the 'X' in the diagram) so that 'B' is not seeing an incoming signal. 'B' will be in red alarm, and 'B' will transmit back to 'A' a yellow alarm indicator. When 'A' see the yellow alarm indicator, 'A' will go into yellow alarm. So basically, a yellow alarm as shown by zttool etc just means that the remote equipment is sending a yellow alarm indicator. Which is odd, because in this case, equipment 'B' is a BT NTE51D (like one of these: http://www.sjgl.co.uk/isdn/pri-nte.htm) which was showing no faults whatsoever (you can log into it over a serial link and read the fault log). For the record, I think the fault has now been resolved, by the BT engineers fiddling about with stuff in the exchange. Their report was Monitored line and found line level was dipping below specified levels. Reterminated jumpers and reseated LTE. So it was their fault all along - it just would have been nice to have been able to give them a bit more information than um, our kit is showing a yellow alarm, but I don't know what that means. Thanks again, Richard -- Richard van der Hoff [EMAIL PROTECTED] Project Manager Tel: +44 (0) 845 666 7778 http://www.mxtelecom.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT SIP phones
Håkan Källberg schrieb: On Thu, Sep 13, 2007 at 06:05:51PM -0600, Stephen Bosch wrote: I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). Hello! I would reccomend the Kirk DECT gateway. It is SIP capable and avilable for N America. We have tested the Kirk DECT Gateway for internal use with we SIP implementation. To set up the base station to work with Asterisk and to register phones at a base station it quite a bit of work, but manageable. Sound is quite good, in some cases too good. They shut down the mic while you are not talking, to eliminate background noise while the other party is speaking. The absolute calmness leads the other party to the conclusion that the call was disconnected, very annoying. The phones are quite expensive (up to €170 per unit + €600 for the base station). For this price we wanted an distributed phone book, maybe the base station being able to look up the numbers in an extern ldap server. Support od Kirk told us, that there some development in this sector, but nothing ready for deployment. After this conclusion, we have sent the kirk setup back und bought the siemens S450 setup. The Base station is limited to 2 concurrent calls, but if you only register 2 phones per base station this is no drawback. Now we have 5 base stations running with 2 or 3 phones per station for less money than we kirk setup. So far it works very good. Although there is no solution for a distributed telephone book. You can sent phone books to the telephone per web interface as an Vcard-File. We are working on an automated solution that erases the phone in all the telephones and sent new phonebook files to them. Regards, -- Tobias Wolf Leiter Softwareentwicklung / Kommunikationslösungen Evision GmbH Wittekindstr. 105 44139 Dortmund Tel: +49 (0)231 - 47790 307 Fax: +49 (0)231 - 47790 500 http://www.evision.de This electronic mail transmission and any accompanying attachments contain confidential information intended only for the use of the individual or entity named above. Any dissemination, distribution, copying or action taken in reliance on the contents of this communication by anyone other than the intended recipient is strictly prohibited. If you have received this communication in error please immediately delete the E-mail and notify the sender at the above E-mail address. Thank you. Hövener Trapp Evision GmbH, Dortmund - HRB Nr.12477, Registergericht Dortmund - Geschäftsführer Christoph Begall ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff
--- Steve Langstaff [EMAIL PROTECTED] wrote: I don't know about the 1.4 source, but in 1.2 I guess you would have to add some more code to handle_response_peerpoke() to handle the case where you got a 486 response from the peer. ok thanks, so that just seems to confirm that Asterisk 1.2 DND's behavior can't be modified/customized without patching the source code. I might forward this issue to asterisk-devel. What do you mean modified/customized exactly? If you mean can you know whether a device has DND enabled or not before sending a call then no, even an OPTIONS packet won't tell you that. You send a call, they reject (and sometimes they even use a response code that doesn't indicate it's DND). Same goes for call forwarding. You send a call, they reject saying go here instead. Joshua Colp Software Developer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Serusers] user meeting (beer drinking in Vienna)
Curses! I just got BACK from Vienna yesterday. I should have stayed another week. :) N. Klaus Darilion wrote: Hi! I proudly announce the first ser/openser/asterisk beer drinking evening in Vienna. When: Thursday (thirsty day) 20. September 2007, 19:00 CEST Where: Vienna, a bar in an inner district - exact location to be announced If you want to join please send a short reply so that I know for how many people I have to arrange a table. cu Klaus ___ Serusers mailing list [EMAIL PROTECTED] http://lists.iptel.org/mailman/listinfo/serusers ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallWithUs Service?
There has to be some reasonable priced sip provider that would perform just as well as ATT. Does it exist? The problem is that there is no QoS control between you and the provider, so a lot of the quality issues you have are probably not related to the specific provider, but just the general Internet. Until there is QoS everywhere, nobody is going to perform as well as ATT and certainly not at what everybody thinks is reasonable (1 cent per minute). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mutipoint Conferencing?
I am trying to determine what would need to be done/modified to enable the following: I have a SIP extension come into my asterisk box, and I then need it to call 6-10 remote Sip Stations that are set to Auto-Answer... (note, my remote sip stations are actually cisco h323 devices, I can call them fine from any softphone, or other device, and have full-duplex audio, however, i need to be able to conference bring all the remote stations automatically.w/Full duplex audio. Or if someone could direct me to a list that would actually be able to answer this question.. Thanks, W. Stillwell Sent via the WebMail system at kotbh.net ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff
The OP was asking whether they could update Asterisk's DND status for the extension to mirror a DND button on the (SIP) phone. I suggested that they might act on the response code to an OPTIONS. I think that they *actually* want to do some queue management based on the DND button of the (SIP) phone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: 14 September 2007 08:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff --- Steve Langstaff [EMAIL PROTECTED] wrote: I don't know about the 1.4 source, but in 1.2 I guess you would have to add some more code to handle_response_peerpoke() to handle the case where you got a 486 response from the peer. ok thanks, so that just seems to confirm that Asterisk 1.2 DND's behavior can't be modified/customized without patching the source code. I might forward this issue to asterisk-devel. What do you mean modified/customized exactly? If you mean can you know whether a device has DND enabled or not before sending a call then no, even an OPTIONS packet won't tell you that. You send a call, they reject (and sometimes they even use a response code that doesn't indicate it's DND). Same goes for call forwarding. You send a call, they reject saying go here instead. Joshua Colp Software Developer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk voice quality tuning
I have both type of call sip-2-pstn and pstn-2 -sip but quality is not good so how to check asterisk voice quality and codec quality i am useing G.711 alaw and ulaw and it is my LAN network so is there any specific perameter or option to improve quality of voice ??? Adrian Marsh [EMAIL PROTECTED] wrote: Satish, Whats your network setup? Do you get bad quality on internal-asterisk calls, or only on external calls? Are you making pure IP calls (sip2sip), or are there E1/T1 cards involved? What codecs are you currently using? What devices are you using? Adrian Marsh From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of satish patel Sent: 14 September 2007 06:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk voice quality tuning Dear all I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more!___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallWithUs Service?
On Thursday 13 September 2007 02:32:52 pm John Meksavan wrote: I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? it was your message that prompted me to take a look at callwithus.com. i currently use diamondcard.us (via iax2) and have had only 2 issues in 9 months where some calls to verizon cell phones would get a congestion signal if they didn't answer instead of going to their voicemail. i called diamondcard and they fixed the trunk issue in a matter of an hour. call quality is decent. after signing up with callwithus.com, i find the call quality to be the same as diamondcard, though diamondcard bills in 30sec increments at 1.7 cents/min in the us and callwithus bills in 1 minute increments at 1.4 cents/min in the us. callwithus also has this thing where if you add a *31 to the number, it will choose their cheapest route. i'd say they are worth trying, so is diamondcard.us. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outgoing call restriction in extention.conf
Dear all I have asterisk PBX and 100 endpoint i want to block STD for specific users or password protect so is it possible users can set passwd on his/her phone and password automaticaly reflacted on asterisk in short i want to restrict STD call of users of outgoing Regards satish patel - Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff
SIP response 486 is Busy Here according to RFC 3326. Polycoms at least (and I think Cisco phones) do not send back a different message depending on if DND is enabled .vs. the line appearance simply being busy. Personally I can't see how the people that designed SIP could justify not being able to get the DND status or CFWD status of a SIP device. Steve Langstaff wrote: The OP was asking whether they could update Asterisk's DND status for the extension to mirror a DND button on the (SIP) phone. I suggested that they might act on the response code to an OPTIONS. I think that they *actually* want to do some queue management based on the DND button of the (SIP) phone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: 14 September 2007 08:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff --- Steve Langstaff [EMAIL PROTECTED] wrote: I don't know about the 1.4 source, but in 1.2 I guess you would have to add some more code to handle_response_peerpoke() to handle the case where you got a 486 response from the peer. ok thanks, so that just seems to confirm that Asterisk 1.2 DND's behavior can't be modified/customized without patching the source code. I might forward this issue to asterisk-devel. What do you mean modified/customized exactly? If you mean can you know whether a device has DND enabled or not before sending a call then no, even an OPTIONS packet won't tell you that. You send a call, they reject (and sometimes they even use a response code that doesn't indicate it's DND). Same goes for call forwarding. You send a call, they reject saying go here instead. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff
--- Steve Langstaff [EMAIL PROTECTED] wrote: The OP was asking whether they could update Asterisk's DND status I think that they *actually* want to do some queue management based on the DND button of the (SIP) phone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp What do you mean modified/customized exactly? As Steve pointed out, we need to manage queues where the strict behavior is not exactly what we want. Here's a snippet of my earlier post: -- Also, can this problem be handled the other way around? Can Asterisk be configured somehow so that whenever someone tries to call a particular extension and the latter yields a 'response 486 Do Not Disturb' then the DND field in AstDB for that extension is updated? This way the custom AGI script would only need to execute database show dnd... I need this particularly for queues that have the strict option for joining and leaving. In this situation a custom cron script adds and removes members dynamically from the queues. The problem I found is that strict behavior works as expected when the agents, even if added via AddQueueMember, are logged off or have their softphone turned off but fails if they activate DND (either by pressing the softphone DND button or dialing *78). So a solution I am thinking of implementing is to change this custom cron script and make it detect if certain SIP extensions have DND on or not (either with database show dnd or any other reliable method). If it detects an activated DND then it will execute a RemoveQueueMember(queuenum, sipnum). If it detects that DND is off again then it will run an AddQueueMember(queuenum, sipnum). Tonight's top picks. What will you watch tonight? Preview the hottest shows on Yahoo! TV. http://tv.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff
To work around our inability to know in advance whether or not, an extension is forwarded or DNDed, we disabled those features (using hardphone settings) and provided a software replacement (which edit database from which Asterisk check user preferences for every call). This is completely against SIP spirit as intelligence is then mostly concentrated in Asterisk but we couldn't find anything else. I would be surprised to find anytime soon, a SIP phone, allowing DND or call forwardings to be handled externally. Maybe one day, a SIP method would easy administrators to query SIP phone status. Regards ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff
Eric ManxPower Wieling wrote: SIP response 486 is Busy Here according to RFC 3326. Polycoms at least (and I think Cisco phones) do not send back a different message depending on if DND is enabled .vs. the line appearance simply being busy. Personally I can't see how the people that designed SIP could justify not being able to get the DND status or CFWD status of a SIP device. Steve Langstaff wrote: The OP was asking whether they could update Asterisk's DND status for the extension to mirror a DND button on the (SIP) phone. I suggested that they might act on the response code to an OPTIONS. I think that they *actually* want to do some queue management based on the DND button of the (SIP) phone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: 14 September 2007 08:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff --- Steve Langstaff [EMAIL PROTECTED] wrote: I don't know about the 1.4 source, but in 1.2 I guess you would have to add some more code to handle_response_peerpoke() to handle the case where you got a 486 response from the peer. ok thanks, so that just seems to confirm that Asterisk 1.2 DND's behavior can't be modified/customized without patching the source code. I might forward this issue to asterisk-devel. What do you mean modified/customized exactly? If you mean can you know whether a device has DND enabled or not before sending a call then no, even an OPTIONS packet won't tell you that. You send a call, they reject (and sometimes they even use a response code that doesn't indicate it's DND). Same goes for call forwarding. You send a call, they reject saying go here instead. When a device is called and it is in CFWD mode it sends back a redirect message (Moved Temporarily), Asterisk displays in the CLI Recieved Moved Temporarily trying XX thanks to XXX.XXX.XXX.XXX or something along those lines. This helps you know what is in SIP messages: http://www.ietf.org/rfc/rfc3263.txt -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing call restriction in extention.conf
satish patel wrote: Dear all I have asterisk PBX and 100 endpoint i want to block STD for specific users or password protect so is it possible users can set passwd on his/her phone and password automaticaly reflacted on asterisk in short i want to restrict STD call of users of outgoing Regards satish patel Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://us.rd.yahoo.com/evt=48224/*http://sims.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This page has an example of call limiting, to make them need an access code, try using authenticate. http://www.voip-info.org/wiki/index.php?page=Asterisk+func+GROUP This site is your friend and should be your first stop when trying to do basic asterisk things. -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNOW + legacy PBX integration
Hi, I wonder if this question has been answered before, but im kind of stuck.. I have been trying to setup AsteriskNOW with a Digium TDM844B card with 4FXS/4FXO modules.. trying to connect it with a Panasonic KT616 PABX.. this has 6CO ports and 16 extensions. All the extensions are used up, the only free ports are the CO ports which have never been used. My layout is to connect PSTN connections to the 4FXO ports , and have the 4FXS ports connect to the Panasonic PABX. I wish to be able to have asteriskNOW as the telephony gateway to the organization, from the PSTN lines. There is a remote office with about 5 users, i expect to be able to have them use SIP phones, as the two offices are connected with a high bandwidth radio connection. I wish to be able to use asteriskNOW for interoffice calling, IVR, and call hunting. My confusion is how to setup the Panasonic PABX on asteriskNOW.. so that SIP users can dial extensions on the Panasonic PABX, and the Panasonic extensions can dial the SIP users in the remote office on AsteriskNOW. How do i properly define the Panasonic,in asteriskNOW, so that this is possible? Without breaking any configs? I tried adding new contexts to the extensions.conf but they were not recognized.. How do i properly edit the existing dial plan to include my needs? I also need to be able to achieve this fairly graphically so that if i need to leave, the other designated IT guys or a member of staff can make changes to the system without messing with configuration files, and it would be a lot easier to support. Im trying to convince the boss to invest in a channel bank, or astribank, but im not having much luck as i have to justify the TDM844B or its my salary :) Can anyone advise? Thanks very much in advance -- Shina Owolabi 2348034022578 23417203257 2341360480 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff
Anthony Francis wrote: Eric ManxPower Wieling wrote: SIP response 486 is Busy Here according to RFC 3326. Polycoms at least (and I think Cisco phones) do not send back a different message depending on if DND is enabled .vs. the line appearance simply being busy. Personally I can't see how the people that designed SIP could justify not being able to get the DND status or CFWD status of a SIP device. Steve Langstaff wrote: The OP was asking whether they could update Asterisk's DND status for the extension to mirror a DND button on the (SIP) phone. I suggested that they might act on the response code to an OPTIONS. I think that they *actually* want to do some queue management based on the DND button of the (SIP) phone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: 14 September 2007 08:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff --- Steve Langstaff [EMAIL PROTECTED] wrote: I don't know about the 1.4 source, but in 1.2 I guess you would have to add some more code to handle_response_peerpoke() to handle the case where you got a 486 response from the peer. ok thanks, so that just seems to confirm that Asterisk 1.2 DND's behavior can't be modified/customized without patching the source code. I might forward this issue to asterisk-devel. What do you mean modified/customized exactly? If you mean can you know whether a device has DND enabled or not before sending a call then no, even an OPTIONS packet won't tell you that. You send a call, they reject (and sometimes they even use a response code that doesn't indicate it's DND). Same goes for call forwarding. You send a call, they reject saying go here instead. When a device is called and it is in CFWD mode it sends back a redirect message (Moved Temporarily), Asterisk displays in the CLI Recieved Moved Temporarily trying XX thanks to XXX.XXX.XXX.XXX or something along those lines. This helps you know what is in SIP messages: http://www.ietf.org/rfc/rfc3263.txt But there is no method to QUERY the CFWD status. I understand what happens when a call goes to a SIP device with CFWD enabled. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DISA and DTMF detection problem w/ FXO port on a TDM400
Originally posted at http://forums.digium.com/viewtopic.php?t=18045
Hi!
I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing
DISA seems to prevent any DTMF detection capability when using the FXO
port of the TDM400.
Below, config A and B and their debug logs.
In Config A I use Authenticate() instead of using DISA password since it
demonstrates that it's DISA that seems to prevent DTMF detection when
using Zap/1. Otherwise DISA works flawlessly when calls are coming from
FXS port (TDM400), IAX, SIP channels and we have absolutely not
other problem detecting DTMF that we are aware of...
I see no active bug related to DISA at bugs.digium.com...
Any idea?
Ben.
*Code:*
---
zapata.conf
---
context=inbound-pstn
signalling=fxs_ks
rxgain=10
txgain=3
language=fr
channel = 1
I have tried to change gains without any result ...
(http://forums.digium.com/viewtopic.php?t=17769highlight=disa+dtmf)
; --- Config A --- ;
*Code:*
exten = 111,1,Answer
exten = 111,n,Authenticate(111)
exten = 111,n,DISA(no-password|internal)
; --- Dial sequence --- ;
*Code:*
PSTN line - TDM400
enter extension 111 - dial tone
enter password 111 - new dial tone
enter extension - I still getting the dial tone whatever I'm entering
timeout.
Here the debug log:
*Code:*
snip
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF end accepted without begin '1' on Zap/1-1
DTMF end passthrough '1' on Zap/1-1
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
Oooh, got something to jump out with ('1')!
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
== CDR updated on Zap/1-1
Launching 'Answer'
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack
Launching 'Authenticate'
-- Executing [EMAIL PROTECTED]:2] Authenticate(Zap/1-1, 111) in new
stack
Set channel Zap/1-1 to write format gsm
Scheduling timer at 160 sample intervals
-- Zap/1-1 Playing 'agent-pass' (language 'fr')
Scheduling timer at 0 sample intervals
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: # on Zap/1-1
DTMF end '#' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '#' with duration 100 queued on Zap/1-1
DTMF end emulation of '#' queued on Zap/1-1
Set channel Zap/1-1 to write format gsm
Scheduling timer at 160 sample intervals
-- Zap/1-1 Playing 'auth-thankyou' (language 'fr')
Scheduling timer at 0 sample intervals
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
Launching 'DISA'
-- Executing [EMAIL PROTECTED]:3] DISA(Zap/1-1,
no-password|internal) in new stack
Digittimeout: 3000
Responsetimeout: 1
Mailbox:
Context: internal
DISA no-password login success
Set channel Zap/1-1 to write format slin
Scheduling timer at 160 sample intervals
Scheduling timer at 0 sample intervals
[ asterisk isn't detecting any DTMF... -- ]
DISA extension entry timeout on chan Zap/1-1
Requested indication 8 on channel Zap/1-1
Set channel Zap/1-1 to write format ulaw
Scheduling timer at 0 sample intervals
Spawn extension (compagnie,111,3) exited non-zero on 'Zap/1-1'
== Spawn extension (compagnie, 111, 3) exited non-zero on 'Zap/1-1'
Soft-Hanging up channel 'Zap/1-1'
Hanging up channel 'Zap/1-1'
zt_hangup(Zap/1-1)
Hangup: channel: 1 index = 0, normal = 7, callwait = -1, thirdcall = -1
disabled echo cancellation on channel 1
Set option TDD MODE, value: OFF(0) on Zap/1-1
Updated conferencing on 1, with 0 conference users
-- Hungup 'Zap/1-1'
snip
; --- Config B --- ;
*Code:*
exten = 111,1,Answer
exten = 111,n,DISA(111|internal)
; --- Dial sequence --- ;
*Code:*
PSTN line - TDM400
enter extension 111 - dial tone
enter password 111 - I still getting the dial tone whatever I'm entering
password timeout.
Here the debug log:
*Code:*
snip
DTMF digit: 1 on Zap/1-1
DTMF end
Re: [asterisk-users] DECT SIP phones
On Thursday 13 September 2007 19:05:51 Stephen Bosch wrote: I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some annoying issues anyway. Can anyone suggest a solid alternative DECT SIP phone that is available in North America? I don't know how solid you would consider them, but I have repurposed the ATS X10001P phones that are sold for use with Lingo into phones that can be used with Asterisk. At $70US, I suspect they are the least expensive SIP DECT phones available. http://asterisk.drunkcoder.com/hacks/ats-config/ -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mutipoint Conferencing?
On 14 Sep 2007, at 13:45, William Stillwell (Ki4swy) wrote: I am trying to determine what would need to be done/modified to enable the following: I have a SIP extension come into my asterisk box, and I then need it to call 6-10 remote Sip Stations that are set to Auto-Answer... (note, my remote sip stations are actually cisco h323 devices, I can call them fine from any softphone, or other device, and have full-duplex audio, however, i need to be able to conference bring all the remote stations automatically.w/Full duplex audio. Or if someone could direct me to a list that would actually be able to answer this question.. Oddly enough I've just done a quick hack like that. Basically, the dialplan for the incoming call execs System(/usr/ local/bin/mix) then drops it into a meetme. Mix is a shell script that puts call files into the asterisk spool directory. The spool files dial the autoanswer sip devices and drop them into the same meetme Drop me a mail offlist if you need help. Tim. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk voice quality tuning
I will try to answer it this way: G.711 is toll quality voice, if everything is functioning properly should be almost identical to a regular phone call. You will need to do trouble shooting to (in the words drilled into me by an old boss): isolate, identify and quantify the issue. I would start by setting up a record/playback extension, call it from PSTN and call it from the SIP phones, see where the noise is being introduced, from there could be hundreds of different things (LAN congestion, interrupt sharing on the PSTN card, bad wiring, faulty switch, etc). So to answer your question there isn't a parameter that says Noise=Yes/No, you need to: isolate, identify and quantify the noise. On 9/14/07, satish patel [EMAIL PROTECTED] wrote: I have both type of call sip-2-pstn and pstn-2 -sip but quality is not good so how to check asterisk voice quality and codec quality i am useing G.711 alaw and ulaw and it is my LAN network so is there any specific perameter or option to improve quality of voice ??? *Adrian Marsh [EMAIL PROTECTED]* wrote: Satish, Whats your network setup? Do you get bad quality on internal-asterisk calls, or only on external calls? Are you making pure IP calls (sip2sip), or are there E1/T1 cards involved? What codecs are you currently using? What devices are you using? Adrian Marsh From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of satish patel Sent: 14 September 2007 06:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk voice quality tuning Dear all I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Catch up on fall's hot new showshttp://us.rd.yahoo.com/tv/mail/tagline/falltv/evt=47093/*http://tv.yahoo.com/collections/3658+%0Aon Yahoo! TV. Watch previews, get listings, and more! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to determine if a SIP extension has DNDonoroff
On 9/14/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Anthony Francis wrote: When a device is called and it is in CFWD mode it sends back a redirect message (Moved Temporarily), Asterisk displays in the CLI Recieved Moved Temporarily trying XX thanks to XXX.XXX.XXX.XXX or something along those lines. This helps you know what is in SIP messages: http://www.ietf.org/rfc/rfc3263.txt But there is no method to QUERY the CFWD status. I understand what happens when a call goes to a SIP device with CFWD enabled. Isn't it possible to start call and then just cancel it, right before media? I don't know internals of SIP, but i think this would be possible. Of course - there is chance that phone couldn't support it and would crash or start some unexpected behavior. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent Callback Login in 1.4
On 9/13/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: It shouldn't be that hard to translate the AEL example into traditional dialplan language; in fact, Asterisk does that itself when you load the AEL into memory, so if you load it yourself and then do a 'dialplan show' you'll see the translated version, which you can then copy into your database. You can also use 'aelparse -w' to dump extensions.ael as extensions.ael.dumpto assist in this. The branching and labeling of priorities is designed for efficiency, not readability, so you'll have to go over it carefully to get a good feel for how AEL constructs are turned into extensions. According to Murf, one of the purposes of this switch was to allow people to write dialplan in AEL and insert it into * installations where AEL was either not supported (1.2) or not viable (GUIs, realtime, resistance to change, etc.). -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent Callback Login in 1.4
James FitzGibbon wrote: On 9/13/07, *Kevin P. Fleming* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: It shouldn't be that hard to translate the AEL example into traditional dialplan language; in fact, Asterisk does that itself when you load the AEL into memory, so if you load it yourself and then do a 'dialplan show' you'll see the translated version, which you can then copy into your database. You can also use 'aelparse -w' to dump extensions.ael as extensions.ael.dump to assist in this. The branching and labeling of priorities is designed for efficiency, not readability, so you'll have to go over it carefully to get a good feel for how AEL constructs are turned into extensions. According to Murf, one of the purposes of this switch was to allow people to write dialplan in AEL and insert it into * installations where AEL was either not supported (1.2) or not viable (GUIs, realtime, resistance to change, etc.). -- j. Thank you for the awesome help! -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help Drop Calls
hi. i have a prob hope someone has a solution for it. here is the set up local_number calls--- another_local_number --callforward-- toll free number -- enters asterisk --queue(agents) the problem is we have lots of drop calls the reason being the original caller puts down his phone before an agent answers. testing this setup we noticed that it takes around 6 rings before the agents phone rings. is the problem with asterisk or before the call enters asterisk?? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT SIP phones
On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote: Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some annoying issues anyway. Can anyone suggest a solid alternative DECT SIP phone that is available in North America? Look for the Aastra SIP-DECT solution which is available in NA. You can get very good coverage and call volume by adding internal and external access points. http://www.aastra.com/cps/rde/xchg/SID-3D8CCB73-F1EF48FA/04/hs.xsl/21410.htm -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Drop Calls
Yeah you can do nothing about the routing time out on the PSTN, and there is always a bit of processing time when a call enters the queue. paul aldee wrote: hi. i have a prob hope someone has a solution for it. here is the set up local_number calls--- another_local_number --callforward-- toll free number -- enters asterisk --queue(agents) the problem is we have lots of drop calls the reason being the original caller puts down his phone before an agent answers. testing this setup we noticed that it takes around 6 rings before the agents phone rings. is the problem with asterisk or before the call enters asterisk?? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
Matthew Fredrickson wrote: shadowym wrote: Maybe his comments were taken out of context as they don't have the whole interview posted. Why is he talking about queue games, Biologicall and other extremely niche crap when there are huge holes in the basic offering (SLA and SCA)? Considering it is an open source project, anybody that has access to the source code (i.e. everybody) can work on whatever they want to, whether it be SLA, SCA, or queue games for the more light hearted. Amen, brother. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can Asterisk match a literal * in extensions.conf
I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal * followed by 1 or more numbers would match. This is not the case, and it matches any extension that starts with a number. Thank you in advance for your assistance. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Prompt for extension with standard dial-tone.
Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long dial tone and play it with Read(). Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
. matches any number of the preceding character, change it to _X.*X. Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal * followed by 1 or more numbers would match. This is not the case, and it matches any extension that starts with a number. Thank you in advance for your assistance. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.4.8 Released
Hello everyone, AstLinux 0.4.8 has been released. The only updates were to Asterisk and Zaptel. Most of the development effort is focused on implementing Asterisk 1.4 and releasing AstLinux 0.5, which should be both happen fairly soon. Expect many more changes in those releases! http://www.astlinux.org -- Kristian Kielhofner ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
I don't think * means anything special to A*k, But wouldn't it be: _X.*X. To match as you ask ? (number)(wildcard)*(number)(wildcard) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: 14 September 2007 17:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Can Asterisk match a literal * in extensions.conf I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal * followed by 1 or more numbers would match. This is not the case, and it matches any extension that starts with a number. Thank you in advance for your assistance. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to route outgoing calls on IP-level
Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to use one IP address, and others to go through another address. is there a way to achive that using asterisk ? Cheers, Kate ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
On Friday 14 September 2007 11:39:40 Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) The problem you're seeing is that the period is a short-circuit operator. It says if you match everything so far and at least one more character, then you have a match, no need to go any further. You CANNOT match past a '.'. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH Files Volume
Is there a way to decrease the volume on the native files version of MOH in 1.4? I've had several people complain that it is too loud. Peder ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype + Asterisk
Did you got a response for your questions? Recently found this URL in Google SiSky http://www.yeastar.com/ProductsforAsterisk.asp Regards, Alejandro Lengua On 9/6/07, John Meksavan [EMAIL PROTECTED] wrote: Has anybody ever integrated Skype with Asterisk? If you have, which software would you recommend to accomplish such a task? ChanSkype? And how reliable are the calls? Did the DTMF tones work? Thanks in advance. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing call restriction in extention.conf
The best way of restricting users from STD is making different context in
extensions.conf, in that context allow STD. and in sip.conf for those users
make that context.
extensions.conf
[local]
exten = _0[1-9].,1,Answer
exten = _0[1-9].,2,Dial(${TRUNK}/${EXTEN:1}w)
exten = _0.,3,Hangup
[STD]
exten = _0.,1,Answer
exten = _0.,2,Dial(${TRUNK}/${EXTEN:1}w)
exten = _0.,3,Hangup
include = local
and sample sip.conf---
if User 101 is allowed for Local only and if 102 is for STD also
[101]
type=friend
username=101
;secret=101
host=dynamic
port=5060
dtmfmode=rfc2833
canreinvite=no
context=local
disallow=all
allow=ulaw
[102]
type=friend
username=102
;secret=102
host=dynamic
port=5060
dtmfmode=rfc2833
canreinvite=no
context=STD
disallow=all
allow=ulaw
--
--
You can also use Authenticate command in your dial plan for password
authenticate if you want.
Regards,
Kesh
satish patel [EMAIL PROTECTED] wrote: Dear all
I have asterisk PBX and 100 endpoint i want to block STD for
specific users or password protect so is it possible users can set passwd on
his/her phone and password automaticaly reflacted on asterisk in short i want
to restrict STD call of users of outgoing
Regards
satish patel
-
Moody friends. Drama queens. Your life? Nope! - their life, your story.
Play Sims Stories at Yahoo! Games.
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Kesh
Lets change the future...lets change the world.
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Re: [asterisk-users] MOH Files Volume
Peder @ NetworkOblivion wrote: Is there a way to decrease the volume on the native files version of MOH in 1.4? I've had several people complain that it is too loud. run the files through sox -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: . matches any number of the preceding character, change it to _X.*X. That still won't help. Once the Asterisk pattern matching parser sees a period in the pattern, it ignores anything after it. (I'm not exactly happy about that, but that's the way it is.) In short, Asterisk doesn't currently have a good way of handling this situation. Hopefully somebody infinitely smarter than I am will take pity on our plight and give us a some more advanced pattern-matching tools. (Hint, hint) -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prompt for extension with standard dial-tone.
On Fri, 2007-09-14 at 19:49 +0300, Atis wrote: What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. Does the DISA() application do what you want? -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to route outgoing calls on IP-level
On Fri, 14 Sep 2007, Kate Kretz wrote: Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to use one IP address, and others to go through another address. is there a way to achive that using asterisk ? I doubt it, but in any case, you really ought to do it at the Linux routing level. And it might well happen that it happens automatically, anyway. In the absence of anything otherwise, Linux will pick the right interface for the network that interface is pointing to. Unless you've got something really weird that is, in which case you need a networking guru and not an asterisk guru :) Give us more details and see... Gordon ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
i just met the same problem. i want to match extension that end with a number, but can not find a way. i also found that _.X match all extension, but won't match any caller-id number in dialplan. maybe it is a bug. but it seems not important since _.X is useless anyway. 2007/9/15, Tilghman Lesher [EMAIL PROTECTED]: On Friday 14 September 2007 11:39:40 Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) The problem you're seeing is that the period is a short-circuit operator. It says if you match everything so far and at least one more character, then you have a match, no need to go any further. You CANNOT match past a '.'. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to route outgoing calls on IP-level
Kate Kretz wrote: Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to use one IP address, and others to go through another address. is there a way to achive that using asterisk ? Cheers, Kate This is the job of your network, not Asterisk. Policy-based routing is not much fun (unless you think the Cisco CLI is really cool) but it can be done. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype + Asterisk
Alejandro, Thanks for replying. I did come by this website before. I was just wandering, if anybody actually tried Skype with Asterisk. My experimentation with the Sip Protocol and Asterisk is at end because I could never get QOS with any sip provider, ie Broadvoice, Vitelity, and Teliax, when connecting directly to the General Internet. In my past experience, Skype has been the only VOIP that works very well. If I could just integrate this with my Asterisk at work, it would really make my boss happy. From: Alejandro Lengua [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Skype + Asterisk Date: Fri, 14 Sep 2007 13:02:19 -0500 Did you got a response for your questions? Recently found this URL in Google SiSky http://www.yeastar.com/ProductsforAsterisk.asp Regards, Alejandro Lengua On 9/6/07, John Meksavan [EMAIL PROTECTED] wrote: Has anybody ever integrated Skype with Asterisk? If you have, which software would you recommend to accomplish such a task? ChanSkype? And how reliable are the calls? Did the DTMF tones work? Thanks in advance. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get the device you want, with the Hotmail® you love. http://www.microsoft.com/windowsmobile/mobilehotmail/default.mspx?WT.mc_ID=MobileHMTagline ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX
On Thu, Sep 13, 2007 at 11:55:59PM -0700, Vieri wrote: Thank you, I did what you mentioned below. It seems that I'm getting a hangupcause of 0 which I believe is not defined. Is Alcatel the first party that is trying to disconnect or is it Asterisk? (Not sure how to interpret the debug info I'm posting below) Whether it's Alcatel or Asterisk, what could be the actual cause? (or where should I start looking?) Thanks INF-VOIP*CLI pri debug span 1 Enabled debugging on span 1 -- Executing NoOp(SIP/4053-083189e8, [ALCATEL TEST] Start) in new stack -- Executing Dial(SIP/4053-083189e8, Zap/g1/5900) in new stack 1 -- Making new call for cr 32781 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (8) len=32 1 Call Ref: len= 2 (reference 13/0xD) (Originator) 1 Message type: SETUP (5) 1 [04 03 80 90 a3] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [18 04 e9 81 83 81] 1 Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 1 ChanSel: Reserved 1Ext: 1 DS1 Identifier: 1 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 1 ] 1 [6c 06 21 80 34 30 35 33] 1 Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number not screened (0) '4053' ] 1 [70 05 a1 35 39 30 30] 1 Called Number (len= 7) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5900' ] 1 [a1] 1 Sending Complete (len= 1) -- Called g1/5900 1 Protocol Discriminator: Q.931 (8) len=10 1 Call Ref: len= 2 (reference 13/0xD) (Terminator) 1 Message type: CALL PROCEEDING (2) 1 [18 03 a9 83 81] 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 1 ChanSel: Reserved 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 1 ] 1 -- Processing IE 24 (cs0, Channel Identification) -- Zap/1-1 is proceeding passing it to SIP/4053-083189e8 1 Protocol Discriminator: Q.931 (8) len=5 1 Call Ref: len= 2 (reference 13/0xD) (Terminator) 1 Message type: ALERTING (1) This is normal, right? -- Zap/1-1 is ringing -- Zap/1-1 is busy Huh? why is Zap/1-1 suddenly busy? 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered, peerstate Call Received 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 13/0xD) (Originator) 1 Message type: DISCONNECT (69) 1 [08 02 81 90] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] So your side initiation disconnecting? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to route outgoing calls on IP-level
Check the route command on your Linux system. The gateway route should be the ethX and network whatever you want. At 01:41 p.m. 14/09/2007, Drew Gibson wrote: Kate Kretz wrote: Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to use one IP address, and others to go through another address. is there a way to achive that using asterisk ? Cheers, Kate This is the job of your network, not Asterisk. Policy-based routing is not much fun (unless you think the Cisco CLI is really cool) but it can be done. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users RafaelCanchola Product Development Engineer, FonetGlobal Inc. [EMAIL PROTECTED] http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztdummy kills audio
I'm running asterisk/zaptel 1.4.5. If I load the ztdummy module, the dialplan hangs when it tries to play audio (i.e. Playback) -- and I just hear static on the line. I'm running this on a debian system. I actually have it working on a different debian system but have yet to discover the important difference between the two installations. Any advice on where to look? Thanks John ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy kills audio
On Fri, Sep 14, 2007 at 03:05:49PM -0400, John Albano wrote: I'm running asterisk/zaptel 1.4.5. If I load the ztdummy module, the dialplan hangs when it tries to play audio (i.e. Playback) -- and I just hear static on the line. I'm running this on a debian system. I actually have it working on a different debian system but have yet to discover the important difference between the two installations. What is the output from running: zttest -v for about a minute? (press ctrl-C to stop) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy kills audio
Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 On Fri, Sep 14, 2007 at 03:05:49PM -0400, John Albano wrote: I'm running asterisk/zaptel 1.4.5. If I load the ztdummy module, the dialplan hangs when it tries to play audio (i.e. Playback) -- and I just hear static on the line. I'm running this on a debian system. I actually have it working on a different debian system but have yet to discover the important difference between the two installations. What is the output from running: zttest -v for about a minute? (press ctrl-C to stop) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging to external speaker like in airports etc...
Thats AMAZING! This google you have shown me is truly a modern marvel
of the interwebs.
You know what would be EVEN BETTER though?
If idiots (such as you and I) would find something better to do with our
time than mock others on mailing lists in a pitiful attempt to appear
more knowledgeable/cool/hip/popular/what have you.
Nobody likes you or thinks you're pretty or will ask you to prom.
Welcome to the asshole club, we get badges... and black eyes.
Lacy Moore - Aspendora wrote:
On 9/13/07, *Deepak Naidu* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi, I have a production asterisk-1.2.8 system with FreePBX PRI
Digium card.
I am looking for a paging system to an external speaker. I can
page to internal Polycom 501 VoIP.
But, what hardware or system do I need to integrate with the
asterisk to have this acheived.
You know what would be even better? If we had a search engine that
you could type something into and it would produce a list of pages
related to this.
Oh wait, maybe that's what this does:
http://www.google.com/search?hl=enq=Asterisk+paging
http://www.google.com/search?hl=enq=Asterisk+paging
Google is a wonderful tool, learn to use it...
--
Deepak
*Linux your Life,** Don't Window it [[]] *
*{ All for the best }*
Yahoo! Answers - Get better answers from someone who knows. Try it
now
http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc2VjA21haWwEc2xrA3RhZ2xpbmU.
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Re: [asterisk-users] Paging to external speaker like in airports etc...
Anthony Kepler wrote:
Thats AMAZING! This google you have shown me is truly a modern marvel
of the interwebs.
You know what would be EVEN BETTER though?
If idiots (such as you and I) would find something better to do with our
time than mock others on mailing lists in a pitiful attempt to appear
more knowledgeable/cool/hip/popular/what have you.
Nobody likes you or thinks you're pretty or will ask you to prom.
Welcome to the asshole club, we get badges... and black eyes.
Unfortunately, that seems to be more and more the way of the world,
though I will say that this kind of unproductive attitude and
intolerance of others is more prevalent on these kinds of computer
related lists than some other lists.
Something about being able to deal better with inanimate objects than humans
Peg Leg O'Brien
Lacy Moore - Aspendora wrote:
On 9/13/07, *Deepak Naidu* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi, I have a production asterisk-1.2.8 system with FreePBX PRI
Digium card.
I am looking for a paging system to an external speaker. I can
page to internal Polycom 501 VoIP.
But, what hardware or system do I need to integrate with the
asterisk to have this acheived.
You know what would be even better? If we had a search engine that
you could type something into and it would produce a list of pages
related to this.
Oh wait, maybe that's what this does:
http://www.google.com/search?hl=enq=Asterisk+paging
http://www.google.com/search?hl=enq=Asterisk+paging
Google is a wonderful tool, learn to use it...
--
Deepak
*Linux your Life,** Don't Window it [[]] *
*{ All for the best }*
Yahoo! Answers - Get better answers from someone who knows. Try it
now
http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc2VjA21haWwEc2xrA3RhZ2xpbmU.
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--
Dog is my co-pilot
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[asterisk-users] g729 on 1.4.10.1
I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question. I even attempted to setup the phone with only the allow=g729, but in that instance it won't even complete the call. We had to add g711 support to the gateway in question for now to get it up and running, but we want to get it back to using only g729. CLI show modules like g729 Module Description Use Count format_g729.so Raw G729 data 0 codec_g729a.so Annex A/B (floating point) G.729 Codec ( 0 2 modules loaded I downloaded the pre-compiled g729 module from Digium. The sip.conf references g729 and the codec module is loaded. Unless there's anything else I need to do that I'm missing? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Force a new user to configure Comedian mail?
In Asterisk 1.4, is there any way to force new users to configure their mailbox? I'm thinking a simple IVR that holds a user's hand through changing their PIN, recording their name, and setting up one or both greetings, the very first time they use the account. I know I can publish docs that tell them how to use the 0 menu and do this by hand... but users are lazy and resent documentation. Thanks! Jeremy Wadhams Yahoo Inc ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy kills audio
Yes, that was after approx a minute. Output from lsmod is... zaptel182948 4 zttranscode,ztdummy crc_ccitt 3072 1 zaptel On Fri, Sep 14, 2007 at 03:32:00PM -0400, John Albano wrote: Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 Is this after a minute? What is the output of: lsmod | grep zaptel ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging to external speaker like in airports etc...
At 01:11 PM 9/14/2007, you wrote: Unfortunately, that seems to be more and more the way of the world, though I will say that this kind of unproductive attitude and intolerance of others is more prevalent on these kinds of computer related lists than some other lists. Something about being able to deal better with inanimate objects than humans Once upon a time it cost $20/hr over a 9600 baud link to read stuff like this and people tended to think before they asked questions, now there is no barrier and anyone is allowed in to ask anything they want. Soon the inane questions become so prevalent that all the people who know the answers go away as the learning experience goes away. Ira ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy kills audio
On Fri, Sep 14, 2007 at 03:32:00PM -0400, John Albano wrote: Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 Is this after a minute? What is the output of: lsmod | grep zaptel -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
On Friday 14 September 2007 12:37:11 pm Tilghman Lesher wrote: On Friday 14 September 2007 11:39:40 Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) The problem you're seeing is that the period is a short-circuit operator. It says if you match everything so far and at least one more character, then you have a match, no need to go any further. You CANNOT match past a '.'. Thank you all. I knew I wasn't nuts, but this is the infomation being posted at http://freenum.org/cookbook/ I'll just have to add a prefix. I was hoping to avoid that. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI script fails on IAX channels (from call file).
Hi Guys,
I have already tried this one on the developers list. I have not been
successful getting much back there and I have notified them that I will
post this on the users list instead. Hopefully somebody have tried
something similar and can help out.
I am developing AGI scripts on Asterisk and have run into some very
strange behaviour and I think this is a bug, but I am not completely
sure. Any suggestions are highly appreciated.
Let me first state the ground here
Facts
1. I am on asterisk 1.4.11
2. What I am trying to do works on SIP phones and SIP channels
3. What I am trying to do FAILS on IAX phone (iaxy) and IAX channels
Having stated this let's now only focus on the IAX channel. To avoid
lengthy code reading I will state the problem first and then later the code.
I have an AGI scrip that takes a single input parameter. You can call it
from the dial plan like
exten = *66,2,AGI(test.agi|670507)
This AGI script starts with a SAY DIGITS on the parameter 670507.
Then it gives you a choice with some STREAM FILE and finally a GET
DATA. Once you have made your choice the AGI script tells you what you
chose and hangs up.
That's it. Really handy little script, right. This is obviously made
just to demonstrate the problem I am having. Note again, only for IAX.
Problem
1. This does work when the the IAX based phone executes the script
from the dial plan. Then there is no problem what so ever replying
to with DTMF from the phone.
2. This DOES NOT work if I execute the AGI script from a call file. I
get the phone call to the IAX based phone and the streams work
fine. I just CANNOT reply with DTMF in the GET DATA part. It
just times out.
Tracing with
iax2 set debug on
Reveals that I get a /Rx-Frame Retry[ No] -- OSeqno: 006 ISeqno: 004
Type: DTMF_E/ every time I press a key in both cases (execution from
the dial plan or call file), but it times out when it is executed from a
call file.
Now some data: I am sorry if there is some garbage in the traces. I have
tried to cut out stuff that shouldn't be of any concern, but I was
scared to cut too much.
Call File
=== Call File ==
channel: Local/[EMAIL PROTECTED]
maxretries: 3
retrytime: 60
waittime: 60
callerid: Test *66
application: AGI
data: test.agi|670507
= End Call File
Perl AGI Scrip
test.agi =
#!/usr/bin/perl
use strict;
use Time::Local;
$|=1;
# Setup some variables
my %AGI; my $DEBUG=1;
my $DEBUGOUT = filehandle;
my $debugfile=/tmp/agi_debug.log;
check_result ##
# Use this to check the result of #
# a sent command #
# I pretty much stole this from#
# the regular agi-test.agi #
sub checkresult {
my ($res) = @_;
my $retval;
chomp $res;
if ($res =~ /^200/) {
$res =~ /result=(-?\d+)/;
if (!length($1)) {
print DEBUGOUT FAIL ($res)\n;
exit(1);
} elsif ($DEBUG=1){
print DEBUGOUT PASS ($1)\n;
}
} else {
print STDERR FAIL (unexpected result '$res')\n;
exit(1);
}
}
send_file #
# Use this to send a wave file on #
# the channel #
# #
sub send_file {
my ($myfile) = @_;
chomp($myfile);
if ($DEBUG == 1 ) {
print DEBUGOUT Sending stream $myfile \n;
}
print STREAM FILE $myfile \0123456789\\n;
my $result = STDIN;
checkresult($result);
}
hangup ###
# Use this to hand up a channel#
# the channel #
# #
sub hangup {
if ($DEBUG == 1 ) {
print DEBUGOUT Hanging up \n;
}
print HANGUP \\ \n;
my $result = STDIN;
checkresult($result);
}
say_digits
# Use this to say a digits #
# over the channel #
# #
sub say_digits {
my ($mynumber) = @_;
chomp($mynumber);
if ($DEBUG == 1 ) {
print DEBUGOUT Saying digits $mynumber \n;
}
print SAY DIGITS $mynumber \0123456789\\n;
my $result = STDIN;
checkresult($result);
}
get_data ##
# Feed with (file, maxnumbers) #
# where file is the sound file #
# to be played and maxnumbers is #
# the maximum amount of digits to #
# allow in the answer #
sub get_data {
my @mydata = @_;
my $myfile = $mydata[0];
my $mymax = $mydata[1];
if ($DEBUG == 1 ) {
print DEBUGOUT Getting data \n;
}
print GET DATA $myfile 15000 $mymax \n;
my $result = STDIN;
checkresult($result);
$result =~ /result=(-?\d+)/;
return $1;
}
Re: [asterisk-users] Prompt for extension with standard dial-tone.
On 9/14/07, Jared Smith [EMAIL PROTECTED] wrote: On Fri, 2007-09-14 at 19:49 +0300, Atis wrote: What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. Does the DISA() application do what you want? Thanks, from description it would give me exactly what i want. I will try it on Monday :) Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a new user to configure Comedian mail?
Jeremy Wadhams wrote: In Asterisk 1.4, is there any way to force new users to configure their mailbox? I'm thinking a simple IVR that holds a user's hand through changing their PIN, recording their name, and setting up one or both greetings, the very first time they use the account. I know I can publish docs that tell them how to use the 0 menu and do this by hand... but users are lazy and resent documentation. Thanks! Jeremy Wadhams Yahoo Inc In the sample voicemail.conf file you should find this section: forcename=yes ; Forces a new user to record their name. A new user is ; determined by the password being the same as ; the mailbox number. The default is no. forcegreetings=yes ; This is the same as forcename, except for recording ; greetings. The default is no. If these are set to yes and the user's voicemail password is set to their mailbox number, then the next time they enter the voicemail box it will ask them to record their name, greetings and change their password. NOTE Make sure you tell them NOT to set their new password to their extension when they reset it. They will end up going through all these steps the next time they enter their mailbox :) I forgot to tell a couple users this and I got a call asking why they had to record their greetings every time they went into their voicemail. -Dave ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a new user to configure Comedian mail?
On 9/14/07, Jeremy Wadhams [EMAIL PROTECTED] wrote: In Asterisk 1.4, is there any way to force new users to configure their mailbox? I'm thinking a simple IVR that holds a user's hand through changing their PIN, recording their name, and setting up one or both greetings, the very first time they use the account. If your pin is equal to your mailbox, VoiceMailMain() does this automatically when you log in. -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a new user to configure Comedian mail?
Jeremy Wadhams wrote: I know I can publish docs that tell them how to use the 0 menu and do this by hand... but users are lazy and resent documentation. As are Asterisk administrators (sometimes) :-) See the 'forcename' config option in voicemail.conf.sample. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a new user to configure Comedian mail?
Jeremy Wadhams wrote: In Asterisk 1.4, is there any way to force new users to configure their mailbox? I'm thinking a simple IVR that holds a user's hand through changing their PIN, recording their name, and setting up one or both greetings, the very first time they use the account. Yep. In fact, it was one of the first patches I ever wrote for Asterisk. :) Here are the relevant options from voicemail.conf: ; forcename=yes ; Forces a new user to record their name. A new user is ; determined by the password being the same as ; the mailbox number. The default is no. ; forcegreetings=no ; This is the same as forcename, except for recording ; greetings. The default is no. -- Russell Bryant Software Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal * followed by 1 or more numbers would match. This is not the case, and it matches any extension that starts with a number. Thank you in advance for your assistance. . must ONLY be the LAST character in a pattern match. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
On 9/14/07, Jared Smith [EMAIL PROTECTED] wrote:
On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote:
. matches any number of the preceding character, change it to _X.*X.
That still won't help. Once the Asterisk pattern matching parser sees a
period in the pattern, it ignores anything after it. (I'm not exactly
happy about that, but that's the way it is.) In short, Asterisk doesn't
currently have a good way of handling this situation. Hopefully
somebody infinitely smarter than I am will take pity on our plight and
give us a some more advanced pattern-matching tools. (Hint, hint)
Well, you can have some 10 or so patterns (how long can the number
before be), with X, as X means one digit..
For example:
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XXX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XXX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
[default-wildcard]
exten = _X.,1,Macro(whatever)
Regards,
Atis
--
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org
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Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
Jared Smith wrote: On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: . matches any number of the preceding character, change it to _X.*X. That still won't help. Once the Asterisk pattern matching parser sees a period in the pattern, it ignores anything after it. (I'm not exactly happy about that, but that's the way it is.) In short, Asterisk doesn't currently have a good way of handling this situation. Hopefully somebody infinitely smarter than I am will take pity on our plight and give us a some more advanced pattern-matching tools. (Hint, hint) Asterisk's pattern matching is NOT a regex. . means match 1 or more character. It has nothing to do with the preceding characters and must ALWAYS be the last character in a pattern match. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a new user to configure Comedian mail?
Jeremy Wadhams wrote: In Asterisk 1.4, is there any way to force new users to configure their mailbox? I'm thinking a simple IVR that holds a user's hand through changing their PIN, recording their name, and setting up one or both greetings, the very first time they use the account. You can use the forcename and forcegreetings settings to get this behavior. The way to let the voicemail system know the user is a new user is to to set the mailbox number and password the same for that user. If you do this, then the first time the person calls VoiceMailMain(), they will be walked through the process of changing their PIN, recording their name, and their greetings. Mark Michelson ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a new user to configure Comedian mail?
Jeremy Wadhams wrote: In Asterisk 1.4, is there any way to force new users to configure their mailbox? I'm thinking a simple IVR that holds a user's hand through changing their PIN, recording their name, and setting up one or both greetings, the very first time they use the account. I know I can publish docs that tell them how to use the 0 menu and do this by hand... but users are lazy and resent documentation. Yes. See the voicemail.conf.sample which is included in the asterisk source directory. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 3 way Calling
Hello, I have recently installed the TrixBOX CE 2.2.4. How can I make calls and use the 3 way calling? can it be done with any IP phone or softphone? should I do any special configuration on TrixBox? Regards, Seysan ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a new user to configure Comedian mail?
Thanks for the tip, all! I forgot that the sample .confs are as much a source of documentation as voip-info.org --JW -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Friday, September 14, 2007 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Force a new user to configure Comedian mail? Jeremy Wadhams wrote: I know I can publish docs that tell them how to use the 0 menu and do this by hand... but users are lazy and resent documentation. As are Asterisk administrators (sometimes) :-) See the 'forcename' config option in voicemail.conf.sample. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
shadowym wrote: Yes thank you for reminding me it is open source. Thank you for reminding me people can write their own code for it. I'll get right on rewriting the entire sip code. Should only take me a few hours. Including a couple hours to learn how to write c code. How hard can it be! I can't tell whether you're intending to prove the point that was being made, or trying to be sarcastic. Knowing your posting history, I'll assume the latter. But in case you're serious, and you really do believe the coders owe you something, here's another translation of the situation: If you code, if you contribute to the coding effort by intense testing and/or filing bug reports, if you carry Red Bull to the programmers during hacking sessions, etc., then--in the vernacular of the Church of the Subgenius--you buy slack. And once you have slack, you can say, Let's do this, or Let's do that, and the developers will consider it and--maybe--implement it. When, instead, you are 100% slack-free and have been noted before nipping nasty mots at the hands that feed you code, the chances of having your tart remarks about SLA taken seriously are pretty slim. But, and here's the point: It's Open Source. If the developers look the other way when you ask for something, if they don't answer your emails, if they don't drop everything when you demand something and do what you want, FORK IT! Take the code THEY they wrote and do with it what you will. It's free. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI script fails on IAX channels (from call file).
Jonas Arndt wrote: Hi Guys, I have already tried this one on the developers list. I have not been successful getting much back there and I have notified them that I will post this on the users list instead. Hopefully somebody have tried something similar and can help out. I am developing AGI scripts on Asterisk and have run into some very strange behaviour and I think this is a bug, but I am not completely sure. Any suggestions are highly appreciated. Let me first state the ground here Facts 1. I am on asterisk 1.4.11 2. What I am trying to do works on SIP phones and SIP channels 3. What I am trying to do FAILS on IAX phone (iaxy) and IAX channels *snipped when i was doing some testing i had to use READ, and because i did then i had to use GET VARIABLE to retrieve the inputted dtmf. http://dynx.net/ASTERISK/gnudialer/agiIVR.agi search for sub question_1 to see how i did it. i hope this helps. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a new user to configure Comedian mail?
Russell Bryant wrote: Jeremy Wadhams wrote: Yep. In fact, it was one of the first patches I ever wrote for Asterisk. :) And under 1.2 it can be easily bypassed. After the password is changed, if the user hangs up, the next time they call into the voice mail system, it doesn't continue to force them to do the recordings. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
On Friday 14 September 2007 15:35:47 Anthony Messina wrote:
On Friday 14 September 2007 12:37:11 pm Tilghman Lesher wrote:
On Friday 14 September 2007 11:39:40 Anthony Messina wrote:
I am working on getting freenum.org ISN/ITAD numbers integrated into my
exiting dialplan however I am having trouble getting the extension
matches to work as expected.
I would like to be able to do something like:
exten = _X.*.,1,Macro(isn-outbound...)
The problem you're seeing is that the period is a short-circuit operator.
It says if you match everything so far and at least one more character,
then you have a match, no need to go any further. You CANNOT match past
a '.'.
Thank you all. I knew I wasn't nuts, but this is the infomation being
posted at http://freenum.org/cookbook/
I'll just have to add a prefix. I was hoping to avoid that.
exten = _X.,1,Set(firstpart=${CUT(EXTEN,*,1)})
exten = _X.,n,Set(secondpart=${CUT(EXTEN,*,2)})
exten = _X.,n,GotoIf($[${LEN(${secondpart})}=0]?i,1)
exten = _X.,n,Macro(foo,${firstpart},${secondpart})
--
Tilghman
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[asterisk-users] Zaptel ztdummy module causes playback to fail
I'm using asterisk 1.4.11 and Zaptel version 1.4.5.1 with kernel 2.6.22. I have the ztdummy module loaded, which is using zaptel and rtc. When the ztdummy module is loaded, sounds are not heard when using the asterisk background command. When the ztdummy module is unloaded, which also removes zaptel and rtc, sounds are heard. I've also tested this under kernel 2.6.21 with the same results. The zttest program reports an error when ztdummy and associated modules are not present, and hangs when they are loaded. This is an AMD Athlon(tm) 64 X2 Dual Core Processor 4200+ with 2GB ram. Any idea of what is causing this problem and how it can be solved? Chris -- Chris Nestrud Email: [EMAIL PROTECTED] http://www.panix.com/~ccn/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy kills audio
On Fri, Sep 14, 2007 at 04:41:22PM -0400, John Albano wrote: Yes, that was after approx a minute. Output from lsmod is... zaptel182948 4 zttranscode,ztdummy crc_ccitt 3072 1 zaptel What release of Debian is it? What kernel do you use? Packaged or self-built? uname -a -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a new user to configure Comedian mail?
On Fri, Sep 14, 2007 at 05:04:09PM -0400, Dave Fullerton wrote: Jeremy Wadhams wrote: In Asterisk 1.4, is there any way to force new users to configure their mailbox? I'm thinking a simple IVR that holds a user's hand through changing their PIN, recording their name, and setting up one or both greetings, the very first time they use the account. I know I can publish docs that tell them how to use the 0 menu and do this by hand... but users are lazy and resent documentation. Thanks! Jeremy Wadhams Yahoo Inc In the sample voicemail.conf file you should find this section: forcename=yes ; Forces a new user to record their name. A new user is ; determined by the password being the same as ; the mailbox number. The default is no. forcegreetings=yes ; This is the same as forcename, except for recording ; greetings. The default is no. If these are set to yes and the user's voicemail password is set to their mailbox number, then the next time they enter the voicemail box it will ask them to record their name, greetings and change their password. NOTE Make sure you tell them NOT to set their new password to their extension when they reset it. They will end up going through all these steps the next time they enter their mailbox :) I forgot to tell a couple users this and I got a call asking why they had to record their greetings every time they went into their voicemail. I believe those will break with configuration generated by the asterisk-gui , as it defaults to a constant password. Right? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
On Sat, 2007-09-15 at 00:12 +0300, Atis wrote:
On 9/14/07, Jared Smith [EMAIL PROTECTED] wrote:
On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote:
. matches any number of the preceding character, change it to _X.*X.
That still won't help. Once the Asterisk pattern matching parser sees a
period in the pattern, it ignores anything after it. (I'm not exactly
happy about that, but that's the way it is.) In short, Asterisk doesn't
currently have a good way of handling this situation. Hopefully
somebody infinitely smarter than I am will take pity on our plight and
give us a some more advanced pattern-matching tools. (Hint, hint)
Well, you can have some 10 or so patterns (how long can the number
before be), with X, as X means one digit..
For example:
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XXX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XXX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
Atis--
People are spoiled by regex's, and they want to able to make a match vs.
something I call trailing context. What they don't realize is that
such
matches take (possibly) large amounts of time to complete, because they
loop
or are recursive, depending on the implementation.
Thus, a regex like X+\* (which would mean 1 or more X's followed by
an asterisk. would expand out to the 10 (actually perhaps many more)
lines above-- and run (unexpectedly) slower.
The trouble is, the pattern matcher wouldn't know how long an
expression
like X+\* should be, and could generate hundreds of entries. (if the
pattern
length is limited to 256 chars, say).
It is far better to explode out the entries yourself, as you outlined
above.
You know the max size of incoming stream
murf
[default-wildcard]
exten = _X.,1,Macro(whatever)
Regards,
Atis
--
Steve Murphy
Software Developer
Digium
smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] Force a new user to configure Comedian mail?
Doug Lytle wrote: Russell Bryant wrote: Yep. In fact, it was one of the first patches I ever wrote for Asterisk. :) And under 1.2 it can be easily bypassed. After the password is changed, if the user hangs up, the next time they call into the voice mail system, it doesn't continue to force them to do the recordings. I'm ... sorry? However, it does behave exactly as is documented. It specifies that the only check it does to see if it is a new user is by password. If someone wanted to improve this so the password doesn't matter and it actually checks to see if a name and/or greetings are recorded, then that would be great. -- Russell Bryant Software Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a new user to configure Comedian mail?
Russell Bryant wrote: Doug Lytle wrote: I'm ... sorry? However, it does behave exactly as is documented. It specifies No need to be, I was just making an observation. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy kills audio
I'm seeing the problem on both etch and lenny releases. Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux What release of Debian is it? What kernel do you use? Packaged or self-built? uname -a ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT SIP phones
On Fri, 2007-09-14 at 12:00 -0500, [EMAIL PROTECTED] wrote: Date: Fri, 14 Sep 2007 09:32:35 -0500 From: Tilghman Lesher [EMAIL PROTECTED] Subject: Re: [asterisk-users] DECT SIP phones To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 On Thursday 13 September 2007 19:05:51 Stephen Bosch wrote: I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some annoying issues anyway. Can anyone suggest a solid alternative DECT SIP phone that is available in North America? I don't know how solid you would consider them, but I have repurposed the ATS X10001P phones that are sold for use with Lingo into phones that can be used with Asterisk. At $70US, I suspect they are the least expensive SIP DECT phones available. Wal-Mart sells the ATS X10001P for $55, and claims it has a fax port: http://www.walmart.com/catalog/product.do?dest=97product_id=6457851sourceid=1503142050 . Is there a way to fax with these phones without Lingo? How does Lingo do it (over the phone's Internet connection), if Asterisk can't? http://asterisk.drunkcoder.com/hacks/ats-config/ Your server seems very slow, often timing out. -- Tilghman -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISA and DTMF detection problem w/ FXO port on a TDM400
i did have same issue with DISA in 1.4 and TDM400 FXO,
I switched back to Authenticate and waitexten.
On 9/14/07, Benjamin M. [EMAIL PROTECTED] wrote:
Originally posted at http://forums.digium.com/viewtopic.php?t=18045
Hi!
I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing
DISA seems to prevent any DTMF detection capability when using the FXO
port of the TDM400.
Below, config A and B and their debug logs.
In Config A I use Authenticate() instead of using DISA password since it
demonstrates that it's DISA that seems to prevent DTMF detection when
using Zap/1. Otherwise DISA works flawlessly when calls are coming from
FXS port (TDM400), IAX, SIP channels and we have absolutely not
other problem detecting DTMF that we are aware of...
I see no active bug related to DISA at bugs.digium.com...
Any idea?
Ben.
*Code:*
---
zapata.conf
---
context=inbound-pstn
signalling=fxs_ks
rxgain=10
txgain=3
language=fr
channel = 1
I have tried to change gains without any result ...
(http://forums.digium.com/viewtopic.php?t=17769highlight=disa+dtmf)
; --- Config A --- ;
*Code:*
exten = 111,1,Answer
exten = 111,n,Authenticate(111)
exten = 111,n,DISA(no-password|internal)
; --- Dial sequence --- ;
*Code:*
PSTN line - TDM400
enter extension 111 - dial tone
enter password 111 - new dial tone
enter extension - I still getting the dial tone whatever I'm entering
timeout.
Here the debug log:
*Code:*
snip
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF end accepted without begin '1' on Zap/1-1
DTMF end passthrough '1' on Zap/1-1
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
Oooh, got something to jump out with ('1')!
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
== CDR updated on Zap/1-1
Launching 'Answer'
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack
Launching 'Authenticate'
-- Executing [EMAIL PROTECTED]:2] Authenticate(Zap/1-1, 111) in new
stack
Set channel Zap/1-1 to write format gsm
Scheduling timer at 160 sample intervals
-- Zap/1-1 Playing 'agent-pass' (language 'fr')
Scheduling timer at 0 sample intervals
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: # on Zap/1-1
DTMF end '#' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '#' with duration 100 queued on Zap/1-1
DTMF end emulation of '#' queued on Zap/1-1
Set channel Zap/1-1 to write format gsm
Scheduling timer at 160 sample intervals
-- Zap/1-1 Playing 'auth-thankyou' (language 'fr')
Scheduling timer at 0 sample intervals
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
Launching 'DISA'
-- Executing [EMAIL PROTECTED]:3] DISA(Zap/1-1,
no-password|internal) in new stack
Digittimeout: 3000
Responsetimeout: 1
Mailbox:
Context: internal
DISA no-password login success
Set channel Zap/1-1 to write format slin
Scheduling timer at 160 sample intervals
Scheduling timer at 0 sample intervals
[ asterisk isn't detecting any DTMF... -- ]
DISA extension entry timeout on chan Zap/1-1
Requested indication 8 on channel Zap/1-1
Set channel Zap/1-1 to write format ulaw
Scheduling timer at 0 sample intervals
Spawn extension (compagnie,111,3) exited non-zero on 'Zap/1-1'
== Spawn extension (compagnie, 111, 3) exited non-zero on 'Zap/1-1'
Soft-Hanging up channel 'Zap/1-1'
Hanging up channel 'Zap/1-1'
zt_hangup(Zap/1-1)
Hangup: channel: 1 index = 0, normal = 7, callwait = -1, thirdcall = -1
disabled echo cancellation on channel 1
Set option TDD MODE, value: OFF(0) on Zap/1-1
Updated conferencing on 1, with 0 conference users
-- Hungup 'Zap/1-1'
snip
; --- Config B --- ;
*Code:*
exten = 111,1,Answer
exten =
Re: [asterisk-users] CallWithUs Service?
In VOIP, your quality of your voice is as good as your network. if you want clear call quality, QOS is a must. Well, when the call leaves your network and enters internet, QOS is not enforced. As a general rule choose the closest to your network. for me its Teliax, i get to their proxy after 7 hops. On 9/14/07, Anthony Messina [EMAIL PROTECTED] wrote: On Thursday 13 September 2007 02:32:52 pm John Meksavan wrote: I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? it was your message that prompted me to take a look at callwithus.com. i currently use diamondcard.us (via iax2) and have had only 2 issues in 9 months where some calls to verizon cell phones would get a congestion signal if they didn't answer instead of going to their voicemail. i called diamondcard and they fixed the trunk issue in a matter of an hour. call quality is decent. after signing up with callwithus.com, i find the call quality to be the same as diamondcard, though diamondcard bills in 30sec increments at 1.7cents/min in the us and callwithus bills in 1 minute increments at 1.4 cents/min in the us. callwithus also has this thing where if you add a *31 to the number, it will choose their cheapest route. i'd say they are worth trying, so is diamondcard.us. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
Jared Smith wrote: On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: . matches any number of the preceding character, change it to _X.*X. That still won't help. Once the Asterisk pattern matching parser sees a period in the pattern, it ignores anything after it. (I'm not exactly happy about that, but that's the way it is.) In short, Asterisk doesn't currently have a good way of handling this situation. Hopefully somebody infinitely smarter than I am will take pity on our plight and give us a some more advanced pattern-matching tools. (Hint, hint) Like PCRE maybe hmm. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to route outgoing calls on IP-level
well, the situation is: we have two-state VoIP-routing customers (h323,sip) --- asterisk -- our home made h323 proxy the final billing is done at h323 proxy, and it distingushes customers by their IP addresses. so, if I want to bill two group of SIP customers separately, I need to route calls to h323 proxy with different outgoing addresses. It's easy to buy extra IP-address, but I've no idea how to teach route command to do things like hey, it Bill Clinton calling, I see SIP headers, we ought to use X.X.X.X as outgoing IP address, not Y.Y.Y.Y On 9/15/07, Rafael Canchola [EMAIL PROTECTED] wrote: Check the route command on your Linux system. The gateway route should be the ethX and network whatever you want. At 01:41 p.m. 14/09/2007, Drew Gibson wrote: Kate Kretz wrote: Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to use one IP address, and others to go through another address. is there a way to achive that using asterisk ? Cheers, Kate This is the job of your network, not Asterisk. Policy-based routing is not much fun (unless you think the Cisco CLI is really cool) but it can be done. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- * Rafael*Canchola *Product Development Engineer*, FonetGlobal Inc. [EMAIL PROTECTED] http://www.fonetglobal.com *Ph. *+ 52 800 022 10 21 ext. 214 + 52 442 167 08 00 *VoIP* 523663899 *d00d! *cyberalph ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to route outgoing calls on IP-level
What are the factors in deciding which interface the traffic needs to go out of? Is it based on IP address, is it based on the terminating carrier? --Joe On 9/14/07, Kate Kretz [EMAIL PROTECTED] wrote: Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to use one IP address, and others to go through another address. is there a way to achive that using asterisk ? Cheers, Kate ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --Joe Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit. --Fred Franzia (The famous wine guy) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
On Friday 14 September 2007 04:12:48 pm Atis wrote:
exten = _XXX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
excellent sir! thank you! actually, since i'm using this for testing
ISN/ITAD, which currently only has ITAD domains with 3 digits i used:
exten = _XXX*XXX,1,Macro(isn,${EXTEN})
exten = _*XXX,1,Macro(isn,${EXTEN})
exten = _X*XXX,1,Macro(isn,${EXTEN})
(i use the macro to set callerid, etc)
would _XXX*XXX be slower to match than _XXX*. since the . ignores everything
after it as posted by another user?
again, thanks. -a
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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