Re: [asterisk-users] partial ChanSpy
i'm taking a look to app_chanspy.c what do you intend for trunk? the last cvs? can i download the last cvs and then write a patch for the actual 1.2 branch stable? thanks On 8/3/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 8/3/07, nik600 [EMAIL PROTECTED] wrote: is it possible to spy (not record, spy) partially on a channel? for exaple, i'd like to listen only the input or the output voice. trunk has added an 'o' option to ChanSpy: o - Only listen to audio coming from this channel.\n You might be able to achieve what you want by alternately spying on either side of the bridged call using 'o' both times. I'm not sure if this would be portable back into 1.4 though or if you'll have to wait for 1.6 to be released. -- j. ___ --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 -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --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] 답장: Asterisk ref book
Hi. If you are beginner, that is good book. If you get more depth skill, you need find out another book. Han clive.chan(atn) [EMAIL PROTECTED] 쓰기: Hi all, Can some one tell me about the book name call “ Asterisk Configuration Guide” comment? Or Review about this book. Does it useful for entry to medium level skills of Asterisk system? Thank you. ___ --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 - 지금, 고마운 사람에게 따뜻한 이메일 한 통 보내세요! 모두가 행복한 세상이 됩니다. 고마운 메일 보내러가기 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk always rining phone
Dear all I have setup of asterisk 1.2.14 with 100 SIP phone and it is working fine but thing is that when i call to somebody on local extention my asterisk not give me notification like party phone is busy or busy tone alway it give me rining single how can i justify other party is not pickup the phone or he/she talking with somebody on phone caz my phone rining on both stages is there any special configuration for it ?? - Got a little couch potato? Check out fun summer activities for kids.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.2.14 with GUI
dear all is there any GUI application with support asterisk 1.2 version i am useing 1.2 and i have fine more about GUI base configuration but i didnt got any GUI package for asterisk 1.2 - Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. ___ --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] IAX2 - DualServer Problem
Hi, I have two asterisk servers and I want to make these servers call each other as they were internal. I have succeeded in one way. Server B can call Server A without problem, but Server A cannot call Server B. Here's the iax configuration of servers Server A: == [ipek] auth=rsa context=from-internal host=XXX.XXX.XXX.XXX inkeys=ipek outkey=odtu peercontext=from-internal type=friend username=odtu Server B: == [odtu] auth=rsa context=from-internal host=YYY.YYY.YYY.YYY inkeys=odtu outkey=ipek peercontext=from-internal type=friend username=ipek When I try to call from A to B, I hear a message saying All circuits are busy now, Please try again later, and also here's the iax2 debug of Server A. I also tried to get debug output of Server B, when A is calling, but failed. Server B doesn't create any debug output, when A calls B. Another issues is, both servers are behind nat, but the UDP 4569 port is forwarded in each servers. I can connect to the servers with netcat from outside. Ports are forwarded perfectly. I am trying to overcome the issue for days, I am in a desperate situation. Please help me. Saki Server A (IAX2 Debug) Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 3 DCall: 0 [XXX.XXX.XXX.XXX:4569] VERSION : 2 CALLED NUMBER : 214 CODEC_PREFS : (ulaw|alaw) CALLING NUMBER : 105 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: 105 LANGUAGE: en CALLED CONTEXT : from-internal USERNAME: odtu FORMAT : 4 CAPABILITY : 63500 ADSICPE : 2 DATE TIME : 2007-08-03 01:58:56 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 3 DCall: 0 [10.10.10.73:4569] VERSION : 2 CALLED NUMBER : 214 CODEC_PREFS : (ulaw|alaw) CALLING NUMBER : 105 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: 105 LANGUAGE: en CALLED CONTEXT : from-internal USERNAME: odtu FORMAT : 4 CAPABILITY : 63500 ADSICPE : 2 DATE TIME : 2007-08-03 01:58:56 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 1ms SCall: 4 DCall: 3 [10.10.10.73:4569] CAUSE : No authority found CAUSE CODE : 50 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 1ms SCall: 4 DCall: 3 [88.248.2.48:4569] CAUSE : No authority found CAUSE CODE : 50 Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 3 DCall: 4 [88.248.2.48:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 3 DCall: 4 [10.10.10.73:4569] ___ --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] Connecting two Asterisk servers with a frame relay connection
Hello all, I have to connect two Asterisk servers with a frame relay connection but i do not know what is the hardware to use and how to connect them. Have anyone an idea about that. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware advice for 100 extensions, routing via ISDN
I would be grateful for some comments on our proposed machine specs for a new Asterisk installation at a client with an initial 70 extensions. The system should be able to handle 100 extensions. The system will have the following main features: - PSTN connection via ISDN 30, dealing with all incoming calls. Outgoing will be through ISDN initially - 70-100 Snom 300 handsets - 1-2 Snom 370 reception phones - voicemail voicemail to email - occasional conferencing requirements This is a normal office environment (architects) and we do not anticipate exceptionally heavy call volumes; on the other hand some conversations will last a very long time. I've had a look at http://voip-info.org/wiki/view/Asterisk+dimensioning We are presently intending to put in 2 number CHASSIS/CASE: 2U 2HotSwap Bay 510W PSU MOTHERBOARD: Tyan s5197G2NR CPU(s): Core2Due E6600 (2*2.4GHz) MEMORY: 4GB 667 ECC (2*2048) HDD: 2*150GB Raptor HDD CD/DVD: DVD/RW OTHER: Sangoma A101PCI Card We will be running on 64 bit Debian. The second machine is to be used in place of the first in case of failure. Advice gratefully received. Rory -- Rory Campbell-Lange Campbell-Lange Workshop Ltd. [EMAIL PROTECTED] www.campbell-lange.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 1.2.14 with GUI
On Sat, Aug 04, 2007 at 01:42:35AM -0700, satish patel wrote: dear all is there any GUI application with support asterisk 1.2 version i am useing 1.2 and i have fine more about GUI base configuration but i didnt got any GUI package for asterisk 1.2 freepbx? destar? voiceone? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by 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] IAX Encryption
IAX is not encrypted. What you're seeing in wireshark is likely the authentication method you've chosen. (RSA or MD5) You can encrypt it with a VPN as long as you have a pipe fat enough to deal with the overhead a VPN puts on packets. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Wednesday, July 25, 2007 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Encryption On 23 Jul 2007, at 15:53, Matthew Brothers wrote: I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it looks like the encryption is being setup normally when trunking is enabled. I just can't tell if the voice frame within the trunked packet is encrypted. Any assistance would be appreciated. I thought that Encryption and Trunking are mutually exclusive in IAX. What does the iax debug in asterisk show? Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall and Private CID
Moises Silva wrote: I would not call that properly a fix. We need to know why is failing in newer spandsp versions in the first place. Can you make a diff and post it? Why are people so determined to break things. If you want to use unicall-0.0.3pre11, use it with spandsp-0.0.2. The latest versions of unicall (0.0.5) work with the latest spandsp (0.0.4), but I have done nothing about making either of them work with Asterisk. On 8/3/07, Carlos Chavez [EMAIL PROTECTED] wrote: On Fri, 2007-08-03 at 00:23 -0300, Luis Antonio Prata Barbosa wrote: Hi Carlos, I suggest you download spandsp-0.0.3pre22. (http://www.neuwald.biz/files/spandsp-0.0.3pre22.gz) I don´t know why , spandsp after that uses digits 1,2..8,9,A,B,C,D,E,F instead of 1,2,..,9,0,A,B,C,D,E. So, do you get F digits that are incompatible with mfcr2 . Its OK. I know why. :-) Its because people kept sending me bogus problem reports saying I should be getting signal 15 and I get 'E'. Well 'E' was signal 15, but that seemed to confuse people. I have made matching changes in more recent versions of spandsp and Unicall, to make signals 11 to 15 give 'B' to 'F', instead of 'A' to 'E'. It doesn't affect the behaviour of the software at all, as long as you use a matching set of spandsp and unicall versions. Thank you. I got an older set of files I had on another server (pre6) and now everything is working. The customer now gets CID and calls from Nextel. This is probably the way to fix Unicall on 1.4 since it uses a newer version of spandsp and has the exact same problem. Steve ___ --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] Connecting two Asterisk servers with a frame relay connection
You could use SIP if the servers are on routable IPs or the same subnet, if not you could use IAX but I think OpenVPN is your best choice for using SIP over different NATed networks. I do not think you need any hardware except for what is needed for the Frame Relay. QoS and traffic shaping would be a good idea if other traffic is going over your link. Thanks, Steve Totaro MOSBAH ABDELKADER wrote: Hello all, I have to connect two Asterisk servers with a frame relay connection but i do not know what is the hardware to use and how to connect them. Have anyone an idea about that. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 - DualServer Problem
I have seen this No Authority Found many times. Not sure what the fix was, I just kept playing with it until I got it working. I suggest getting rid of the inkeys, auth=rsa, and adding a secret. Make the username and passwords the same on both sides as well. If that works, then you know something is wrong with your more advance authentication mechanisms. Thanks, Steve Totaro Mustafa Sakalsiz wrote: Hi, I have two asterisk servers and I want to make these servers call each other as they were internal. I have succeeded in one way. Server B can call Server A without problem, but Server A cannot call Server B. Here's the iax configuration of servers Server A: == [ipek] auth=rsa context=from-internal host=XXX.XXX.XXX.XXX inkeys=ipek outkey=odtu peercontext=from-internal type=friend username=odtu Server B: == [odtu] auth=rsa context=from-internal host=YYY.YYY.YYY.YYY inkeys=odtu outkey=ipek peercontext=from-internal type=friend username=ipek When I try to call from A to B, I hear a message saying All circuits are busy now, Please try again later, and also here's the iax2 debug of Server A. I also tried to get debug output of Server B, when A is calling, but failed. Server B doesn't create any debug output, when A calls B. Another issues is, both servers are behind nat, but the UDP 4569 port is forwarded in each servers. I can connect to the servers with netcat from outside. Ports are forwarded perfectly. I am trying to overcome the issue for days, I am in a desperate situation. Please help me. Saki Server A (IAX2 Debug) Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 3 DCall: 0 [XXX.XXX.XXX.XXX:4569] VERSION : 2 CALLED NUMBER : 214 CODEC_PREFS : (ulaw|alaw) CALLING NUMBER : 105 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: 105 LANGUAGE: en CALLED CONTEXT : from-internal USERNAME: odtu FORMAT : 4 CAPABILITY : 63500 ADSICPE : 2 DATE TIME : 2007-08-03 01:58:56 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 3 DCall: 0 [10.10.10.73:4569] VERSION : 2 CALLED NUMBER : 214 CODEC_PREFS : (ulaw|alaw) CALLING NUMBER : 105 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: 105 LANGUAGE: en CALLED CONTEXT : from-internal USERNAME: odtu FORMAT : 4 CAPABILITY : 63500 ADSICPE : 2 DATE TIME : 2007-08-03 01:58:56 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 1ms SCall: 4 DCall: 3 [10.10.10.73:4569] CAUSE : No authority found CAUSE CODE : 50 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 1ms SCall: 4 DCall: 3 [88.248.2.48:4569] CAUSE : No authority found CAUSE CODE : 50 Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 3 DCall: 4 [88.248.2.48:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 3 DCall: 4 [10.10.10.73:4569] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk always rining phone
Sounds like you have call waiting on the phones. You can disable this on the Asterisk side. To verify, make a call on your phone and then dial yourself from another phone. Depending on the phone, you will have some sort of indication that a second call is coming in. Thanks, Steve Totaro satish patel wrote: Dear all I have setup of asterisk 1.2.14 with 100 SIP phone and it is working fine but thing is that when i call to somebody on local extention my asterisk not give me notification like party phone is busy or busy tone alway it give me rining single how can i justify other party is not pickup the phone or he/she talking with somebody on phone caz my phone rining on both stages is there any special configuration for it ?? Got a little couch potato? Check out fun summer activities for kids. http://us.rd.yahoo.com/evt=48248/*http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Turn off musiconhold
How do you disable musiconhold for a single sip peer? I see that there is a musicclass setting, but what do you set it to so that you disable musiconhold? ___ --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] partial ChanSpy
ok, i've taken a look at the actual app_chanspy.c and the newest i've tried to comment ast_set_flag(csth.spy, CHANSPY_MIXAUDIO); and recompile asterisk, now i can hear only the input stream of the channel spyed. that's fine! thansk On 8/4/07, nik600 [EMAIL PROTECTED] wrote: i'm taking a look to app_chanspy.c what do you intend for trunk? the last cvs? can i download the last cvs and then write a patch for the actual 1.2 branch stable? thanks On 8/3/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 8/3/07, nik600 [EMAIL PROTECTED] wrote: is it possible to spy (not record, spy) partially on a channel? for exaple, i'd like to listen only the input or the output voice. trunk has added an 'o' option to ChanSpy: o - Only listen to audio coming from this channel.\n You might be able to achieve what you want by alternately spying on either side of the bridged call using 'o' both times. I'm not sure if this would be portable back into 1.4 though or if you'll have to wait for 1.6 to be released. -- j. ___ --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 -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --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] Difference between WaitExten and TIMEOUT (response)
The difference is in the scope of the command. Think of it this way: WaitExten gives the user more time to enter digits before the dialplan moves on to the next instruction in the dial plan. Timeout is the max number of seconds to wait at any point in the current context before deciding the user either got confused, doesn't know what their doing, or fell asleep. Clear as mud? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Friday, August 03, 2007 8:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Difference between WaitExten and TIMEOUT (response) Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? Regards Bilal Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time Limit on Call or Conference Room? NEW ASTERISK PROVERB
On Fri, 3 Aug 2007, JR Richardson wrote: Can anyone point me int he right direction? At the risk of coming off in a gratuitiously self-aggrandising manner quoting myself: http://lists.digium.com/pipermail/asterisk-users/2007-May/188438.html -- Alex Balashov Thank you, Alex. As I've said many times, this community has the smartest people in the world. It is with great humbleness, I offer this to all. New Asterisk Proverb: Asterisk is like an onion with many, many layers. With 160+ applications and seemingly endless options a person just can't know it all. Often one needs a new way to manipulate calls, searches and discovers the solution, realizing it was in the code all along. Inevitability lures one to investigate, deeper understanding is accomplished, maybe even profound but never complete. As the layers of the Onion are peeled back, wear proudly the malodorous smell of knowledge that is Asterisk. JR Richardson Engineering for the Masses I have the Asterisk stink on me! ___ --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] Digium FTP server will be replaced with HTTP server
So where will the files be then? What will the new link be? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Thursday, July 26, 2007 1:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Digium FTP server will be replaced with HTTP server Some time in the next two weeks, Digium will be shutting down our FTP server, located at ftp.digium.com, and begin using only the existing HTTP server on the same system instead. We have decided to only offer our public downloads over the HTTP protocol, not the FTP protocol, primarily for reasons related to our marketing department :-) The site will still be called ftp.digium.com, but will no longer respond to requests made via the FTP protocol; only the HTTP protocol will be supported. There should be no other user-visible changes when this change is made to the server. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FTP server will be replaced with HTTP server
It says clearly in the email from Digium. The link is the same. ftp.digium.com but no FTP. Sounds silly to me. So it is http://ftp.digium.com. Thanks, Steve Totaro Michael Munger wrote: So where will the files be then? What will the new link be? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Thursday, July 26, 2007 1:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Digium FTP server will be replaced with HTTP server Some time in the next two weeks, Digium will be shutting down our FTP server, located at ftp.digium.com, and begin using only the existing HTTP server on the same system instead. We have decided to only offer our public downloads over the HTTP protocol, not the FTP protocol, primarily for reasons related to our marketing department :-) The site will still be called ftp.digium.com, but will no longer respond to requests made via the FTP protocol; only the HTTP protocol will be supported. There should be no other user-visible changes when this change is made to the server. ___ --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] Time Limit on Call or Conference Room? NEW ASTERISK PROVERB
JR Richardson wrote: Thank you, Alex. As I've said many times, this community has the smartest people in the world. It is with great humbleness, I offer this to all. New Asterisk Proverb: Asterisk is like an onion with many, many layers. With 160+ applications and seemingly endless options a person just can't know it all. Often one needs a new way to manipulate calls, searches and discovers the solution, realizing it was in the code all along. Inevitability lures one to investigate, deeper understanding is accomplished, maybe even profound but never complete. As the layers of the Onion are peeled back, wear proudly the malodorous smell of knowledge that is Asterisk. JR Richardson Engineering for the Masses I have the Asterisk stink on me! Take a shower for the love of humanity! ___ --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] Digium FTP server will be replaced with HTTP server
Steve Totaro wrote: It says clearly in the email from Digium. The link is the same. ftp.digium.com but no FTP. Sounds silly to me. So it is http://ftp.digium.com. We added http://downloads.digium.com/. We will be using that URL for all of our links from now on. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] Unicall and Private CID
On 8/4/07, Steve Underwood [EMAIL PROTECTED] wrote: Why are people so determined to break things. If you want to use unicall-0.0.3pre11, use it with spandsp-0.0.2. Not really determined to break things, but to understand failures, even when those failures are because of version missmatching :) The latest versions of unicall (0.0.5) work with the latest spandsp (0.0.4), but I have done nothing about making either of them work with Asterisk. Minor changes were needed to chan_unicall. Anyone interested in using it can find it here: http://www.moythreads.com/astunicall/ Its OK. I know why. :-) Its because people kept sending me bogus problem reports saying I should be getting signal 15 and I get 'E'. Well 'E' was signal 15, but that seemed to confuse people. I have made matching changes in more recent versions of spandsp and Unicall, to make signals 11 to 15 give 'B' to 'F', instead of 'A' to 'E'. It doesn't affect the behaviour of the software at all, as long as you use a matching set of spandsp and unicall versions. Understood. Thanks. Moy ___ --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] Outcall 1.40 released
Hi OutCALL 1.40 is released. It is available in two flavours: - Without extension authentication - With extension authentication Changelog: OutCALL 1.40 (2007-06-29): - Multi-language support (French-Canada is included in the setup, while the English PO file is distributed with OutCALL setup which can be translated and added into OutCALL in run-time) Please use http://www.poedit.net/ for translation - Support for Skinny protocol - It is possible to define prefix for outgoing calls (Settings-General) - It is possible to define one or more prefixes which will be deleted from the incoming CallerID (Settings-General) - In Settings dialog, after you Apply changes, OutCALL automatically reconnects using new Server details (if those are changed) - It is possible to Import Contacts from CSV file which is generated using Outlook Export Wizard ( File-Export-Comma Separated Values (Windows) ) BUG fixes: - Critical BUG when Loading Outlook Contacts (some contacts would not be loaded if Contact's info contains some escaping characters) - Settings and other dialogs cannot be opened twice - Settings and other dialogs can now be accessed from the taskbar - Added all DLL dependencies into the setup Available at: http://outcall.sourceforge.net/ Regards, Senad Jordanovic www.bicomsystems.com [EMAIL PROTECTED] +1 (212) 400 7921 +44 (20) 7043 3488 Regards, Senad Jordanovic www.bicomsystems.com [EMAIL PROTECTED] +1 (212) 400 7921 +44 (20) 7043 3488 ___ --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] Configuring Sangoma A101D with Asterisk 1.2.18 zaptel-1.2.17.1
Deepak Naidu wrote: It would help to know exactly what Dell Poweredge you were considering. They do vary. I have Dell Power Edge 850 Also how do I enable DTMF hardware detection. There are no drivers which support it. I have the lastest Beta drivers installed, they seem to show yes in the logs, but the hardware DTMF didnt work, so I wrote a mail, to the developer of the drivers he said they are still working in the lab probably have one within a week. You should try relaxdtmf=yes in zapata.conf first. -Stephen- ___ --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] polycom custom ring tones (slightly OT)
Doug wrote: At 21:59 7/29/2007, Paul Hales wrote: I even got a Polycom here saying I'll be back which was funny for about an hour, then not funny at all. PaulH Kewwl! How do you get the .wav files into the Polycom? If it's not obvious, I'd be interested in this information too. Most people seem to think you can't change the ringtones on the Polycom sets. -Stephen- ___ --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] Royalty for On Hold Music ?
Steve Kennedy wrote: On Tue, Jul 31, 2007 at 05:22:20PM -0400, Jon Pounder wrote: Quoting John Millican [EMAIL PROTECTED]: there are plenty of radio stations with internet feeds of their audio, piping that in would not change any coverage area since anyone with internet could listen anywhere already, you're only providing that to the listener through a phone handset instead of a computer speaker, which amounts to just another audio device controlled by an internet connected computer. No it's not, you're rebroadcasting and that would incur a difference license (if legal at all). What if the radio is on in the background when I make a call ? is that rebroadcasting ? kind of gets blurry on the definitions there. That's not as you're listening to it and not trying to rebroadcast. Well, this is approaching the absurd. Do you know how many Meridian systems have radios plugged into them for on-hold background sound? Nobody pays royalties on those. There are the rules and then there are the practical realities. -Stephen- ___ --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] Teliax Quality of Service
Douglas Garstang wrote: I confused by this. Don't ITSP's have redundancy? Don't they have multiple edge systems for accepting incoming calls? Don't their multiple edge systems have multiple interfaces, connected to multiple subnets, via multiple switches? And, don't they have multiple upstream providers? About the only thing that could go wrong that affects all service like this would be a badly pushed out software update, affecting all systems? Don't be confused. The answer to most of your questions is no. Barriers to entry are too small for ITSPs, and there are lots of basement operations masquerading as big carriers. -Stephen- ___ --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] Pre-recorded first and last names audio database
Hi! My application needs to look up (by spelling) the first and last names of a person and then insert the corresponding pre-recorded audio file to personalize the message. E.g. Hi, John Brown. Your book is due back at the library. So I need John and Brown in audio files along with LOTS of other names - Do such databases of sound files already exist or do I have to record my own? I'm not sure how many first and last names I'd have to record but it seems like thousands for both genders first names and then thousands more for last names to cover a significant proportion of the people in the USA - Any and all help appreciated! Thanks, John ___ --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] quintum AFT200 connection to Asterisk
Hi, I have an asterisk and a quintum AFT200 with two FXO ports, and want to use it as a gateway to handle outgoing and incoming calls. I have found this thread, http://lists.digium.com/pipermail/asterisk-users/2005-February/084015.html But I think I need a little more help, could anyone knows where I can find the basic configuration for this quintum to get it connected to Asterisk using SIP, no NAT needed. thanks. -- Guillermo Garron Linux IS user friendly... It's just selective about who its friends are. (Using FC6, CentOS4.4 and Ubuntu 6.06) http://feeds.feedburner.com/go2linux http://www.go2linux.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Update zaptel on business edition.
This seems like something I should know... but I don't. How do you update zaptel / libpri on a Business Edition box running rPath? Tried running conary, but got 'Insufficient permission to access server conary.digium.com. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk servers with a frame relay connection
Hello, Have i to buy an asterisk card like TDM400P to connect the two asterisk servers with frame relay. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Royalty for On Hold Music ?
Stephen Bosch wrote: Well, this is approaching the absurd. Do you know how many Meridian systems have radios plugged into them for on-hold background sound? Nobody pays royalties on those. IF they are discovered by ASCAP and receive a letter demanding payment they will. Not absurd at all. Simply because many do it in ignorance doesn't make it legal ASCAP goes on campaigns on a regular basis. Home residential users are probably safe though not legal. Business users have a greater visibility though There are all sorts of royalty free music sources available. No excuse not to use it. Or simply pay the yearly fee to ASCAP ( in the US ) There are the rules and then there are the practical realities. -Stephen- ___ --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 -- Dog is my co-pilot ___ --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] * and SIP ocupped
I'm using asterisk 1.2 with debian and I have configured a SIP account in * and using it trought X-Lite. My problem is that I can't do phone call(voip or PSTN) to the SIP account(a friend account) because it always answer ocupped. Using this SIP account on ATA it doesnt happen and i can do and receiva calls. The other problem is that i can't do phone call neither PSTN or VoIP, trying to do phone calls i always listen a this number doesnt exist from the VoIP provider. can someone take a look in this case to me ? here are the config files and debug: http://pastebin.ca/644967 jp. ___ --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] Connecting two Asterisk servers with a framerelay connection
What modules do you want on it? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MOSBAH ABDELKADER Sent: Saturday, August 04, 2007 3:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Connecting two Asterisk servers with a framerelay connection Hello, Have i to buy an asterisk card like TDM400P to connect the two asterisk servers with frame relay. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pre-recorded first and last names audio database
On 8/4/07, John Vogel [EMAIL PROTECTED] wrote: Hi! My application needs to look up (by spelling) the first and last names of a person and then insert the corresponding pre-recorded audio file to personalize the message. E.g. Hi, John Brown. Your book is due back at the library. So I need John and Brown in audio files along with LOTS of other names - Do such databases of sound files already exist or do I have to record my own? I'm not sure how many first and last names I'd have to record but it seems like thousands for both genders first names and then thousands more for last names to cover a significant proportion of the people in the USA - I haven't seen something like this, but if you figure it out, I'd like to know. There's a piece of software called HouseCalls that reminds people of appointments. The proprietary software prompts the person setting up the automated reminders to record each name individually. In the beginning, it's a bear, but over time, it gets better. I guess something in Asterisk would have to do the same, right? I mean, a general list of John Jon Jonh in some person's voice, and the rest of the prompt in another, wouldn't be much better than having Festival say the name, would it? ___ --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] Difference between WaitExten and TIMEOUT (response)
Dear Michael; I understood it in that way (please advise me if I am correct): WaitExten is for the time to complete entering the digits, while timeout is specified wether user responded by dialing any thing or not. Please advise. regards The difference is in the scope of the command. Think of it this way: WaitExten gives the user more time to enter digits before the dialplan moves on to the next instruction in the dial plan. Timeout is the max number of seconds to wait at any point in the current context before deciding the user either got confused, doesn't know what their doing, or fell asleep. Clear as mud? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Friday, August 03, 2007 8:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Difference between WaitExten and TIMEOUT (response) Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? Regards Bilal Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Royalty for On Hold Music ?
Exactly, with the amount of royalty free music out there why bother. Just go searching for some you like, download it and while you are at it tip the author/performer a couple of bucks into their myspace tip jar or similar. For $10 why take the risk with ascap. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Saturday, 4 August 2007 2:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Royalty for On Hold Music ? Stephen Bosch wrote: Well, this is approaching the absurd. Do you know how many Meridian systems have radios plugged into them for on-hold background sound? Nobody pays royalties on those. IF they are discovered by ASCAP and receive a letter demanding payment they will. Not absurd at all. Simply because many do it in ignorance doesn't make it legal ASCAP goes on campaigns on a regular basis. Home residential users are probably safe though not legal. Business users have a greater visibility though There are all sorts of royalty free music sources available. No excuse not to use it. Or simply pay the yearly fee to ASCAP ( in the US ) There are the rules and then there are the practical realities. -Stephen- ___ --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 -- Dog is my co-pilot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between WaitExten and TIMEOUT (response)
On 05:27, Fri 03 Aug 07, bilal ghayyad wrote: Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? WaitExten is waiting for you to type an extension. TIMEOUT is to set the default timeout for promtps in IVR and stuff but is not actually waiting for you to provide an extension -- 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? ___ --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] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box
Steve, On 8/3/07, Steve Totaro [EMAIL PROTECTED] wrote: I just tried to call in after creating an account. After the call connects, enter the show id: 22622# and your_PIN# I dial in and am asked for the podcast id, I enter 22622# and am told that my passcode is not correct. I also tried just entering my passcode but got the same error message. What am I doing wrong? Nothing. What time did you do this? Are you sure the conference was on? If it isn't live, you can't get in. ___ --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] Teliax Quality of Service
Stephen Bosch wrote: Douglas Garstang wrote: I confused by this. Don't ITSP's have redundancy? Don't they have multiple edge systems for accepting incoming calls? Don't their multiple edge systems have multiple interfaces, connected to multiple subnets, via multiple switches? And, don't they have multiple upstream providers? About the only thing that could go wrong that affects all service like this would be a badly pushed out software update, affecting all systems? Don't be confused. The answer to most of your questions is no. Barriers to entry are too small for ITSPs, and there are lots of basement operations masquerading as big carriers. -Stephen- There are also lots of big carriers masquerading as big carriers. ;) If the ONLY people who could get into the business were the ones who could, before offering any services to customers, afford to build out multiple edge systems for accepting incoming calls, each with multiple interfaces connected to multiple subnets via multiple switches using multiple upstream providers, you would have ONE single choice for an ITSP. And ATT doesn't have that amount of redundancy in their network. Working in the carrier networking business, I can assure you that we've NEVER run across a SINGLE carrier network (not from the largest to the smallest) that has redundancy in ALL aspects (or even MOST aspects) of its network. This is why there are uptime policies that allow a percentage of outages to occur. Triple 9 uptime (Exceedingly rare, but a purported goal -- 99.999%) still allows 15 full hours of downtime a year. And that rarely includes the occasional lost packet or latency. Face it. If you want service that never goes down, you're either able to pay the hundreds of millions to provide your own networks and build out your own redundancy, or you're stuck in the same boat with the rest of us -- be it that you choose a gigantic carrier or a mom 'n' pop ITSP. N. h ___ --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] Royalty for On Hold Music ?
John Novack wrote: Stephen Bosch wrote: Well, this is approaching the absurd. Do you know how many Meridian systems have radios plugged into them for on-hold background sound? Nobody pays royalties on those. IF they are discovered by ASCAP and receive a letter demanding payment they will. Not absurd at all. Simply because many do it in ignorance doesn't make it legal ASCAP goes on campaigns on a regular basis. Home residential users are probably safe though not legal. Business users have a greater visibility though There are all sorts of royalty free music sources available. No excuse not to use it. Or simply pay the yearly fee to ASCAP ( in the US ) The fact that ASCAP goes on campaigns doesn't make it any less absurd (or, for that matter, any more likely that the average business is going to be taken to task); the reality is that thousands upon thousands of interconnects install PBX systems with radio ports on them that are plugged into cheap transistor radios bought at Wal-Mart and similar places, and nobody -- not the client, nor the interconnect -- has any clue about any royalty obligations that entails. People do it, think nothing of it (not least because the PBX vendors promote it as a feature!) and I think neither ASCAP nor any other royalty agency has the necessary resources to make even a dent in this kind of use. It's one thing if you're Dell or Microsoft and you are using music for your call centre, and another if you're the neighbourhood dental practice. I'd be interested in getting in touch with any small businesses which have been given a cease and desist letter or demand for payment because they piped radio into their phone systems. -Stephen- ___ --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] Teliax Quality of Service
SIP wrote: There are also lots of big carriers masquerading as big carriers. ;) *lol* If the ONLY people who could get into the business were the ones who could, before offering any services to customers, afford to build out multiple edge systems for accepting incoming calls, each with multiple interfaces connected to multiple subnets via multiple switches using multiple upstream providers, you would have ONE single choice for an ITSP. And ATT doesn't have that amount of redundancy in their network. Working in the carrier networking business, I can assure you that we've NEVER run across a SINGLE carrier network (not from the largest to the smallest) that has redundancy in ALL aspects (or even MOST aspects) of its network. This is why there are uptime policies that allow a percentage of outages to occur. Triple 9 uptime (Exceedingly rare, but a purported goal -- 99.999%) still allows 15 full hours of downtime a year. And that rarely includes the occasional lost packet or latency. In other words, you can blame the marketing departments in various big carriers for creating these unrealistic expectations in the marketplace :) -Stephen- ___ --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] zttool says tdm800 is OK, but it won't recieve calls.
I have a TDM800 that is installed and working. (TDM800 + 2 X QUAD FXO). Zttool says it is configured, ok, and there are no issues. Ztcfg -vvv shows that all the channels are configured. Zap show channels in the CLI show all 8 channels configured as they are supposed to be. When I plug in a pots line from the telco and make a call to that line, asterisk does not respond. (No Starting Simple Switch). If I plug a regular telephone into that line, and call the phone number, it rings, people can answer it, and it works. What am I missing here? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] text2wave Voices Improvements?
I currently have an AGI that calls the Festival text2wave app to write a wav file that my dialplan plays into a call with the Background() command. But the voice sounds terrible: like SAM, the 1980s 6502 voice synthesizer. I tried to slow it down by calling (text2wav -eval (Parameter.set 'Duration_Stretch 1.4) -scale 2.0 [...]), but it still sounds like it's talking while sucking down a strawful of spaghetti. How do I install a different voice, to speak basically simple emails? I'm (APT) installing on Debian 3.1/Sarge, Asterisk 1.4.x . Also, is there a way to call Background or some other Asterisk command to take the WAV data from a pipe to a running text2wav process, rather than writing a file with text2wave and then reading it (and then deleting it) in the dialplan/AGI? -- (C) Matthew Rubenstein ___ --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] IAX Encryption
Iax channel can be encrypted. Not just the authentication, even rtp data, see: http://www.voip-info.org/wiki/view/IAX+encryption On 8/4/07, Michael Munger [EMAIL PROTECTED] wrote: IAX is not encrypted. What you're seeing in wireshark is likely the authentication method you've chosen. (RSA or MD5) You can encrypt it with a VPN as long as you have a pipe fat enough to deal with the overhead a VPN puts on packets. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Wednesday, July 25, 2007 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Encryption On 23 Jul 2007, at 15:53, Matthew Brothers wrote: I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it looks like the encryption is being setup normally when trunking is enabled. I just can't tell if the voice frame within the trunked packet is encrypted. Any assistance would be appreciated. I thought that Encryption and Trunking are mutually exclusive in IAX. What does the iax debug in asterisk show? Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
Stephen Bosch wrote: Douglas Garstang wrote: I confused by this. Don't ITSP's have redundancy? Don't they have multiple edge systems for accepting incoming calls? Don't their multiple edge systems have multiple interfaces, connected to multiple subnets, via multiple switches? And, don't they have multiple upstream providers? About the only thing that could go wrong that affects all service like this would be a badly pushed out software update, affecting all systems? Don't be confused. The answer to most of your questions is no. I don't think he's really confused. Doug has a penchant for the provocative, historically. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Teliax Quality of Service
From: SIP Sent: Saturday, August 04, 2007 2:57 PM Stephen Bosch wrote: Douglas Garstang wrote: I confused by this. Don't ITSP's have redundancy? Don't they have multiple edge systems for accepting incoming calls? Don't their multiple edge systems have multiple interfaces, connected to multiple subnets, via multiple switches? And, don't they have multiple upstream providers? About the only thing that could go wrong that affects all service like this would be a badly pushed out software update, affecting all systems? Don't be confused. The answer to most of your questions is no. Barriers to entry are too small for ITSPs, and there are lots of basement operations masquerading as big carriers. -Stephen- There are also lots of big carriers masquerading as big carriers. ;) If the ONLY people who could get into the business were the ones who could, before offering any services to customers, afford to build out multiple edge systems for accepting incoming calls, each with multiple interfaces connected to multiple subnets via multiple switches using multiple upstream providers, you would have ONE single choice for an ITSP. And ATT doesn't have that amount of redundancy in their network. Working in the carrier networking business, I can assure you that we've NEVER run across a SINGLE carrier network (not from the largest to the smallest) that has redundancy in ALL aspects (or even MOST aspects) of its network. This is why there are uptime policies that allow a percentage of outages to occur. Triple 9 uptime (Exceedingly rare, but a purported goal -- 99.999%) still allows 15 full hours of downtime a year. And that rarely includes the occasional lost packet or latency. Your math is incorrect. FIVE nines (99.999) allows only 5.26 MINUTES of annual downtime. Triple nine (99.9%) allows for 8.76 hours of annual downtime. Keep in mind that most SLAs do not include planned maintenance in their guaranteed uptime. Face it. If you want service that never goes down, you're either able to pay the hundreds of millions to provide your own networks and build out your own redundancy, or you're stuck in the same boat with the rest of us -- be it that you choose a gigantic carrier or a mom 'n' pop ITSP. N. h Trevor Hammonds ___ --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] ! Command from -rx?
This may sound stupid.. so bear with me for a moment. Assuming the only access I have to a machine is through asterisk -rx can I use the ! command? asterisk -rx help includes the ! command, but I can't seem to get it to work ie: asterisk -rx ! ls Any help? ___ --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] Royalty for On Hold Music ?
Stephen Bosch wrote: John Novack wrote: Stephen Bosch wrote: Well, this is approaching the absurd. Do you know how many Meridian systems have radios plugged into them for on-hold background sound? Nobody pays royalties on those. IF they are discovered by ASCAP and receive a letter demanding payment they will. Not absurd at all. Simply because many do it in ignorance doesn't make it legal ASCAP goes on campaigns on a regular basis. Home residential users are probably safe though not legal. Business users have a greater visibility though There are all sorts of royalty free music sources available. No excuse not to use it. Or simply pay the yearly fee to ASCAP ( in the US ) The fact that ASCAP goes on campaigns doesn't make it any less absurd (or, for that matter, any more likely that the average business is going to be taken to task); the reality is that thousands upon thousands of interconnects install PBX systems with radio ports on them that are plugged into cheap transistor radios bought at Wal-Mart and similar places, and nobody -- not the client, nor the interconnect -- has any clue about any royalty obligations that entails. People do it, think nothing of it (not least because the PBX vendors promote it as a feature!) and I think neither ASCAP nor any other royalty agency has the necessary resources to make even a dent in this kind of use. Simply put - tell it to the judge. Drivers speed , change lanes, cut others off every day and MOSTLY get away with it. Doesn't make it legal, does it? Not any different than stealing software is it? It's one thing if you're Dell or Microsoft and you are using music for your call centre, and another if you're the neighbourhood dental practice. In the eyes of the law, it makes NO difference. Do it until you are caught, you say? I'd be interested in getting in touch with any small businesses which have been given a cease and desist letter or demand for payment because they piped radio into their phone systems. Not only their phone systems but their waiting rooms Next time you go into an office or store and you see the yellow ASCAP label on the door, you know they probably have gotten a letter. MANY interconnects now have discovered they can make extra by selling a message on hold system that not only hawks the wares of the firm but escapes the clutches of ASCAP. You remind me of a friend who enjoys a good argument with a tree stump. John Novack -- Dog is my co-pilot ___ --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 1.2.14 with GUI
satish patel wrote: dear all is there any GUI application with support asterisk 1.2 version i am useing 1.2 and i have fine more about GUI base configuration but i didnt got any GUI package for asterisk 1.2 If you're a windows user, you can also check out DialplanPro: http://www.datatrakpos.com/pos/datatalk We're still considering it beta, but we use it for our own pbx and those of the few clients we have using Asterisk and it works very well. It's also commercial (or will be someday...) Either way, its in beta and free to use if you like. --- Warm Regards, Lee ___ --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] ! Command from -rx?
On 8/4/07, Matt wrote: This may sound stupid.. so bear with me for a moment. Assuming the only access I have to a machine is through asterisk -rx can I use the ! command? asterisk -rx help includes the ! command, but I can't seem to get it to work ie: asterisk -rx ! ls Any help? asterisk -rx `! ls myout.txt` will save the output in myout.txt asterisk -rx `! ls` will give the command output sandwiched between * msgs. reduce *'s verbosity and you may have what you need. -- ___ --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] ! Command from -rx?
On Sat, Aug 04, 2007 at 09:16:22PM -0400, Matt wrote: This may sound stupid.. so bear with me for a moment. Assuming the only access I have to a machine is through asterisk -rx can I use the ! command? asterisk -rx help includes the ! command, but I can't seem to get it to work ie: asterisk -rx ! ls What do you need that for? '!' is pointless with asterisk -rx: with asterisk -r, '!' runs a local command in a subshel (or starts a new subshell) by the local cleint asterisk. It does nothing by the server. So you might as well just run: ls -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by 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] ! Command from -rx?
On Sat, Aug 04, 2007 at 10:05:34PM -0400, Baji Panchumarti wrote: On 8/4/07, Matt wrote: This may sound stupid.. so bear with me for a moment. Assuming the only access I have to a machine is through asterisk -rx can I use the ! command? asterisk -rx help includes the ! command, but I can't seem to get it to work ie: asterisk -rx ! ls Any help? asterisk -rx `! ls myout.txt` Huh? Those are backticks. They get translated by the shell (e.g.: bash) to the output of the command '! ls myout.txt' It seems that the '!' is interpeded here as a command, rather than as a part of history substitusion. See: $ echo `!ls` bash: !ls`: event not found $ echo `! ls` bash: echo: command not found As that specific command's output is redirected to a file, it will be expanded to: asterisk -rx '' Which is probably not what you wanted. will save the output in myout.txt asterisk -rx `! ls` Here the results will actually be the same, because '! ls' will not produce any output. But if it did, e.g: asterisk -rx `ls` you'd probably notice that asterisk normally doesn't like an arbitrary list of files as comands. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by 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] Teliax Quality of Service
Trevor G. Hammonds wrote: From: SIP Sent: Saturday, August 04, 2007 2:57 PM Stephen Bosch wrote: Douglas Garstang wrote: I confused by this. Don't ITSP's have redundancy? Don't they have multiple edge systems for accepting incoming calls? Don't their multiple edge systems have multiple interfaces, connected to multiple subnets, via multiple switches? And, don't they have multiple upstream providers? About the only thing that could go wrong that affects all service like this would be a badly pushed out software update, affecting all systems? Don't be confused. The answer to most of your questions is no. Barriers to entry are too small for ITSPs, and there are lots of basement operations masquerading as big carriers. -Stephen- There are also lots of big carriers masquerading as big carriers. ;) If the ONLY people who could get into the business were the ones who could, before offering any services to customers, afford to build out multiple edge systems for accepting incoming calls, each with multiple interfaces connected to multiple subnets via multiple switches using multiple upstream providers, you would have ONE single choice for an ITSP. And ATT doesn't have that amount of redundancy in their network. Working in the carrier networking business, I can assure you that we've NEVER run across a SINGLE carrier network (not from the largest to the smallest) that has redundancy in ALL aspects (or even MOST aspects) of its network. This is why there are uptime policies that allow a percentage of outages to occur. Triple 9 uptime (Exceedingly rare, but a purported goal -- 99.999%) still allows 15 full hours of downtime a year. And that rarely includes the occasional lost packet or latency. Your math is incorrect. FIVE nines (99.999) allows only 5.26 MINUTES of annual downtime. Triple nine (99.9%) allows for 8.76 hours of annual downtime. Keep in mind that most SLAs do not include planned maintenance in their guaranteed uptime. You are quite right, sir. I've no idea what I was doing with my math there. I can't even REPLICATE what I was doing with my math there. N. ___ --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] Sangoma PRI
Hi, I have a client who has a system with a Sangoma 1 port PRI card with echo canceling in it.For some reason, when the system comes up the PRI will stay up for about 4-5 hours, then drop. zap show status shows everything as ok, but we can't make or receive any calls until the system is rebooted. Just restarting asterisk does not fix the problem. I am going to call Verizon, however wanted to consult the list to see if anyone here had any ideas. At this point, I am putting my finger on a Verizon issue, as in our lab the system did not have any issues keeping the PRI active and taking calls. Any thoughts? ___ --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] Agents being bounced from queues after a call and sometimes randomly...
I am having a serious problem with agents being logged out of the queue after they finish a call. I am using static agents and agents.conf. I am running 2.1.17. Anyone having these problems or could think of anything that would cause them. Jordan Novak Telecommunications Engineer ___ --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] Time Limit on Call or Conference Room?
This might get you going: http://www.voip-info.org/wiki/view/Asterisk+cmd+AbsoluteTimeout -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, August 03, 2007 1:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Time Limit on Call or Conference Room? Hi All, I recently had an incident where a conf bridge was left open due to improper disconnection. I've read about the meetme options and marked callers closing the bridge when they exit. This is OK for meetme, but I'm really interested in a call timer that can be set on inbound and outbound calls within the dial plan, per call. I have another customer who wants to offer free calls, for 5-10 minutes with auto disconnect. Can anyone point me int he right direction? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.11.4/936 - Release Date: 8/4/2007 2:42 PM ___ --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
