Re: [Asterisk-Users] R2 support
Hi I know there is no support for R2, but I succcessfully compiled and ran the libr2. However, I am not able to initiate calls. The error I get is Couldn't call g3/71605538 -- Hungup 'Zap/32-1' == Everyone is busy at this time I understand that the idle signaling is not working right, any ideas on what I can do to fix this problem? Looking forward to your responses. Regards, Jorge On Fri, 2004-04-30 at 22:58, Steve Underwood wrote: jorge verastegui wrote: Hi i have successfully downloaded and compiled libr2 from source. But i dont seem to find how to properly configure it. When i run it (partcially unconfigured) the following error occurrs Signalling requested is R2 Signalling but line is in PRI Signalling signalling This error is easy to fix by changing ccs to cas, and removing crc4, in your zaptel.conf file. However libr2 does not work. It is a partly implemented solution which I abandoned. It only gets you about 10% of the way to a working R2 system :-( My current R2 software is completely different. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jorge Verastegui [EMAIL PROTECTED] RedCetus S.R.L. --NOTA DE REDCETUS S.R.L. : La informacin contenida en este E-mail y sus anexos, slo puede ser utilizada por el individuo o la compaa a la cual est dirigido. Si no es el receptor autorizado, cualquier retencin, difusin, distribucin o copia de este mensaje es prohibida y sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo y borrar el mismo
RE: [Asterisk-Users] Dell server for asterisk question!
And in the words of bkw all before him... NEXT!!! bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BGM Music
Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? -- respectfully, Joseph - (606) 477-2355 x140 --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Budgetone iLBC to IAX2 iLBC
We use 30ms and I think the payload is 98 if I recall correctly. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Stenton Sent: Thursday, May 13, 2004 9:48 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Budgetone iLBC to IAX2 iLBC I have updated my budgetone phone to the 1.04.63 firmware and am trying to use the iLBC codec for IAX2 passthrough to my PSTN breakout provider. The IAX2 gateway is set to use only iLBC. However, the SIP connection from the budgetone to the * server will use any other codec in preference to iLBC. If I force the SIP connection to be iLBC only, then it will make an iLBC connection. This makes me think that the iLBC codec in asterisk has a different sampling rate or payload size to default budgetone values. Anyone tell me what payload and sampling rate * uses for iLBC? Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to play dialtone on channel xx (Zaptel TE405P)
Hello, Im running * on a very basic configuration. I have a Wildcard TE405P with the first T1 connected to a PRI line and the remaining three to Adtran TA750 channel banks with FXS modules. I successfully configured everything to work with a couple of Swissvoice IP10S handsets (MGCP) and analog extensions connected to the channel banks. The problem Im having is that when I pick up any of the analog handsets I get no dial tone. Switching works, I can place and receive calls on both analog and MGCP handsets. The MGCP phones give me dial tone fine when I take them off the hook, only the analog ones dont do it. When I run the console with a lot of verbosity I see a message like: Apr 26 21:19:57 WARNING[1200825920]: chan_zap.c:4848 handle_init_event: Unable to play dialtone on channel 25 And then, right after I get: -- Starting simple switch on 'Zap/25-1' ...and Im able to dial normally, but no dial tone whatsoever. Any ideas??? Thanks in advance. Alejandro Sosa.
Re: [Asterisk-Users] R2 support
Hi, I make it compile. I just never finished it. :-) What I did to make R2 work properly was throw away the code that is now in CVS at Digium, and write a new implementation from scratch. I guess that is not quite the answer you wanted to hear. :-( Regards, Steve Jorge Verastegui wrote: Hi I know there is no support for R2, but I succcessfully compiled and ran the libr2. However, I am not able to initiate calls. The error I get is Couldn't call g3/71605538 -- Hungup 'Zap/32-1' == Everyone is busy at this time I understand that the idle signaling is not working right, any ideas on what I can do to fix this problem? Looking forward to your responses. Regards, Jorge On Fri, 2004-04-30 at 22:58, Steve Underwood wrote: jorge verastegui wrote: Hi i have successfully downloaded and compiled libr2 from source. But i dont seem to find how to properly configure it. When i run it (partcially unconfigured) the following error occurrs Signalling requested is R2 Signalling but line is in PRI Signalling signalling This error is easy to fix by changing ccs to cas, and removing crc4, in your zaptel.conf file. However libr2 does not work. It is a partly implemented solution which I abandoned. It only gets you about 10% of the way to a working R2 system :-( My current R2 software is completely different. Regards, Steve ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P Hang up detection
I have a problem that incoming calls on a T100P can be hung up by the remote party and the card does not seem to notice the call drop by the remote side. Watching this from the remote T1 side you can see the trunk goes idle after caller drops call, than * actually seizes the trunk again and makes it busy until it times out and hangs it up. Settings for the card are: /etc/zaptel.conf span=1,0,0,d4,ami fxsks=1-24 /etc/asterisk/zapata.conf group = 1 signalling=fxs_ks context=pan_pbx channel = 1-24 busydetect=yes Any one have some helpful tips? -- respectfully, Joseph - (606) 477-2355 x140 --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR with chan_h323 module
Hi there, I have the following scenario: I have a GK that controls my H323 GWs. And, I am needing an IVR royalty-free system. More than an IVR, it's only something that should play a sound file depending on the called digits. So, I wanted to use asterisk as my IVR. So, I've installed asterisk with the chan_h323 module and used the Playback comand in my extensions.conf file. It worked, but the problem is that in order to listen to the sound file, the call needs to be answered first. That is not good for me. Because, when the call gets answered, the end user is charged for that call because my H323 GW receives a Connect message. And these calls should not be charged. I have also tried the Playback comand this way: exten = 1007,1,Playback(/home/test|noanwer) and it did not work. Can someone give me a sugestion on how to solve this problem? thanks in advance, Pablo Salinas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BGM Music
Thinking about it further you could set the 6th line to autoanswer and have the pbx call you and play MOH when none of your lines on the asterisk box are in use. On Thu, 2004-05-13 at 10:57, Joseph wrote: Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BGM Music
Sure, create an extension that has on-hold music and dial it on the speaker phone using the second line. [mohtest] exten = 22,1,Ringing exten = 22,2,Answer exten = 22,3,MusicOnHold,classic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Thursday, May 13, 2004 10:58 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] BGM Music Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? -- respectfully, Joseph - (606) 477-2355 x140 --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2.05a firmware
- Original Message - From: Justin Carlson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 12, 2004 5:21 PM Subject: [Asterisk-Users] 2.05a firmware where can I get the 2.05 firmware all i see is the 2.04 firmwares :-) http://www.snom.com/download/share/snom200-2.05a-SIP.bin Christian Hoffmeyer YottaDot Solutions Huntsville, AL (iax) 700.859.4508 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Losing my PRI Interface every 20-30 minutes???
On Wed, 2004-05-12 at 23:07, Shad Mortazavi wrote: May 13 03:27:51 WARNING[1192437440]: chan_zap.c:5834 zt_pri_error: PRI: Read on 49 failed: Unknown error 500 I call this The Dreaded 500 Error. You didn't search the mailing list archives before you posted did you? Most people do NOT get this error, a few do. I don't recall seeing a solution to the problem, but check the mailing list archives. To search the Asterisk mailing list archive go to www.google.com and put site:lists.digium.com in addition to your other query terms. -- Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the Unofficial Links) and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to play dialtone on channel xx (Zaptel TE405P)
- Original Message - From: Alejandro Sosa To: [EMAIL PROTECTED] Sent: Thursday, May 13, 2004 10:15 AM Subject: [Asterisk-Users] Unable to play dialtone on channel xx (Zaptel TE405P) The problem I'm having is that when I pick up any of the analog handsets I get no dial tone. Switching works, I can place and receive calls on both analog and MGCP handsets. The MGCP phones give me dial tone fine when I take them off the hook, only the analog ones don't do it. Any ideas??? - Let's see the relavent sections of your zaptel.conf and zapata.conf files. Christian Hoffmeyer YottaDot Solutions Huntsville, AL (iax) 700.859.4508 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BGM Music
Interesting. Would that call take a lot of * resources? Being up all the time... On Thu, 2004-05-13 at 11:37, Joseph Finley wrote: Sure, create an extension that has on-hold music and dial it on the speaker phone using the second line. [mohtest] exten = 22,1,Ringing exten = 22,2,Answer exten = 22,3,MusicOnHold,classic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Thursday, May 13, 2004 10:58 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] BGM Music Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? -- respectfully, Joseph - (606) 477-2355 x140 --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BGM Music
On Thu, May 13, 2004 at 10:57:30AM -0400, Joseph spake thusly: Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Why would you want to? The sound quality is horrible for music even on a good speakerphone. You've probably got a computer and decent speaks right there, why not just fire up xmms? -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] R2 support
Are there any signaling converters? From R2 to something which is supported in asterisk ? Bart - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 13, 2004 12:17 PM Subject: Re: [Asterisk-Users] R2 support Hi, I make it compile. I just never finished it. :-) What I did to make R2 work properly was throw away the code that is now in CVS at Digium, and write a new implementation from scratch. I guess that is not quite the answer you wanted to hear. :-( Regards, Steve Jorge Verastegui wrote: Hi I know there is no support for R2, but I succcessfully compiled and ran the libr2. However, I am not able to initiate calls. The error I get is Couldn't call g3/71605538 -- Hungup 'Zap/32-1' == Everyone is busy at this time I understand that the idle signaling is not working right, any ideas on what I can do to fix this problem? Looking forward to your responses. Regards, Jorge On Fri, 2004-04-30 at 22:58, Steve Underwood wrote: jorge verastegui wrote: Hi i have successfully downloaded and compiled libr2 from source. But i dont seem to find how to properly configure it. When i run it (partcially unconfigured) the following error occurrs Signalling requested is R2 Signalling but line is in PRI Signalling signalling This error is easy to fix by changing ccs to cas, and removing crc4, in your zaptel.conf file. However libr2 does not work. It is a partly implemented solution which I abandoned. It only gets you about 10% of the way to a working R2 system :-( My current R2 software is completely different. Regards, Steve ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2.05a firmware
- Original Message - From: Justin Carlson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 12, 2004 5:21 PM Subject: [Asterisk-Users] 2.05a firmware where can I get the 2.05 firmware all i see is the 2.04 firmwares :-) Looks like 2.05b is out. http://www.snom.com/download/share/snom200-2.05b-SIP.bin Christian Hoffmeyer YottaDot Solutions Huntsville, AL (iax) 700.859.4508 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan Capi error
I am using chan_capi 0.3.1 and I am experiencing a strange problem, I have # transfer enabled for inbound calls to * however the * does not detect the # being pressed, it is just passed out as a tone, inbound calls # transfer works fine. and with every tone sent a warning appears in the console log Unknown RTP codec 109 received Is this a bug in chan_capi ? Jason Williams ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan Capi error
Just read my own message and it is not clear. An inbound call to * via a BRI when the caller presses # asterisk detects # and plays transfer message when the call recipient (on SIP) presses # the tone is sent out over the BRI rather than detected by * and the error displayed as below. Jason At 07:32 13/05/2004 +0100, you wrote: I am using chan_capi 0.3.1 and I am experiencing a strange problem, I have # transfer enabled for inbound calls to * however the * does not detect the # being pressed, it is just passed out as a tone, inbound calls # transfer works fine. and with every tone sent a warning appears in the console log Unknown RTP codec 109 received Is this a bug in chan_capi ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Questions
hi, c) We are getting some NOTICE: sched.c:218 sched_settime: Request to schedule in the past?!?! on the CLI, we don't know yet its cause or what it means. I had the same messages with * 0.9.0 but found no problems caused by that. The notice vanished after I switched from a 400 MHz CPU to a 2,2 GHz CPU, so the notice could just mean unexpected slow machine. ;-) Regards Andreas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP channel problem
Hello I have a problem with my MGCP voice gateway. I use D-Link DG104S Boot PROM Version 3.0B38-D Firmware Version3.0T86-D I tried asterisk v 0.7.2 and I am using latest CVS version now. When I dial a number very fast, or when I use a redial function, my asterisk receives coupled digits. My co-worker called number 245005111, these are a few lines of my debug. The identifier of first digit (2) is 7152 then asterisk received second digit (4), identifier 7153 and then asterisk received third digit... (2) with identifier 7152 so, asterisk dialed number 24254.. all debug is in attachment 1 headers, 0 lines Urgent handler MGCP read: NTFY 7152 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 N:[217.66.161.5]:2427 X:23f9c13c O: 2 from 217.66.161.122:2427Verb: 'NTFY', Identifier: '7152', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 4 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 7152 OK to 217.66.161.122:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '2' -- MGCP Asked to indicate tone: ro on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 8819 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 23f9c13c R: hu(N), hf(N), D/[0-9#*](N) S: ro to 217.66.161.122:2427 Urgent handler MGCP read: NTFY 7153 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 N:[217.66.161.5]:2427 X:23f9c13c O: 4 from 217.66.161.122:2427Verb: 'NTFY', Identifier: '7153', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 4 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 7153 OK to 217.66.161.122:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '4' -- MGCP Asked to indicate tone: ro on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 8820 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 23f9c13c R: hu(N), hf(N), D/[0-9#*](N) S: ro to 217.66.161.122:2427 Urgent handler MGCP read: NTFY 7152 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 N:[217.66.161.5]:2427 X:23f9c13c O: 2 from 217.66.161.122:2427Verb: 'NTFY', Identifier: '7152', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 4 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 7152 OK to 217.66.161.122:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '2' -- MGCP Asked to indicate tone: ro on aaln/[EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 8821 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 23f9c13c R: hu(N), hf(N), D/[0-9#*](N) S: ro to 217.66.161.122:2427 Urgent handler MGCP read: NTFY 7154 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 N:[217.66.161.5]:2427 X:23f9c13c O: 5 from 217.66.161.122:2427Verb: 'NTFY', Identifier: '7154', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 4 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] This problem is only while my colleague is downloading any data from internet. The voip gateway is on the same internet line as colleague's computer. I have these problemes everywhere with higher latence. Can I set digit report on my MGCP gateway to block mode ? I tried it, but no effect. I changed xgcp set_digit_report to 1 But it doesn't work :( My MGCP gateway always reports DTMF in comma separated. Can you help me please ? Thank you Vit Bohacek debug.txt Description: Binary data
Re: [Asterisk-Users] Help with initial setup
Tony Are you able to make this configuration work with 2 sip phone on same Asterisk server? I am also trying to do the same using xlite softphone abailable on www.xten.com site. Please let me know wgat you did? Thanks Deepak Quoting Tony [EMAIL PROTECTED]: On Sun, 2004-05-09 at 18:51, [EMAIL PROTECTED] wrote: Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid=Me 2124 [phone2] type=friend ;secret=blah host=dynamic defaultip=192.168.1.107 dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid=Mini Me 2123 And in extensions.conf at the very end: [sip] exten = 1,1,Dial(SIP/phone1,20,tr) exten = 2,1,Dial(SIP/phone2,20,tr) exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr) These are budgetone 102's, so I've then proceeded to their admin interface, and told them that the sip server is: 192.168.1.13. For phone1, all I've set is the sip id/username as phone1 and likewise for phone2 on phone number two. Rebooted.. But I do not seem to be able to get them to talk to asterisk. When issuing a sip show peers in asterisk, it displays: Name/usernameHost Mask Port Status phone2/phone2192.168.1.107 (D) 255.255.255.255 5060 Unmonitored phone1 (Unspecified) (D) 255.255.255.255 0 Unmonitored And when a sip show registry is issued, nothing seems to be connected: Host Username Refresh State Could there be something I'm missing in order to get the very basic working and then expand on that? Thanks in advance. Matthew Matt has made a great start page - http://astguiclient.sourceforge.net/scratch_install.html Just change the ip's to match your own - you'll be going in minutes! t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP calls-per-second performance test tool
At 11:39 AM -0700 on 5/12/04, Chris A. Icide wrote: On 01:16 PM 5/10/2004, John Todd wrote: http://sipp.sourceforge.net/ Anyone care to throw this at Asterisk to see what happens? I would, but I am having significant temporal shortfalls recently due to the apparent warping of the space/time continuum when I answer the phone with clients/associates. It seems that entire days pass by before I hang up... very odd, and very counter-productive to getting good Asterisk work done. JT JT, I ran this against my home office asterisk box (4 analog lines, about 20 sip UA's, 2.6G P4, 512MB system). I just ran the basic test, routing the request to Playback(invalid) then Hangup. During the test I had two UA's (a cisco 7960 and an analog phone connected to an ATA 186) dialed into MoH. Asterisk was running in background with no options to the command line, and one remote CLI connection. The system was able to handle 20 calls per second without any call failures. Beyond 20 calls per second I began to see call failure. The quality of the two MoH calls was perfect the entire time. I then proceeded to crank up the call volume and right about 200 calls per second, all call attempts became failures, and no new calls succeeded). At this point I got some interesting errors on the CLI related to maximum file descriptors (which I didn't worry too much about at the time), however, when I cranked the call volume back down to under 20 cps, all calls still failed. I had to shut down asterisk and restart to restore the system. However on an interesting note, at no time during any of the tests did the MoH calls lose quality or suffer any artifacts. Interesting program, and I'll set up a much more scientific test system and post some results on multiple systems (1G Pentium, 2.6G Pentium, and a Dual AMD system on 2.4 and 2.6 kernels) sometime soon. -Chris Chris - It does not appear that sipp is a User Agent that is authenticated, which is probably something that needs to be included in the tests, since that adds ~30% additional packet chatter on an INVITE, and involves some computation which could significantly change the results of what SIPP finds vs. real-world situations. More investigation would lead me back to sipsak (http://sipsak.berlios.de/) to see if perhaps some grafting of the two packages could be made, such that the extended features of sipsak (including authorization) could be programmed to include the RTP echo module and end-to-end mode that sipp appears to support. I'm not sure which program would be better to modify... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM200 + 2.05a Firmware + G729b BUG
If i use the snom 200 with firmware 2.05a (not tested with 2.04) and the G739b codec. Then the keys on the snom do not work with gsm it is ok. greetings nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM200 + 2.05a Firmware + G729b BUG
If using gsm, comment out the dtmfmode line in your sip.conf entity (i.e. take the default) and it should work fine. Tan Telappliant.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nicolas Sent: 13 May 2004 09:22 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SNOM200 + 2.05a Firmware + G729b BUG If i use the snom 200 with firmware 2.05a (not tested with 2.04) and the G739b codec. Then the keys on the snom do not work with gsm it is ok. greetings nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: X100P keeping PSTN line Offhook
What did you set your busy count variable to when the calls started to drop? I had the same issue until I changed it to 6 from 4 and so far everything seems to be working fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shahid Sent: Monday, May 10, 2004 7:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: X100P keeping PSTN line Offhook Tom, Rich and Atif, Regarding your responses, 1. I have previously tried the callprogrees=no. Doesnt solve the problem. 2. If busydetect=yes, calls to PSTN get droped in the middle of the conversations. 3. Havent looked into the MOH thingy. This feature has caused me other problems. Thinking of turning it off altogether. Anyone has any ideas about alternatives ? Thanks for all your help guys. Regards -shahid Shahid [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! pbx1*CLI zap show channel 1 Channel: 1 File Descriptor: 31 Span: 1 Extension: Context: bell Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Offhook = zapata.conf == busydetect=no musiconhold=default group=1 pickupgroup=1 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel = 1 pickupgroup=1 immediate=no signalling=fxs_ks callerid=asreceived channel = 2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel does not compile on latest RHEL kernel
Hi After updating some Red Hat Enterprise linux machines to the latest RHEL kernel (RHEL crashes on vanilla kernels :)), I get tons of errors when trying to compile zaptel: In file included from tor2.c:30: /usr/src/linux-2.4/include/linux/kernel.h:61: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:61: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:61: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:62: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:62: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:63: `panic_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:63: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:69: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:69: `simple_strtoul_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:69: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:70: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:70: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:70: `simple_strtol_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:70: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:71: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:71: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:71: `simple_strtoull_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:71: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:73: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:73: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:74: `sprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:74: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:75: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:75: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:75: `vsprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:75: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:76: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:76: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:77: `snprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:77: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:78: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:78: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:78: `vsnprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:78: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:80: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:80: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:81: `sscanf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:81: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:82: `vsscanf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:82: warning: parameter names (without types) in function declaration /usr/src/linux-2.4/include/linux/kernel.h:86: `get_option_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:86: warning: parameter names (without types) in function declaration /usr/src/linux-2.4/include/linux/kernel.h:87: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:87: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:87: `get_options_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:87: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:88: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:88: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:88: `memparse_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:88: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:96: invalid suffix on
Re: [Asterisk-Users] Re: G729 Segmentation fault
nicolas wrote: Thanks but do not solve it: [app_datetime.so] = (Date and Time) == Registered application 'DateTime' [codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator) Warning, flexibel rate not heavily tested! Cannot allocate channels... Process Stopped! Error -11 You must register the codec in order to be able to use it. Michael. May 12 19:27:42 WARNING[16384]: codec_g729b.c:511 load_module: Unable to initialize va stuff: -1 Segmentation fault alberspilnx8:/bin # Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe greetings nico Michael Manousos wrote: Do not start Asterisk from within a directory that contains a 'tmp' subdirectory. Michael. nicolas wrote: I have Now a G729 codec license and when i start it comes: [format_g729.so] = (Raw G729 data) == Registered file format g729, extension(s) g729 [app_datetime.so] = (Date and Time) == Registered application 'DateTime' [codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec [Translator) sh: line 1: tmp: Is a directory rm: cannot remove `tmp': Is a directory Cannot allocate channels... Process Stopped! Error -11 May 12 18:40:08 WARNING[16384]: codec_g729b.c:511 load_module: Unable to initialize va stuff: -1 Segmentation fault alberspilnx8:~ # Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Can anyone help me ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ./M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions in mysql
I have already set up a mysql server and I can already use the sip configuration from mysql, but I'm still having problem with my extensions in mysql. I have followed the instructions in http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql but I still can't use the extensions in mysql. From the sip-mysql , I had to add dbname, dbhost, dbuser and dbpass in the sip.conf under the general entity. I also added these four in the extensions.conf, but it still doesn't work. In using sip-mysql, i have to enable SIP_MYSQL_FRIENDS. Is there anything I have to do like the one in sip-mysql to be able to use extensions in mysql? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax behind a SonicWall
At 8:23 PM -0600 on 5/12/04, Rich Adamson wrote: Current dev cvs install on two systems. System A is behind a SonicWall firewall, and system B is on a registered IP address. (System B has multiple iax links that are fully functional to multiple locations.) System A is correctly registering with System B, with no special firewall rules. Should System B be able to take advantage of the registration to send iax/gsm calls to System A without installing any firewall rules? I assumed it could, but an ethereal trace indicates a new call is placed from B - A, but A never acknowledges the iax2 packet, etc. The trace suggests the registration is happening with src port 28277 (or whatever) - dest port 4569 however, calls are processed with src port 4569 and dest port 4569 Shouldn't we expect src=4569 and dest=4569 on all iax2 interactions? Rich If src=4569 and dst=4569 always, then it would be impossible to have more than one IAX2 talker behind a firewall talking to an external Asterisk server, right? There would be no method by which the firewall would know which packet was destined for what device inside the firewall, since the source port and destination port would be the same for each connection. I'm not thinking this through completely, and it seems like there is a flaw in this argument... but with UDP, there is no sequence number that should have attention paid to it (like TCP) so... er... someone tell me why this is incorrect. note: firewall in this case is really NAT, right? Hi John, Using src=4569 and dst=4569 is not a problem with any firewall as long as the destination IP address differs. The firewall's nat table for two different iax links would look something like: Src: 1.2.3.4 udp 4569 Dst: 5.6.7.8 udp 4569 Src: 1.2.3.4 udp 4569 Dst: 6.7.8.9 udp 4569 Since nat table entries always include all four values (regardless of firewall vendor), there is always uniqueness for the sessions from the firewall's perspective. In the case of iax, if two or more sessions were attempted between like addresses (as in two iax calls), the firewall would not be aware that two sessions were even happening as the udp src dst header data is identical. Asterisk knows the two sessions are different as the iax2 header includes source call: 5 and Destination call: 0 to differentiate the packets (as real examples). If we look closely at a working iax trunk call today, the src and dst ports are actually the same (udp 4569). Its only the initial registration process that actually uses a non-4569 source port number. Following that initial registration, even the registration refresh packet exchange uses src and dst port of 4569. So, it almost looks like either: a) everyone has been content to install a firewall rule to handle inbound udp 4569, or, b) no one has recognized that if the registration process used udp 4569 for its src port, no changes would be needed to any firewall, or, c) there is something wrong with my logic. Since I do a lot of protocol analysis and network security work (as a professional), I'm 98% convinced b is probably correct. If no one can point out the flaw in that logic, I'm tempted to open a bugtracker item to change it. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with analog interface to PBX
On Wed, 12 May 2004, Dan Fernandez wrote: Asterisk should answer the call, playback a message, dial another PBX extension and if no one answers dial another extension (via IAX). The first problem I ran into was that the Flash application doesn't really work. To get around this I added another x100p to dial the new extension. The problem I ran here was that even though I specified in the Dial app to just dial for 30 seconds, it rang forever as if * cannot recongnize that no one had picked up. Asterisk does seem to detect hangups and busy tones (I have busydetect=yes and busycount=10) In the absence of call progress detection settings, Zap analog channels tell Dial() that they are Connected more-or-less as soon as they have completed dialling (I see this on the display of my 7960: I see Proceeding for a second or two, then Connected, when I dial through an X100P). So, the timeout on your Dial() never gets triggered because the channel reports a connected call almost straight away. To do what you want, you would need callprogress=yes -- as long as your Panasonic PBX generates authentic US tones. busydetect will only detect busy (!), not ringback or congestion or any of the other tones you would need to make your application work the way you want -- call progress detection tries to do this for you. The bad news is that even if your PBX generates US tones, reports are that the detection is not too reliable. Am I trying to do something that the x100p is not capable of? Would making changes to the indications.conf help at all? It's not that the X100P can't do the job, it's more that analogue lines can't do the job :) Seriously, if your PBX generates US tones then give callprogress=yes a try. From my reading of the code, the tones specified in indications.conf are unrelated to the way the * DSP does call progress detection (have a look at functions like ast_dsp_call_progress() in dsp.c if you're really curious). 2) I would also like to use * for voicemail. The user should be able to dial the extension where the x100p is connected and asterisk recognized the extension the user is dialing and request for the password? Is this possible? On an analogue channel via an X100P, there is no called number indication. So you can't tell what number the caller dialled to reach you. If you wanted to use the * box as a voicemail-only machine, you could drop the caller straight into VoiceMailMain, but if you wanted other functions (conference rooms, VoIP gateway, etc) you would need to use an IVR... press 1 to access Voicemail... press 2 to reach a Voice-over-IP user... press 3 to join a conference... ... This doesn't really help your original need: to dial another number on the PBX and get control back if needed. If callprogress=yes doesn't work for you, you could try something like this (off the top of my head): exten = 4,1,Playback(trying-press-*-to-come-back) exten = 4,2,Dial(Zap/1/1234,,Hg) exten = 4,3,Goto(103) exten = 4,103,Playback(sorry-cant-reach) exten = 4,104,Goto(menu,s,1) On the Dial(), the option H enables caller hangup using '*', and g says go on in context when the destination channel hangs up. This would put your caller in the driver seat and get them to do the tone detection for you ;) Hope this helps, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sonicwall with Firmware 6.6.02 - SIP?
Sonicwall now has SIP transformations check box in the Access section of the interface - does anyone know how to make sense of this function? Tech support is useless, and the help description is confusing. Using on office network to connect Grandstream and Cisco phones to asterisk PBX at remote location. I hate to use linksys or belkin, but they're (ironically) the only 2 that have worked so far...sure hate to use crappy equipment for mission critical stuff. I'd have to guess (no direct experience) the sip transformation means the firewall will track the sip/rtp port negotiation process, and automatically open the negotiated rtp ports for audio. I'd also have to guess the function will only work under some specific conditions as the sonicwall engineering process seems to only involve limited conditional testing before releasing code to the public. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Consultive Transfer, or faking it
Hi there... I have a simple * setup with about 11 Soft phones (SJ Phone). The clients don't support a consultive or supervised transfer (I believe that's what it is called). Tris is a feature much desired by the powers that be and they want me to make it work :) I was wondering if there was a way to do this with and AGI script or the like so that when Staff 1 gets an external call and wants to put it through to Staff 3, they simply transfer to the person's extension, but that actually connects them to an AGI script which first links the 2 staff members, and then requires another key to be pressed to link Staff 3 with the external call. Has anyone done this or knows how to do it, or something similar? Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where are the list archives??
Hi there, because yesterday I had a problem with my email, I wanted to check the replies (if any) to my question Needed Open ports on the archives but... where are the ones from may?? http://lists.digium.com/pipermail/asterisk-users/2016-May/thread.html I only see 3 posts.. is this the normal behaviour? Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where are the list archives??
-Original Message- From: [EMAIL PROTECTED] Sent: 13 May 2004 10:56 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Where are the list archives?? Hi there, because yesterday I had a problem with my email, I wanted to check the replies (if any) to my question Needed Open ports on the archives but... where are the ones from may?? http://lists.digium.com/pipermail/asterisk-users/2016-May/thread.html I only see 3 posts.. is this the normal behaviour? Thanks, Martin It is for May 2016 :-) Try 2004: http://lists.digium.com/pipermail/asterisk-users/2004-May/thread.html -Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Missing digits on TDM400P incomplete dial string - DTMF problem?
I have the same problem in Spain, but I did not start to change any config.
Is this problem because of the phone or because of the TDM card? Did u get a
solution?
Hiep
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] Im Auftrag von bam
Gesendet: Freitag, 7. Mai 2004 19:31
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] Missing digits on TDM400P incomplete dial
string - DTMF problem?
I turned down the rxgain and txgain to -22.0 and -16.0 respectively and
things started to look a whole lot more acceptable. Then the client
sticks
on his BT DECT phone and I start losing all the 1s from the dial string.
Does anyone know if BT DECT phones have dodgy DTMF tones?
At 17:19 07/05/04, you wrote:
I've had a quick fiddle to little avail, the readings looked prey good
to
be honest before I started fiddling. Looking a little closer it appears
that it is the digit 1 that is being lost more that any other.
At 15:25 07/05/04, you wrote:
Run /usr/src/zaptel/ztmonitor 32 -v
And adjust your gains in /etc/asterisk/zapata.conf accordingly.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bam
Sent: Friday, May 07, 2004 3:35 AM
To: [EMAIL PROTECTED]
Subject: [SPAM] - [Asterisk-Users] Missing digits on TDM400P incomplete
dial string - Email found in subject
We are experiencing problems on a FXS interface where the client is
dialing
numbers, but digits are being dropped somewhere from the dial string.
Typically one or two digits are not being presented. We've tried
different
handsets to no avail, and I am assuming that it is some sort of timing
problem.
Are there any parameters I can tweak to try and rectify this?
zapata.conf
context=hardwire
group=3
signalling=fxo_ks
mailbox=8765
callerid=Acme 8765
channel=32
extensions.conf
[hardwire]
;
exten = _NXX,1,SetCallerID(0141411${CALLERIDNUM})
exten = _NXX,2,CallingPres(3)
exten = _NXX,3,Dial(Zap/g1/0141${EXTEN})
exten = _0.,1,SetCallerID(0141411${CALLERIDNUM})
exten = _0.,2,CallingPres(3)
exten = _0.,3,Dial(Zap/g1/${EXTEN})
exten = t,1,Hangup ; If they take too long, give
up.
exten = i,1,Hangup ; If they get it wrong, give up
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[Asterisk-Users] DASS2 support
My employer wants to use Asterisk, but the E1 circuit providing the current phone system is DASS2 rather than ISDN30. Converting the E1 to ISDN30 is not a practical proposition at this time as it'd stop the legacy phone system from working. Is there any sort of hardware support for DASS2? I speculate that the E100P should be able to deal with the electrical side of it, but I'm unsure of driver support. Has anybody got Asterisk to work with DASS2 circuits? Thanks in advance. -- There are three reasons for becoming a writer: the first is that you need the money; the second that you have something to say that you think the world should know; the third is that you can't think what to do with the long winter evenings. - Quentin Crisp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicetronix's OpenPort4 ANyone?
Thanks Ben.
Wojtek
- Original Message -
From: Ben Kramer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, May 12, 2004 9:05 PM
Subject: Re: [Asterisk-Users] Voicetronix's OpenPort4 ANyone?
Hi Wojtek,
you can call a single port like this:
exten = _9XXX,1,Dial(vpb/1-1/${EXTEN:${TRUNKMSD}})
Or if you have groups defined in your vpb.conf you could so something
like this:
exten = _9XXX,1,Dial(vpb/g1/${EXTEN:${TRUNKMSD}})
Cheers,
Ben.
On Wed, 2004-05-12 at 21:13, Wojciech Tryc wrote:
I am looking for opinions and samples on how to call their ports from
the
extensions.conf file.
Regards,
Wojtek
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--
Ben Kramer [EMAIL PROTECTED]
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[Asterisk-Users] Taiwan calling!!!!
The internationalization of Asterisk moves on. Now, there are two patches in bugs.digium.com that needs your test report and feedback. http://bugs.digium.com/bug_view_page.php?bug_id=0001600 http://bugs.digium.com/bug_view_page.php?bug_id=0001599 These will add support for Taiwanese signals and indications in zaptel and other parts of asterisk. Please test them and report the results on respective bug report page. Thank you! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DASS2 support
Peter Corlett wrote: My employer wants to use Asterisk, but the E1 circuit providing the current phone system is DASS2 rather than ISDN30. Converting the E1 to ISDN30 is not a practical proposition at this time as it'd stop the legacy phone system from working. Is there any sort of hardware support for DASS2? I speculate that the E100P should be able to deal with the electrical side of it, but I'm unsure of driver support. Has anybody got Asterisk to work with DASS2 circuits? Thanks in advance. * has no software support for DASS or DASS2 Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DASS2 support
Hi Peter, PC Has anybody got Asterisk to work with DASS2 circuits? As Steve said there is no native support for DASS2 within *. This leaves you with a couple of choices: a) Ask for a new ETSI Q.931 ISDN circuit from a new Telco (no risk to existing PBX). This can be at a very low cost if you go to a competitive carrier who wants your business! b) Obtain a protocol converter (not cheap). If you're interested in (b), follow the link below and take your pick (but put your cheque book on steroids first): http://tinyurl.com/2wzh8 HTH Darren PS. Please be aware that if you order a new circuit in the UK you should specify ISDN30e to ensure you receive an ETSI compliant circuit that will work with *. -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: 13 May 2004 13:27 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DASS2 support Peter Corlett wrote: My employer wants to use Asterisk, but the E1 circuit providing the current phone system is DASS2 rather than ISDN30. Converting the E1 to ISDN30 is not a practical proposition at this time as it'd stop the legacy phone system from working. Is there any sort of hardware support for DASS2? I speculate that the E100P should be able to deal with the electrical side of it, but I'm unsure of driver support. Has anybody got Asterisk to work with DASS2 circuits? Thanks in advance. * has no software support for DASS or DASS2 Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell server for asterisk question!
I second that warning to stay away from the perc raid, I have one that continuously deals me fits. Leo Ann Boon wrote: The TE410P works with the 2650, I had 1 in there for months. One other thing, avoid the PERC RAID. The Linux driver in kernel 2.4 series is not very stable. FYI. Bartosz Jozwiak wrote: I am planning to buy Dell 2650 server with dual Xeon processors. And I would like to buy two TE410P cards for PCI with 3,3v. This is on Dell site about PCI slots for Dell 2650 server: 3 PCI-X (1x64-bit/133MHz, and 2x64-bit/100MHz) Does that mean I will be able to buy two TE410P cards ? Or I need to buy two TE405P cards ? Thanks for help. bartek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell server for asterisk question!
On Thu, 13 May 2004, Jeff Roberts wrote: I second that warning to stay away from the perc raid, I have one that continuously deals me fits. I've got a couple dozen of them and never had any problems. They are running everything from redhat 6.2 to fedora to rhel 3. dave Leo Ann Boon wrote: The TE410P works with the 2650, I had 1 in there for months. One other thing, avoid the PERC RAID. The Linux driver in kernel 2.4 series is not very stable. FYI. Bartosz Jozwiak wrote: I am planning to buy Dell 2650 server with dual Xeon processors. And I would like to buy two TE410P cards for PCI with 3,3v. This is on Dell site about PCI slots for Dell 2650 server: 3 PCI-X (1x64-bit/133MHz, and 2x64-bit/100MHz) Does that mean I will be able to buy two TE410P cards ? Or I need to buy two TE405P cards ? Thanks for help. bartek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell server for asterisk question!
I think those warnings are silly. We have a perc control that's been in service for 3+ year without 1 OUNCE of trouble. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jeff Roberts Sent: Thursday, May 13, 2004 8:55 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dell server for asterisk question! I second that warning to stay away from the perc raid, I have one that continuously deals me fits. Leo Ann Boon wrote: The TE410P works with the 2650, I had 1 in there for months. One other thing, avoid the PERC RAID. The Linux driver in kernel 2.4 series is not very stable. FYI. Bartosz Jozwiak wrote: I am planning to buy Dell 2650 server with dual Xeon processors. And I would like to buy two TE410P cards for PCI with 3,3v. This is on Dell site about PCI slots for Dell 2650 server: 3 PCI-X (1x64-bit/133MHz, and 2x64-bit/100MHz) Does that mean I will be able to buy two TE410P cards ? Or I need to buy two TE405P cards ? Thanks for help. bartek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with analog interface to PBX
On Wed, 12 May 2004, Dan Fernandez wrote: Folks, For the last few days I've been trying to experiment with a Panasonic PBX and an X100P but have run into quite a few problems which I am not sure if they can be solved with this type of card (how about TDM01B?) 1) I wanted to use *'s IVR capabilities, so I routed the calls to the extension where the x100p was connected to. Asterisk should answer the call, playback a message, dial another PBX extension and if no one answers dial another extension (via IAX). The first problem I ran into was that the Flash application doesn't really work. To get around this I added another x100p to dial the new extension. The problem I ran here was that even though I specified in the Dial app to just dial for 30 seconds, it rang forever as if * cannot recongnize that no one had picked up. Asterisk does seem to detect hangups and busy tones (I have busydetect=yes and busycount=10) For about 6 months, we were using the same logical setup (a channelbank of FXO cards for a Merlin Legend switch, with asterisk doing incoming IVR / autoattendant, then transferring the calls out to the Legend, and handling voicemail). The first problem I encountered that I hadn't expected had to do with asterisk transferring the call back to the Legend. I did a Flash(), a SendDTMF(), and another Flash() - the Legend saw this as an attended transfer, and it caused some oddities. Turns out I needed to Flash(), SendDTMF(), Hangup(). Along the way, I found the Flash times that the legend was expecting to see, and adjusted them in the source code, so as to eliminate occasional flash detection problems. I'd take time to plug an analog set into the extension you have the X100P on, and make sure you can flash/transfer calls like you're expecting asterisk to. There's no reason (that I know of) that your flash can't give you exactly the behavior you're looking for. Good luck to you, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budgetone iLBC to IAX2 iLBC
I have updated my budgetone phone to the 1.04.63 firmware and am trying to use the iLBC codec for IAX2 passthrough to my PSTN breakout provider. The IAX2 gateway is set to use only iLBC. However, the SIP connection from the budgetone to the * server will use any other codec in preference to iLBC. If I force the SIP connection to be iLBC only, then it will make an iLBC connection. This makes me think that the iLBC codec in asterisk has a different sampling rate or payload size to default budgetone values. Anyone tell me what payload and sampling rate * uses for iLBC? Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g.729 - licenses and opinions
Its extortion in my bookI've been told horror stories from 1st party sources about how Voiceage negotiates with their potential customers. Then most of us know how much of PITA Voiceage has made codec_g729b.so, just so they can soak every nickel they possibly can out of Digium. I don't think the $10/port is so much an issue for anyone as it is that it's so amazingly hard (impossible) to transfer those licenses to new machines. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BGM Music
You make a good point. However in a corp environment where there are many users, one distraction is users playing music on there pc's. So the in this case corp has decided that one music can be provided via the phone. If the user wants something to listen to, that is available. Otherwise there is no music. So corp has also decided sound on pc's shall be disabled. Anyway, I just wanted to hear what the options were. On Thu, 2004-05-13 at 11:53, Tracy R Reed wrote: On Thu, May 13, 2004 at 10:57:30AM -0400, Joseph spake thusly: Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Why would you want to? The sound quality is horrible for music even on a good speakerphone. You've probably got a computer and decent speaks right there, why not just fire up xmms? -- respectfully, Joseph --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2.05a firmware
Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP calls-per-second performance test tool
On Thu, 2004-05-13 at 03:58, John Todd wrote: Chris - It does not appear that sipp is a User Agent that is authenticated, Yes. which is probably something that needs to be included in the tests, since that adds ~30% additional packet chatter on an INVITE, and involves some computation which could significantly change the results of what SIPP finds vs. real-world situations. More investigation would lead me back to sipsak (http://sipsak.berlios.de/) to see if perhaps some grafting of the two packages could be made, such that the extended features of sipsak (including authorization) could be programmed to include the RTP echo module and end-to-end mode that sipp appears to support. I'm not sure which program would be better to modify... I like the feature of SIPP to be able to modify the UA using .xml scenarios. And SIPP do echo the received audio the problem is that it doesnt generate audio. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2.05a firmware
Does anyone know what kind of file needs to be uploaded for the custom ring tone? --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g.729 - licenses and opinions
On Thu, 2004-05-13 at 12:07, Andrew Kohlsmith wrote: Just remember that you were given those patents as incentive to invent so that ultimately your work would go into the public domain so we can all enjoy it. We are buying your work with our tax dollars by protecting it for a short period of time so you have a little monetary incentive. BZZZT! Wrong. He was given those patents as in incentive to invent something that he could SELL without everyone on the planet copying his hard work and competing on his idea. Patents put the process out in the public so that it's easy to see when someone's infringing. Lets please remember that this is a global mailing list now and the history of patents may be different from place to place. In the US, patent law is similar to copyright law. For a time you are given exclusive rights to your invention. You are able to charge money for it. You are able to do any number of useful things as the inventor. The tradeoff for patents is that at the end of the patent term, the public domain gets the benefits of your work. Our entire country is built upon a rich and diverse public domain. If one chooses to invent, yet does not choose to patent those inventions, they potentially loose any advantage of being the sole gateway to the invention. Look here and please don't be offended by the kid part, it isn't intentional just a good list. http://www.uspto.gov/go/kids/kidprimer.html 17 years for software patents is FAR too long, IMO, but that's an entirely different story. IMO software patents shoudln't be for more than ~24 months since the industry moves so blazingly fast. I'm of mixed feelings here. I don't like software patents at all, but without them, some of the voice compression that is out there would possibly not have been developed. What would have been the incentive for the telecoms to allow the public in on some of the voice compressions with out getting paid for the work. So while I think it is important, I also can't seem to draw a reasonable line. 24 months in most software isn't enough time from day 0 to make any reward for the work, at least not monetarily. What software project out there do you know had a major roll out sufficiently under 24 months from beginning of programming to have paid the programming staff off after say 1 year past the initial 24 months? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2.05a firmware
They also made a bad (for me) change. In 2.05a the phone would ring normally and I could press OK for headset or pick up the handset for handset. Now, when headset is enabled the phone only rings in the headset (i.e. not through speakerphone). --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g.729 - licenses and opinions
I totally agree software patents are far too long. 24 months seems fair... it also provides some incentives to make a better product. :) bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, May 13, 2004 12:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] g.729 - licenses and opinions Just remember that you were given those patents as incentive to invent so that ultimately your work would go into the public domain so we can all enjoy it. We are buying your work with our tax dollars by protecting it for a short period of time so you have a little monetary incentive. BZZZT! Wrong. He was given those patents as in incentive to invent something that he could SELL without everyone on the planet copying his hard work and competing on his idea. Patents put the process out in the public so that it's easy to see when someone's infringing. 17 years for software patents is FAR too long, IMO, but that's an entirely different story. IMO software patents shoudln't be for more than ~24 months since the industry moves so blazingly fast. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-doc Conference Call - phase 2 :)
Thank you to everyone who has offered so far! I've had formal offers from Martin List-Peterson, William Suffil, Greg Varga, Brian Cuthie and Ed Guy (hopefully I haven't forgotten someone!) Now we just have to decide where the best spot to host it is. What do you guys think? For this week, I don't care if this is a one off. At some point I'd like to have a weekly conference, and if we can get it hosted permanently once a week, that would be ideal. Right now, I'm more worried about just getting a conference going :) I'm going to make a GUESS that we are going to have between 10-15 people. Perhaps more? Maybe we can get a tally of who is expected to be there and then based on that we can decide on a location. The server should be both SIP and IAX accessible. Jared mentioned access via a 1-800 number or PSTN, but I'm not sure how practical, or necessary, that is. Again, your thoughts? Here is some things Jared dumped into the IRC channel the other day that we are going to try and focus on during the conference call: Layout Details (we can't go into too much detail about every possible soft phone/hard phone/voip provider) Goals (first good docs, then maybe get published) Submission process (mailing list? Website shows who's in charge of a certain section?) Focus (What do we want to focus on first? The intro and installation chapters?) Simplecity (Let's make sure a voip-newvie can get up and running, as long as they've used Linux before and know how to use a text editor.) Please feel free to add your suggestions. Tentatively the conference will be scheduled for Sunday evening North American time (I am EST, -0500 GMT). I'd like to try and get as much input as possible from people, but I realize we can't schedule around everyone. For now we will assume Sunday evening is good. In the future we can try a couple of different times so that it can be convenient for others. How are we going to record the thoughts of this conference? I'm a fairly fast typist, so I could attempt to record thoughts and idea's during the conference. Should we record it? At this point I'm going to open the floor to discussion! If you could reply via the asterisk-doc list, that would be best, this list already has too much traffic :) If you would like to contact me off list, feel free to email me, or get a hold of me on the #asterisk-doc IRC channel. Thanks in advance, Leif Madsen aka blitzrage ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] poll vs select in channel.c
Title: poll vs select in channel.c
Hello,
The v1-0_stable cvs release doesn't include the recent change ('poll' instead of
'select') in channel.c. Will it end up there any time soon, or we need to use
cvs head to pick up this change?
Thank you.
Alex Zarubin
Webley Systems
Re: [Asterisk-Users] iax behind a SonicWall
At 3:06 AM -0600 on 5/13/04, Rich Adamson wrote: At 8:23 PM -0600 on 5/12/04, Rich Adamson wrote: Current dev cvs install on two systems. System A is behind a SonicWall firewall, and system B is on a registered IP address. (System B has multiple iax links that are fully functional to multiple locations.) System A is correctly registering with System B, with no special firewall rules. Should System B be able to take advantage of the registration to send iax/gsm calls to System A without installing any firewall rules? I assumed it could, but an ethereal trace indicates a new call is placed from B - A, but A never acknowledges the iax2 packet, etc. The trace suggests the registration is happening with src port 28277 (or whatever) - dest port 4569 however, calls are processed with src port 4569 and dest port 4569 Shouldn't we expect src=4569 and dest=4569 on all iax2 interactions? Rich If src=4569 and dst=4569 always, then it would be impossible to have more than one IAX2 talker behind a firewall talking to an external Asterisk server, right? There would be no method by which the firewall would know which packet was destined for what device inside the firewall, since the source port and destination port would be the same for each connection. I'm not thinking this through completely, and it seems like there is a flaw in this argument... but with UDP, there is no sequence number that should have attention paid to it (like TCP) so... er... someone tell me why this is incorrect. note: firewall in this case is really NAT, right? Hi John, Using src=4569 and dst=4569 is not a problem with any firewall as long as the destination IP address differs. The firewall's nat table for two different iax links would look something like: Src: 1.2.3.4 udp 4569 Dst: 5.6.7.8 udp 4569 Src: 1.2.3.4 udp 4569 Dst: 6.7.8.9 udp 4569 Since nat table entries always include all four values (regardless of firewall vendor), there is always uniqueness for the sessions from the firewall's perspective. In the case of iax, if two or more sessions were attempted between like addresses (as in two iax calls), the firewall would not be aware that two sessions were even happening as the udp src dst header data is identical. Asterisk knows the two sessions are different as the iax2 header includes source call: 5 and Destination call: 0 to differentiate the packets (as real examples). If we look closely at a working iax trunk call today, the src and dst ports are actually the same (udp 4569). Its only the initial registration process that actually uses a non-4569 source port number. Following that initial registration, even the registration refresh packet exchange uses src and dst port of 4569. So, it almost looks like either: a) everyone has been content to install a firewall rule to handle inbound udp 4569, or, b) no one has recognized that if the registration process used udp 4569 for its src port, no changes would be needed to any firewall, or, c) there is something wrong with my logic. Since I do a lot of protocol analysis and network security work (as a professional), I'm 98% convinced b is probably correct. If no one can point out the flaw in that logic, I'm tempted to open a bugtracker item to change it. Rich I also suspect that B is correct, but let me clarify a bit... Let me understand your problem in a bit more detail: you're saying that even though your NAT is creating a mapping for 1.2.3.4 udp 28277 - 5.6.7.8 udp 4569 that this causes your NAT/FW to refuse return connections? Shouldn't your NAT automatically create that mapping and keep it open for some period of time? Or is Asterisk ignoring the 28277 src and sending the reply back on 4569? Thanks for the expanded discussion on NAT; that's helpful for the larger audience. My point in my message was that if I have two IAX devices (let's say, I have two IAXy's) behind the same NAT, and they're both pointed at the same non-NAT (external) Asterisk server, then that would not work. src: 1.2.3.4 udp 4569 dst: 5.6.7.8 udp 4569 src: 1.2.3.5 udp 4569 dst: 5.6.7.8 udp 4569 Packets coming from 5.6.7.8 might have internal (application-layer) flags that assign them to different devices, but the IP header information between packets for either device would be non-distinguishable by the NAT device. It would have a mapping for both IAXy's on the same port. Now, the way I've seen some NAT devices handle this is to give pseudo-random return ports to new sessions (new internal hosts) that request something that is already mapped, so that they can distinguish between return packets on the outside interface. The internal port mappings are kept the same. So, when the first packet is seen from IAXy #2 destined for the remote Asterisk server, the NAT has this internal thought: Whoa! We've already got a mapping for packets coming from 5.6.7.8:4569, so I'd better make a different mapping on my outside interface for this
RE: [Asterisk-Users] poll vs select in channel.c
I suspect you will have to run cvs-head to get that. Expect 1.0-RC1 and
1.1-RC1 to be released at the same time :)
bkw
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Zarubin
Sent: Thursday, May 13, 2004 12:44 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] poll vs select in channel.c
Hello,
The v1-0_stable cvs release doesn't include the recent change ('poll'
instead of
'select') in channel.c. Will it end up there any time soon, or we need to
use
cvs head to pick up this change?
Thank you.
Alex Zarubin
Webley Systems
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RE: [Asterisk-Users] Dell server for asterisk question!
On Thu, 2004-05-13 at 08:55, Jeff Roberts wrote: I second that warning to stay away from the perc raid, I have one that continuously deals me fits. Please take this to the actual PERC mailing list and see where you get. Dell is nice enough to host the list to try and fix the problem. Last I cared to look at it, it was just some 2650's and 1750's that had problems, and that was limited sometimes to certain firmwares on the drives. I think I've said it before on this list or another, but I've have not experienced any problems with the PERC3 or PERC4 controllers on our Dell PowerEdge 1650, 1750, 2600, and 2650 machines. All but three of our servers are running RHEL 3 including ten 1750 servers. Previously these same machines ran RHEL 2.1. Aside from the fact that RHEL 2.1 did not support the PERC out of the box (needed to use a driver disk from Red Hat during install) everything has been fine. -- Tony Kava Senior Network Administrator Pottawattamie County, Iowa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Voicemail: Strange Behaviour
Hi,
whenever I include a Ringing Line in some Voicemail Extension
I get an error when a call from the outside (via ISDN) comes in,
but it works when an internal (SIP-phone) calls the extension.
Here is my configuration for testing:
extensions.conf
[isdnext]
; strep external 101, our number and leave only extension
exten = _101XX.,1,Goto(default,${EXTEN:3},1)
[default]
exten = _50XXX,1,VoiceMailMain(${EXTEN:2})
exten = _51XXX,1,Ringing
exten = _51XXX,2,VoiceMailMain(${EXTEN:2})
extensions.conf
(just fyi: Voice-Mailbox 121 exists)
1. when local SIP-phone calls 50121 or 51121 it
gets the voicemail-password prompt. correct behaviour.
2. when external call via ISDN for 101-50121 comes in it
gets the voicemail-password prompt. correct behaviour.
3. when external call via ISDN for 101-51121 comes in, the
line ist hung-up immediately and the following error
messages are on the * console:
---
-- creating pipe for PLCI=0x101 msn = *
sent ALERT_REQ PLCI = 0x101
-- Executing Goto(CAPI[contr1/10151121], default|51121|1) in new stack
-- Goto (default,51121,1)
-- Executing Ringing(CAPI[contr1/10151121], ) in new stack
-- Executing VoiceMailMain(CAPI[contr1/10151121], 121) in new stack
-- CAPI Answering for MSN 0151121
May 13 20:22:11 ERROR[1114581936]: chan_capi.c:860 capi_write: dont know how to write
subclass 4
May 13 20:22:11 WARNING[1114581936]: res_adsi.c:163 adsi_careful_send: Failed to
carefully write frame
May 13 20:22:11 WARNING[1114581936]: res_adsi.c:205 __adsi_transmit_messages: Unable
to send CAS
May 13 20:22:11 WARNING[1114581936]: file.c:537 ast_readaudio_callback: Failed to
write frame
-- Playing 'vm-password' (language 'en')
May 13 20:22:11 WARNING[1114581936]: app_voicemail.c:2764 vm_execmain: Unable to read
password
== Spawn extension (default, 51121, 2) exited non-zero on 'CAPI[contr1/10151121]'
-- CAPI Hangingup
sent DISCONNECT_B3_REQ NCCI=0x10101
sent DISCONNECT_REQ PLCI=0x101
-- removed pipe for PLCI = 0x101
---
Bug or Feature ?
regards
Andreas
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Re: [Asterisk-Users] BGM Music
Some people like features and this is a feature that many systems have. I have had users ask for this specifically when installing other systems such as the NEC IPK and that system has this feature as well. - Original Message - From: Joseph [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 13, 2004 10:07 AM Subject: Re: [Asterisk-Users] BGM Music You make a good point. However in a corp environment where there are many users, one distraction is users playing music on there pc's. So the in this case corp has decided that one music can be provided via the phone. If the user wants something to listen to, that is available. Otherwise there is no music. So corp has also decided sound on pc's shall be disabled. Anyway, I just wanted to hear what the options were. On Thu, 2004-05-13 at 11:53, Tracy R Reed wrote: On Thu, May 13, 2004 at 10:57:30AM -0400, Joseph spake thusly: Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Why would you want to? The sound quality is horrible for music even on a good speakerphone. You've probably got a computer and decent speaks right there, why not just fire up xmms? -- respectfully, Joseph --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to play dialtone on channel xx (Zaptel TE405P)
Here are the relevant sections of my zaptel.conf and Zapata.conf files: ++ ZAPTEL.CONF span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs bchan=1-23 dchan=24 fxoks=25-96 ++ ZAPATA.CONF ; ; Zapata telephony interface ; ; Configuration file [channels] ;PRI trunk channels context = default language = en signalling = pri_cpe usecallerid = yes hidecallerid = no echocancel = yes echocancelwhenbridged = yes immediate = yes switchtype = 5ess pridialplan = national group = 1 channel = 1-23 ;T1-fxs (inside handsets) on the channel bank context = local language = english signalling = fxo_ks rxwink = 300 usecallerid = yes hidecallerid = no callwaiting = yes callwaitingcallerid = yes threewaycalling = yes transfer = yes cancallforward = yes callreturn = yes echocancel = yes echocancelwhenbridged = no immediate = no rxgain=0.0 txgain=0.0 channel = 25-96 ++ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Christian Hoffmeyer Sent: Thursday, May 13, 2004 11:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unable to play dialtone on channel xx (Zaptel TE405P) - Original Message - From: Alejandro Sosa To: [EMAIL PROTECTED] Sent: Thursday, May 13, 2004 10:15 AM Subject: [Asterisk-Users] Unable to play dialtone on channel xx (Zaptel TE405P) The problem I'm having is that when I pick up any of the analog handsets I get no dial tone. Switching works, I can place and receive calls on both analog and MGCP handsets. The MGCP phones give me dial tone fine when I take them off the hook, only the analog ones don't do it. Any ideas??? - Let's see the relavent sections of your zaptel.conf and zapata.conf files. Christian Hoffmeyer YottaDot Solutions Huntsville, AL (iax) 700.859.4508 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ERROR[147466]: chan_capi.c:1914
Anybody know what that mean ? ERROR[147466]: chan_capi.c:1914 capi_handle_msg: received a call waiting CONNECT_IND nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can asterisk be programmed to make alarm calls?
Those of you whom have a free Washington State phone number from ipkall.om will know that one has to use the number at least every 30 days or else the number becomes disconnected. I have 3 numbers pointed at my asterisk my which work very well but I still had the 30 day problem. Is there a way that I can program asterisk to make a call to my WA numbers so that they wont get disco'd? I'm thinking of something like a alrm call that one has in a hotel room. YOu pick up the phone and program a ring back time. Hope this make sense. Thanks G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can asterisk be programmed to make alarm calls?
On Thu, 2004-05-13 at 13:41, Mark Phillips wrote: Those of you whom have a free Washington State phone number from ipkall.om will know that one has to use the number at least every 30 days or else the number becomes disconnected. I have 3 numbers pointed at my asterisk my which work very well but I still had the 30 day problem. Is there a way that I can program asterisk to make a call to my WA numbers so that they wont get disco'd? I'm thinking of something like a alrm call that one has in a hotel room. YOu pick up the phone and program a ring back time. Hope this make sense. Research sample.call, write cron job to submit one every x days to guarantee you get a call out. I suggest, every 10 days, this gives you 3 opportunities to make the call in case your computer was down or unreachable for some reason during the month. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can asterisk be programmed to make alarm calls?
Sure you could even use the examples posted here and the wiki to use the outgoing spool to make calls. Just use a crontab to place a call file in the outgoing spool every x # of days and problem should be solved. On Thu, 2004-05-13 at 14:41, Mark Phillips wrote: Those of you whom have a free Washington State phone number from ipkall.om will know that one has to use the number at least every 30 days or else the number becomes disconnected. I have 3 numbers pointed at my asterisk my which work very well but I still had the 30 day problem. Is there a way that I can program asterisk to make a call to my WA numbers so that they wont get disco'd? I'm thinking of something like a alrm call that one has in a hotel room. YOu pick up the phone and program a ring back time. Hope this make sense. Thanks G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BGM Music
On 13-May-04, Tracy R Reed wrote: Why would you want to? The sound quality is horrible for music even on a good speakerphone. You've probably got a computer and decent speaks right there, why not just fire up xmms? perhaps he is doing installs / maintence / cabling office wide and he wants to listen to sports or music. so he goes from phone to fone and dials that extension instead of listening to airconditioning hum. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g.729 - licenses and opinions
Steven Critchfield [EMAIL PROTECTED] wrote: 17 years for software patents is FAR too long, IMO, but that's an entirely different story. IMO software patents shoudln't be for more than ~24 months since the industry moves so blazingly fast. I'm of mixed feelings here. I don't like software patents at all, but without them, some of the voice compression that is out there would possibly not have been developed. What would have been the incentive for the telecoms to allow the public in on some of the voice compressions with out getting paid for the work. The advantage should be obvious: The telecom companies need common standards so that equipment from competing suppliers can communicate with one another. Given an openly-usable standard, Voiceage would be free to attempt to sell their sub-standard software with full protection from copyright laws. Others would be equally free to implement an independent version that didn't rely upon IDE disks, channel limits and other nastiness. So while I think it is important, I also can't seem to draw a reasonable line. 24 months in most software isn't enough time from day 0 to make any reward for the work, at least not monetarily. What software project out there do you know had a major roll out sufficiently under 24 months from beginning of programming to have paid the programming staff off after say 1 year past the initial 24 months? Software patents encourage monopoly rather than freedom. Idiots write a line of code and then feel that they've invented something. Luckily, people who live in free countries, such as England, are not subject to such stupidity. We are free to write anything we like without having to hire a lawyer to check and double-check every line of code for patent infringements. If you're not careful, software development will turn into a legal minefield over there; Nobody will feel safe creating code in the USA and will have to turn to free countries, where software patents don't apply, to fill the demand for new software. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g.729 - licenses and opinions
On Thu, 2004-05-13 at 14:45, Kevin Walsh wrote: Steven Critchfield [EMAIL PROTECTED] wrote: So while I think it is important, I also can't seem to draw a reasonable line. 24 months in most software isn't enough time from day 0 to make any reward for the work, at least not monetarily. What software project out there do you know had a major roll out sufficiently under 24 months from beginning of programming to have paid the programming staff off after say 1 year past the initial 24 months? Software patents encourage monopoly rather than freedom. Idiots write a line of code and then feel that they've invented something. Temporary monopoly. Of course with the current time limits, it might as well be permanent since the techniques will be mostly useless by the time they are free. Luckily, people who live in free countries, such as England, are not subject to such stupidity. We are free to write anything we like without having to hire a lawyer to check and double-check every line of code for patent infringements. If you're not careful, software development will turn into a legal minefield over there; Nobody will feel safe creating code in the USA and will have to turn to free countries, where software patents don't apply, to fill the demand for new software. Actually I think it is going to be even worse than you stated. Having software developed in foreign countries will not make it any safer for us in the US to use the software. We will still be treading through that legal mine field. Of course, I think the problem here is that even if you roll back software patents we will have methodologies that can be implemented in software that are still patentable. Ohh well. Thanks for the depressing thread. On to threads that are on-topic for this list. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g.729 - licenses and opinions
BZZZT! Wrong too. Patents are a trade. The holder of the IP opens it up for public scrutiny and in return for exclusive control. Otherwise, companies would (and often do) keep the IP a trade secret. -brian Andrew Kohlsmith wrote: Just remember that you were given those patents as incentive to invent so that ultimately your work would go into the public domain so we can all enjoy it. We are buying your work with our tax dollars by protecting it for a short period of time so you have a little monetary incentive. BZZZT! Wrong. He was given those patents as in incentive to invent something that he could SELL without everyone on the planet copying his hard work and competing on his idea. Patents put the process out in the public so that it's easy to see when someone's infringing. 17 years for software patents is FAR too long, IMO, but that's an entirely different story. IMO software patents shoudln't be for more than ~24 months since the industry moves so blazingly fast. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-doc Conference Call - phase 2 :)
I would highly recommend recording it, even if you are attempting to transcribe it in real time. That way, you can always go back and replay a section you missed or want to clarify. As an added bonus, you could then make it available as an MP3 file, and those that could not partake can listen to it as though they were there. Jeremy -Original Message- From: Leif Madsen [mailto:[EMAIL PROTECTED] Sent: Thursday, May 13, 2004 11:42 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk-doc Conference Call - phase 2 :) Thank you to everyone who has offered so far! I've had formal offers from Martin List-Peterson, William Suffil, Greg Varga, Brian Cuthie and Ed Guy (hopefully I haven't forgotten someone!) Now we just have to decide where the best spot to host it is. What do you guys think? For this week, I don't care if this is a one off. At some point I'd like to have a weekly conference, and if we can get it hosted permanently once a week, that would be ideal. Right now, I'm more worried about just getting a conference going :) I'm going to make a GUESS that we are going to have between 10-15 people. Perhaps more? Maybe we can get a tally of who is expected to be there and then based on that we can decide on a location. The server should be both SIP and IAX accessible. Jared mentioned access via a 1-800 number or PSTN, but I'm not sure how practical, or necessary, that is. Again, your thoughts? Here is some things Jared dumped into the IRC channel the other day that we are going to try and focus on during the conference call: Layout Details (we can't go into too much detail about every possible soft phone/hard phone/voip provider) Goals (first good docs, then maybe get published) Submission process (mailing list? Website shows who's in charge of a certain section?) Focus (What do we want to focus on first? The intro and installation chapters?) Simplecity (Let's make sure a voip-newvie can get up and running, as long as they've used Linux before and know how to use a text editor.) Please feel free to add your suggestions. Tentatively the conference will be scheduled for Sunday evening North American time (I am EST, -0500 GMT). I'd like to try and get as much input as possible from people, but I realize we can't schedule around everyone. For now we will assume Sunday evening is good. In the future we can try a couple of different times so that it can be convenient for others. How are we going to record the thoughts of this conference? I'm a fairly fast typist, so I could attempt to record thoughts and idea's during the conference. Should we record it? At this point I'm going to open the floor to discussion! If you could reply via the asterisk-doc list, that would be best, this list already has too much traffic :) If you would like to contact me off list, feel free to email me, or get a hold of me on the #asterisk-doc IRC channel. Thanks in advance, Leif Madsen aka blitzrage ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g.729 - licenses and opinions
On Thu, May 13, 2004 at 02:58:47PM -0500, Steven Critchfield said: On Thu, 2004-05-13 at 14:45, Kevin Walsh wrote: Steven Critchfield [EMAIL PROTECTED] wrote: So while I think it is important, I also can't seem to draw a reasonable line. 24 months in most software isn't enough time from day 0 to make any reward for the work, at least not monetarily. What software project out there do you know had a major roll out sufficiently under 24 months from beginning of programming to have paid the programming staff off after say 1 year past the initial 24 months? Software patents encourage monopoly rather than freedom. Idiots write a line of code and then feel that they've invented something. Temporary monopoly. Of course with the current time limits, it might as well be permanent since the techniques will be mostly useless by the time they are free. And don't forget that with patents, it actually encourages splintering of technologies and hinders compatability. It happens all around us - GSM vs CDMA, GIF/PNG/JPEG, MPeg/OGG/WMA, etc. With software patents, the only benefit is to the patent holder. Users just get screwed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-doc Conference Call - phase 2 :)
Broadcast with app_ices to a shoutcast server For the world to listen too :P Has anyone gotten that app_ices to accually work? I sure as hell didn't. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jeremy Hall Sent: Thursday, May 13, 2004 3:29 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk-doc Conference Call - phase 2 :) I would highly recommend recording it, even if you are attempting to transcribe it in real time. That way, you can always go back and replay a section you missed or want to clarify. As an added bonus, you could then make it available as an MP3 file, and those that could not partake can listen to it as though they were there. Jeremy -Original Message- From: Leif Madsen [mailto:[EMAIL PROTECTED] Sent: Thursday, May 13, 2004 11:42 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk-doc Conference Call - phase 2 :) Thank you to everyone who has offered so far! I've had formal offers from Martin List-Peterson, William Suffil, Greg Varga, Brian Cuthie and Ed Guy (hopefully I haven't forgotten someone!) Now we just have to decide where the best spot to host it is. What do you guys think? For this week, I don't care if this is a one off. At some point I'd like to have a weekly conference, and if we can get it hosted permanently once a week, that would be ideal. Right now, I'm more worried about just getting a conference going :) I'm going to make a GUESS that we are going to have between 10-15 people. Perhaps more? Maybe we can get a tally of who is expected to be there and then based on that we can decide on a location. The server should be both SIP and IAX accessible. Jared mentioned access via a 1-800 number or PSTN, but I'm not sure how practical, or necessary, that is. Again, your thoughts? Here is some things Jared dumped into the IRC channel the other day that we are going to try and focus on during the conference call: Layout Details (we can't go into too much detail about every possible soft phone/hard phone/voip provider) Goals (first good docs, then maybe get published) Submission process (mailing list? Website shows who's in charge of a certain section?) Focus (What do we want to focus on first? The intro and installation chapters?) Simplecity (Let's make sure a voip-newvie can get up and running, as long as they've used Linux before and know how to use a text editor.) Please feel free to add your suggestions. Tentatively the conference will be scheduled for Sunday evening North American time (I am EST, -0500 GMT). I'd like to try and get as much input as possible from people, but I realize we can't schedule around everyone. For now we will assume Sunday evening is good. In the future we can try a couple of different times so that it can be convenient for others. How are we going to record the thoughts of this conference? I'm a fairly fast typist, so I could attempt to record thoughts and idea's during the conference. Should we record it? At this point I'm going to open the floor to discussion! If you could reply via the asterisk-doc list, that would be best, this list already has too much traffic :) If you would like to contact me off list, feel free to email me, or get a hold of me on the #asterisk-doc IRC channel. Thanks in advance, Leif Madsen aka blitzrage ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Freeworld
I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register = ##:[EMAIL PROTECTED] disallow=all allow=ulaw [iaxfwd] type=user context=fromiaxfwd deny=0.0.0.0/0.0.0.0 permit=65.39.205.0/255.255.255.0 Gotta be something easy im missing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Freeworld
Kyle == Kyle Hagan [EMAIL PROTECTED] writes: Kyle In coming works fine from FreeWorld via IAX. But when Kyle Dialing out i get [an error] ... Does iax2.fwdnet.com even support iax2=fwd? I thought it was just for registering an iax2 endpoint for fwd=iax2 calls. -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com/voip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g.729 - licenses and opinions
I think you patent haters are looking at the negative aspect only. Remember, that competition drives innovation. If everyone used the same product there would be no incentive to develop anything new or along the same lines, where's reward to innovate if there is no incentive, why do it? Incentive being the $$ for your work. This thread could go further into music, art, publications, pharmaceuticals, etc. I don't believe in monopolies, but it would lead to an intellectual monopoly thus a stagnant never changing technology. I know the concept will be hard to understand for some. Don't flame, just understand the other side. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed Sent: Thursday, May 13, 2004 4:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] g.729 - licenses and opinions On Thu, May 13, 2004 at 02:58:47PM -0500, Steven Critchfield said: On Thu, 2004-05-13 at 14:45, Kevin Walsh wrote: Steven Critchfield [EMAIL PROTECTED] wrote: So while I think it is important, I also can't seem to draw a reasonable line. 24 months in most software isn't enough time from day 0 to make any reward for the work, at least not monetarily. What software project out there do you know had a major roll out sufficiently under 24 months from beginning of programming to have paid the programming staff off after say 1 year past the initial 24 months? Software patents encourage monopoly rather than freedom. Idiots write a line of code and then feel that they've invented something. Temporary monopoly. Of course with the current time limits, it might as well be permanent since the techniques will be mostly useless by the time they are free. And don't forget that with patents, it actually encourages splintering of technologies and hinders compatability. It happens all around us - GSM vs CDMA, GIF/PNG/JPEG, MPeg/OGG/WMA, etc. With software patents, the only benefit is to the patent holder. Users just get screwed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-doc Conference Call - phase 2 :)
[EMAIL PROTECTED] wrote: Broadcast with app_ices to a shoutcast server For the world to listen too :P Has anyone gotten that app_ices to accually work? I sure as hell didn't. Yes, it works. Which part are you having problems with? Can you stream something with Icecast? Which config files do you want to see? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BGM Music
On 09:12 AM 5/13/2004, brian wrote: Every time I hear But legacy PBX's (do it like this|has this) it makes me wanna SCREAM. Asterisk is the new, the now, the hip... shed the old and bring on the new. editorial However, there are very few real green fields out there, and people have expectations. It really doesn't matter to the average employee using a telephone that you can do all these new and neat things. What matters is that they knew how to use the phone efficiently yesterday, and today with their 'new-fangled' phone system nothing works like it did. Many organizations who do business centered around the phone (sales, customer support, etc.), create procedures and policies around the way the phone works. So in that case, you either make asterisk work like the previous PBX system, or asterisk doesn't get installed. Such is real life. I was flying home from Sydney to San Fran a while back and I was sitting next to an vehicle engineer for one of the major US manufacturers. He worked in Australia and was telling me how widely CVT transmissions were in use in AU. He told me of all the huge benefits over manual or automatic geared systems, and I asked him why we didn't have CVT transmissions in the US. He said, because having the engine running at one specific RPM even while accelerating or decelerating disorientated drivers used to engine sounds correlated with acceleration/deceleration. I read an article just in the last few days comparing three new convertibles (mercedes, audi, and saab), and they really loved the audi, but one of the negatives they gave it was that it had a CVT and many drivers found that disturbing So from a commercial point of view, people who are working to install asterisk in place of legacy pbx or keysystems will always run into certain expectations, and so you will continue to see the messages on this list asking how to make asterisk do things like model XYZ PBX/key system does. /editorial Asterisk is about new and exciting ways of doing things and not some BACKWARDS old legacy PBX way of doing thing. * will let you be creative in ways PBX's in the past only dream they could. (and some PBX's in the now that cost in the THOUSANDS and have per port and per user license fees) The * has you, follow the white bunny! :P (yes there is no spoon) bkw Chris A. Icide 332 Valdez Ave. Half Moon Bay, CA 94019 650-712-8223 voice 212-400-1698 IP voice 650-712-8995 fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-doc Conference Call - phase 2 :)
Interesting idea :) Leif. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of brian Sent: Thursday, May 13, 2004 4:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk-doc Conference Call - phase 2 :) Broadcast with app_ices to a shoutcast server For the world to listen too :P Has anyone gotten that app_ices to accually work? I sure as hell didn't. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone iLBC to IAX2 iLBC
Dear Chris I have updated my budgetone phone to the 1.04.63 firmware and am trying to use the iLBC codec Where i can find this new firmware? Usualy i can download from http://www.grandstream.com/BETATEST/ but i only the stable version.. Thanks in advance Dimitri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy
Title: Message Not sure if this is the best place but does any one have any used IAXy's they are interested in selling? I am looking topick one up cheap for a proof of concept before going all out on them. Also does any one have any real life practical experience with how well (or not so well) that these devices have worked for them? you can reply to me off list at [EMAIL PROTECTED] Thanks Michael Blood
[Asterisk-Users] Re: Budgetone iLBC to IAX2 iLBC
In article [EMAIL PROTECTED], reseaux [EMAIL PROTECTED] wrote: Dear Chris I have updated my budgetone phone to the 1.04.63 firmware and am trying to use the iLBC codec Where i can find this new firmware? Usualy i can download from http://www.grandstream.com/BETATEST/ but i only the stable version.. http://www.voiptalk.org/products/gt_update.php Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: R2 support
Hi Steve, Are you going to make it available for the * community ? Thanks, -- []'s Raul M. Fragoso In theory, there is no difference between theory and practice. But, in practice, there is. Steve Underwood [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I make it compile. I just never finished it. :-) What I did to make R2 work properly was throw away the code that is now in CVS at Digium, and write a new implementation from scratch. I guess that is not quite the answer you wanted to hear. :-( Regards, Steve Jorge Verastegui wrote: Hi I know there is no support for R2, but I succcessfully compiled and ran the libr2. However, I am not able to initiate calls. The error I get is Couldn't call g3/71605538 -- Hungup 'Zap/32-1' == Everyone is busy at this time I understand that the idle signaling is not working right, any ideas on what I can do to fix this problem? Looking forward to your responses. Regards, Jorge On Fri, 2004-04-30 at 22:58, Steve Underwood wrote: jorge verastegui wrote: Hi i have successfully downloaded and compiled libr2 from source. But i dont seem to find how to properly configure it. When i run it (partcially unconfigured) the following error occurrs Signalling requested is R2 Signalling but line is in PRI Signalling signalling This error is easy to fix by changing ccs to cas, and removing crc4, in your zaptel.conf file. However libr2 does not work. It is a partly implemented solution which I abandoned. It only gets you about 10% of the way to a working R2 system :-( My current R2 software is completely different. Regards, Steve ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Budgetone iLBC to IAX2 iLBC
[EMAIL PROTECTED] wrote: Where i can find this new firmware? Usualy i can download from http://www.grandstream.com/BETATEST/ but i only the stable version.. Thanks in advance Dimitri http://tinyurl.com/23s6m ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] vocera
Someone on the list was asking about Vocera the other day http://searchmobilecomputing.techtarget.com/tip/0,289483,sid40_gci964088,00.html?track=NL-328ad=482142 Cheers, Dean
[Asterisk-Users] recommend a Linux based TFTP server
Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Freeworld
Steven E. Frazier wrote:
Kyle,
I am having issues outgoing, but I get a different problem, I get:
Connected to Asterisk CVS-HEAD-05/12/04-21:18:13 currently running on
asterisk (pid = 1696)
ectionsk*CLI
-- Starting simple switch on 'Zap/5-1'
-- Executing SetCallerID(Zap/5-1, Steven Frazier 299487 ) in
new stack
-- Executing Dial(Zap/5-1,
IAX2/299487:[EMAIL PROTECTED]/93578|60|r) in new stack
-- Called 299487:[EMAIL PROTECTED]/93578
May 13 17:13:56 WARNING[1142135600]: chan_iax2.c:5097 socket_read: Call
rejected by 65.39.205.121: Unable to negotiate codec
-- Hungup 'IAX2[65.39.205.121:4569]/1'
== No one is available to answer at this time
-- Executing Congestion(Zap/5-1, ) in new stack
== Spawn extension (toll-access, 693578, 3) exited non-zero on 'Zap/5-1'
-- Hungup 'Zap/5-1'
asterisk*CLI
So I have an issue with what codec it is using, my iax.conf file is:
Would you have any idea why I would be getting that and can't call out but I
can receive calls? Thanks.
Steve
[general]
port = 5036
disallow=all
allow=gsm
allow=ulaw
allow=alaw
;
;FWD Using IAXTEL - Testing
register=299487:[EMAIL PROTECTED]
;
;bindaddr=0.0.0.0
disallow=all
;allow=ilbc
allow=gsm
bandwidth=low
;jitterbuffer=yes
;tos=lowdelay
tos=reliability
jitterbuffer=yes
dropcount=3
maxjitterbuffer=500
maxexcessjitterbuffer=100
;
;
;FWD EXT 299487
[iaxfwd]
type=user
context=fromiaxfwd
deny=0.0.0.0/0.0.0.0
permit=65.39.205.0/255.255.255.0
allow=ulaw
According to fw's site they only use ulaw. So thats all I have enabled.
here is my iax.conf
[general]
port=5036
register = 410769:[EMAIL PROTECTED]
disallow=all
allow=ulaw
[iaxfwd]
type=user
context=fromiaxfwd
deny=0.0.0.0/0.0.0.0
permit=65.39.205.0/255.255.255.0
my extentions.conf has the following for IAX:
[home]
exten = _6.,1,SetCallerId,Kyle Hagan 410769
exten = _6.,2,Dial(IAX2/410769:[EMAIL PROTECTED]/${EXTEN:1},60,r)
exten = _6.,3,Congestion
[fromiaxfwd]
exten = 410769,1,Dial(sip/104,20,r)
exten = 410769,2,Voicemail,u104
exten = 410769,102,Voicemail,b104
with only the above I got freeworld to work.
I hope this helps.
You can call me at 410769 after you get it working to test it.
___
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[Asterisk-Users] MeetMe with AGI scripts
I've had a quick look through the mail list and wiki but haven't yet resorted to looking at the meetme source code.. I see references to a background agi script that can run if you're using Zap channels. Am I right in saying that that script runs for each channel in the conference? Or is it a one time deal, running when the conference is created? The backgrounder behind my question is that I have an IVR app where the caller will dial in to the system and interact. At some point I'll want them to zero-out to a call center operator, but once that conversation is finished, return them back to the IVR system to complete their transaction. I'm thinking I have to use a meetme conference to do this (as I want the original inbound IVR call to continue after the operator conversation)? A related question: What about letting 2 Zap channels talk online then continue with IVR, like in a chatline type application? I'm thinking I can't do this from within the AGI script spawned when the call arrives and have to use the manager interface to push calls around? Thanks in advance Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recommend a Linux based TFTP server
On Thu, May 13, 2004 at 10:43:58PM +0100, Robert Boardman wrote: Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? I've been using tftpd-hpa on my debian servers for Asterisk. Works' great! http://packages.debian.org/stable/net/tftpd-hpa -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recommend a Linux based TFTP server
Probably best to use what came with your distro. Steve Robert Boardman wrote: Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Robb, I wrote up a small tutorial on setting up the standard tftp server for linux check it out on my site. http://asterisk.titaniumsoft.net/ Mitchel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Robert Boardman Sent: Thursday, May 13, 2004 2:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] recommend a Linux based TFTP server Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb Do you Yahoo!?Yahoo! Movies - Buy advance tickets for 'Shrek 2'
Re: [Asterisk-Users] IAXy
Have you tried calling Digium sales? Andy *** REPLY SEPARATOR *** On 13/05/2004 at 15:24 [EMAIL PROTECTED] wrote: Not sure if this is the best place but does any one have any used IAXy's they are interested in selling? I am looking to pick one up cheap for a proof of concept before going all out on them. Also does any one have any real life practical experience with how well (or not so well) that these devices have worked for them? you can reply to me off list at [EMAIL PROTECTED] Thanks Michael Blood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXy
Title: Message If you just want a test unit, goto www.netxusa.com. Thats where digium sent me if I wasnt ordering in bulk. Brian D'Arcy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, May 13, 2004 2:24 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAXy Not sure if this is the best place but does any one have any used IAXy's they are interested in selling? I am looking topick one up cheap for a proof of concept before going all out on them. Also does any one have any real life practical experience with how well (or not so well) that these devices have worked for them? you can reply to me off list at [EMAIL PROTECTED] Thanks Michael Blood
Re: [Asterisk-Users] MeetMe with AGI scripts
On 13/05/2004 at 14:57 Paul Crick wrote: I've had a quick look through the mail list and wiki but haven't yet resorted to looking at the meetme source code.. I see references to a background agi script that can run if you're using Zap channels. Am I right in saying that that script runs for each channel in the conference? Or is it a one time deal, running when the conference is created? I should point out that you don;t actually have to be *using* a ZAP channel for the background agi to work. The script starts when the first person enters, once the conference is over it;s upto the script to realize this and exit (otherwise you'll end up with lots of processes laying about) The backgrounder behind my question is that I have an IVR app where the caller will dial in to the system and interact. At some point I'll want them to zero-out to a call center operator, but once that conversation is finished, return them back to the IVR system to complete their transaction. I'm thinking I have to use a meetme conference to do this (as I want the original inbound IVR call to continue after the operator conversation)? Ok, here's my quick thoughts on this. When the caller calls, put them into a conference with the background agi running. When they need to talk to an operator, get them to press 0 (for example). When they do this, generate a call file that rings an operator which when they answer puts them in the same conference. When the operator is finished they just hang up. Use MeetMeCount to determine if the operator has left A related question: What about letting 2 Zap channels talk online then continue with IVR, like in a chatline type application? I'm thinking I can't do this from within the AGI script spawned when the call arrives and have to use the manager interface to push calls around? I think you can apply the same principle outlined above for this.. HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Freeworld
James H. Cloos Jr. wrote: Does iax2.fwdnet.com even support iax2=fwd? I thought it was just for registering an iax2 endpoint for fwd=iax2 calls. It does, but I didn't want ulaw to be the default codec and I ended up setting up 2 entries for FWD as the inbound can't have a password so to set the outbound codec etc and supply a password it was all very messy :) -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible TICKING sound
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steven Critchfield wrote: | On Tue, 2004-05-11 at 17:33, Jason A. Pattie wrote: | | |Our problem ended up not being with Asterisk or Digium hardware. It was |the analog cordless phone. We simply have to live with it. What |happens is whenever a connection is established and the phone is |off-hook, an LED on the base lights up in a blink blink . blink |blink . etc. pattern. Everytime the LED lights, a pulse is sent to |the phone. It's especially bad when both lines are in use, as the phone |is a two-line capable device. Then you've got double the pulsing. | |This may have nothing to do with your problem. Just wanted to get it |out there in case anyone else runs into it, too. | | | Sounds like your phone needs either a aux power source to power that | led, or possible a little modification to clip that LED. | | I would make sure your cordless phone's power supply is within spec. If | it is, Maybe you might want to look into one of the other comments a | while back on the list about upping the power on the SLIC(?). You might | be able to provide enough power to the phone to not cause trouble when | it blinks the LED. Well, the phone is using the power supply that came in the box. :) - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFApAtuuYsUrHkpYtARAgkMAJ9X3lCwiqr6OKmjv4slBwbOqqOQvgCeO/rS Xq3C+YsY8pJ1gfmM4CqbDEQ= =fnTl -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pattern matching w/ Cisco dialplans
I don't know specifically about your question, however you can do a MATCH="*" for all matches that don't match anything (no pun intended). Mitchel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Sent: Thursday, May 13, 2004 4:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pattern matching w/ Cisco dialplans I have some Cisco 7940's running SIP image 6.3 and a newphone account. Reguarding my dialplan I'm having a small issue. I'd like to dial 9,2,xxx-xxx- for a LD Nufone calls - however I also need to dial local phone numbers ie 9,2xx- Currently my dialplan looks like so This DOES work - I can call LD using NuPhone and call local numbers that start w/ a 2 - however when I dial local numbers that start w/ a 2 I have to wait 10 seconds for the call to be initiated.. ie pressing 9xxx-, pause 10 seconds, initiate call. Looking over the SIPDefault.cnf I'm not finding a value that I can enter that would shorten this time. I'd like to have a pattern match in say 5 seconds as opposed to 10. Any ideas on how I can accomplish this? -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor[EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 x102 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Yahoo! Movies - Buy advance tickets for 'Shrek 2'
[Asterisk-Users] Consult transfer on SNOM 105
Does anyone have a consult-transfer working on SNOM? Using 2.04g we can't get it to work, Hold works, Calling the 3rd party works, but the transfer button does nothing. Playing with the REFER setting on the snom gives varying results on the Asterisk console... but no working transfer :( Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BGM Music
Call parking is kludgy when compared to the cool instant feedback provided by a Cisco Callmanager solution. This was the one thing I was talking about .. We did app_valetparking to act like cisco CCM in a way. I love it... I hate the current call parking its the most hacked together thing in asterisk IMHO, thats why I refuse to use it and use app_valetparking in its place. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MeetMe with AGI scripts
Quoting Andy Powell: I should point out that you don;t actually have to be *using* a ZAP channel for the background agi to work. The script starts when the first person enters, once the conference is over it;s upto the script to realize this and exit Ah.. right.. this makes sense, I *think*.. so is the AGI script able to interact with the caller? like respond to DTMFs and get files to play using stream file etc? Is it like the AGI is another person in the conference? If so, your comment about putting the caller in the conference immediately then generating a call file for the zero out to operator part makes sense. For the major part of the call, it's a conference with the caller and the AGI script interacting.. then we zero out, monitor participant count, and once that's reduced after the successful interaction with the operator, the AGI call flow continues? I'm not sure how that would work in a chat line type environment though? Person A is in conference 1, talking to the AGI.. Person B is in conference 2 doing the same thing.. they now need to talk to each other.. Would I use the manager interface to put person B in to person A's conference? Or just exec the meetme app with the other person's room number? Person B's AGI would need to do some monitoring to see if Person B hangs up? (or is it covered by the Zap channel handler?) What about DTMF detection in that case - I'd need to know who pressed the key to exit the live chat session.. Doable? Sorry for the million questions - I know what I want to do it, and how it would work in a Dialogic environment (having done this all already) but I'm still trying to get to grips with the way Asterisk works with resources, channels etc. . Thanks again Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
