[Asterisk-Users] sending MWI to a none local client
Hi, I am using * to function as the voice mail system for Vocal. Since I do not have a context in sip.conf file for each vocal client, I can't set the mailbox= in sip.conf. How do I get the MWI to a Vocal client ? Cheers Sathya ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] ISDN (isdn4linux) DDI
The only problems I have: + echo can be bad sometimes + DTMF on outbound, seems that the ISDN transmits it outofbound which means that if your receiving end if a PSTN line, your DTFM code does not do much. Anyone got a solution to this? Regards, Matthew Enger [EMAIL PROTECTED] On Wed, 2003-11-12 at 08:50, Michal Rybarik wrote: Hello Siggi, I have it working on distinguising just the local numbers of our 4 B channels and the number assigned to the group. I have ordered an '100 in-dial range' here in Australia and should have it available to me by the end of next week, I can let you know how it goes. SL Cool. That would make 100 virtual voice modems with I4L ;-) Only 2 (4, 6, ...) modems - depends on how much ISDN B channels you have - but with 100 dialing numbers. Every number can use any free channel, when it needs. Currently I plan only 1 or 2 BRI ISDN, so there will be possible to have 2 or 4 simultanous calls, but if I need, I can upgrade to ISDN PRI / E1 line with 30 channels and keep my numbers. SL I guess you'd rather use chan_capi if you're needing more than a handful SL of numbers! Is there any problem with it, while using I4L? I doesn't have any good experiences with CAPI under Linux. We tried AVM Fritz! ISDN card w/ CAPI under Linux two years ago, but without success. -- Michal Rybarik 21.sk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Enger [EMAIL PROTECTED] Xintegration ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Document Control System?
On Tue, 2003-11-11 at 12:28, Leif Madsen wrote: I'm sorry this is somewhat offtopic, but I do plan to use this to help me create documentation for the * project.. so I guess it is somewhat on topic :) Anyways, I am looking for some sort of document control system. It should act somewhat like a CVS where it keeps previous versions, allows people to submit documentation, keeps track of who has what document open etc.. etc.. The documentation also has to be written on a Windows desktop platform.. preferably would like to somehow use MS Word. Are you trolling here, or are you just clueless about the people who will be helping contribute to your documentation? I'm sure I am not the only one here that goes weeks on end without touching windows. Screw Word and its largely bloated file formats. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Jitter Buffer on chan_sip
mmmh... I'm not sure ig chan_sip has jitter buffer. I think that there isn't a jb in sip, but correct me if I'm wrong. Matteo. Il lun, 2003-11-10 alle 16:14, Andres ha scritto: Hi, I would like to test chan_sip with a bigger jitter buffer. Does anybody know where in the code this is defined? I looked through it but could not find where. If anybody else can find it please let me know. Regards, Andres ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 911 IAX(2): [EMAIL PROTECTED] - ext 911 Iaxtel: 1-700-56-62458 - ext 911 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Document Control System?
Steven Critchfield wrote: Are you trolling here, or are you just clueless about the people who will be helping contribute to your documentation? I'm sure I am not the only one here that goes weeks on end without touching windows. Screw Word and its largely bloated file formats. Hear hear!! I have one Windows system out of the several dozen I operate, and that is only because companies who know no differently make their config/maintenance programs only available on that platform. I spent about 3.5 hours tonight on Win98 trying to install a driver for a wireless card. It was everything I could do to keep from throwing the damn thing through the window (pun intended) as I realized that an incorrect driver installed 2.42 eons ago, long since removed from the system, still had its .inf file squirreled away somewhere, and so there ensued a seemingly-infinite series of change setup, reboot until I finally figured out where the mystery driver was coming from. I haven't had any use for Micro$oft since they stole MS-DOS from the people who developed DOS68 down in Texas, back in the early 80s (late 70s?). Their business since has consisted of nothing more than stealing the successful ideas of their competitors, then embracing and extending them into maddening complexity within their bloated, hyperexpensive, insecure, buggy excuse for an OS. Sorry folks. I'm easy to get set off on this topic. One of the reasons I love asterisk. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreeBSD
Not sure if it matters or not, but have you tried gmake? On Wed, 2003-11-12 at 19:29, Andrew Joakimsen wrote: I am trying to get Asterisk to compile on FreeBSD 4.8. Per bug 389, BSD support should be in CVS. I have also tried applying the patch in bug 374, but always get these messages: click# make Makefile, line 21: Missing dependency operator Makefile, line 23: Need an operator Makefile, line 72: Missing dependency operator Makefile, line 74: Need an operator Makefile, line 76: Need an operator Makefile, line 116: Missing dependency operator Makefile, line 118: Need an operator Makefile, line 119: Missing dependency operator Makefile, line 121: Need an operator Makefile, line 149: Missing dependency operator Makefile, line 151: Need an operator Makefile, line 152: Missing dependency operator Makefile, line 154: Need an operator Makefile, line 155: Missing dependency operator Makefile, line 157: Need an operator Makefile, line 158: Need an operator Makefile, line 159: Need an operator Makefile, line 161: Missing dependency operator Makefile, line 163: Need an operator Makefile, line 164: Missing dependency operator Makefile, line 165: Missing dependency operator Makefile, line 167: Need an operator Makefile, line 168: Need an operator Makefile, line 175: Missing dependency operator Makefile, line 179: Need an operator Makefile, line 182: Need an operator Makefile, line 213: Missing dependency operator Makefile, line 214: Could not find .depend Makefile, line 215: Need an operator Makefile, line 233: Missing dependency operator Makefile, line 236: Need an operator Makefile, line 239: Need an operator make: fatal errors encountered -- cannot continue Any advice? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Enger [EMAIL PROTECTED] Xintegration ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreeBSD
I haven't looked @ Frrebsd support, but possibly using gmake will fix the problem pfor you? On Wed, Nov 12, 2003 at 03:29:58AM -0500, Andrew Joakimsen wrote: I am trying to get Asterisk to compile on FreeBSD 4.8. Per bug 389, BSD support should be in CVS. I have also tried applying the patch in bug 374, but always get these messages: click# make Makefile, line 21: Missing dependency operator Makefile, line 23: Need an operator Makefile, line 72: Missing dependency operator Makefile, line 74: Need an operator Makefile, line 76: Need an operator Makefile, line 116: Missing dependency operator Makefile, line 118: Need an operator Makefile, line 119: Missing dependency operator Makefile, line 121: Need an operator Makefile, line 149: Missing dependency operator Makefile, line 151: Need an operator Makefile, line 152: Missing dependency operator Makefile, line 154: Need an operator Makefile, line 155: Missing dependency operator Makefile, line 157: Need an operator Makefile, line 158: Need an operator Makefile, line 159: Need an operator Makefile, line 161: Missing dependency operator Makefile, line 163: Need an operator Makefile, line 164: Missing dependency operator Makefile, line 165: Missing dependency operator Makefile, line 167: Need an operator Makefile, line 168: Need an operator Makefile, line 175: Missing dependency operator Makefile, line 179: Need an operator Makefile, line 182: Need an operator Makefile, line 213: Missing dependency operator Makefile, line 214: Could not find .depend Makefile, line 215: Need an operator Makefile, line 233: Missing dependency operator Makefile, line 236: Need an operator Makefile, line 239: Need an operator make: fatal errors encountered -- cannot continue Any advice? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MySQL Licence may be changing..
I read on a site yesterday (wish I had saved it now.) that said that MySQL were re-visiting their new licence policy to make it possible for projects to use MySQL again.. Has anyone else seen this? This looks like good news, it means that the MySQL stuff may be able to be merged back into the main Asterisk source so we will not have to hassle with the addons anymore.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pick up ringing exten
Is it possible with Asterisk to pick up ringing extension from other extension? So I do not have to run to other desk to pick up the phone. Sure, just add callgroup=2 pickupgroup=2 to each extension definition in sip.conf as an example. Dial *8 to pick up that ringing extn. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan_h323/g729 - X100P connecting to non-Digium Partner
If anybody still have any G729 handshake problem with Asterisk and other non-Digium partner, I *really* recommend to use this patch: http://bugs.digium.com/bug_view_page.php?bug_id=421 6 monhts passed and finally my problem seems to be solved. Thanks Adam! Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT : For the SQL gurus..
Thanks everyone for your help on this.. For those who are interested I have done some speed tests on these two queries (below) on my server and the results are.. Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s OUCH! those times are lng! Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Licence may be changing..
I guess people are pissed off with them and are looking at the alternatives. I think they are charging too much money for it. Also they must compete against MS free Personal Server (SQL Server but not optimized) and PostgreSQL. -- Original Message -- From: WipeOut [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Wed, 12 Nov 2003 08:58:11 + I read on a site yesterday (wish I had saved it now.) that said that MySQL were re-visiting their new licence policy to make it possible for projects to use MySQL again.. Has anyone else seen this? This looks like good news, it means that the MySQL stuff may be able to be merged back into the main Asterisk source so we will not have to hassle with the addons anymore.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Text entry by DTMF
A phone system providing this kind of directory service usually asks for the first three letters of the person's last name. So, if you wanted to call me and didn't know my extension, you'd press 6 for directory. Then you'll press 746 (for S H O - the beginning of my last name) And the software will direct you to my extension. Now if there was a Joe Pinkovski working here as well, then the software should ask you if you want to be connected to him or to Tom Shoval (preferably in alphabetical order). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen Sent: Tuesday, November 11, 2003 12:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Text entry by DTMF Eric Wieling wrote: You mean something like this: backup-fs-1*CLI show application directory backup-fs-1*CLI -= Info about application 'Directory' =- [Synopsis]: Provide directory of voicemail extensions [Description]: Directory(context): Presents the user with a directory of extensions from which they may select by name. The list of names and extensions is discovered from voicemail.conf. The context argument is required, and specifies the context in which to interpret the extensions. Returns 0 unless the user hangs up. It also sets up the channel on exit to enter the extension the user selected. Ummm.. kind of. I mean, it says Enter the first 3 digits of the persons last night and you enter them via the keypad, it then searches for the names, and says, Calling so-and-so. I think I've seen this feature on a phone system I called once, but I can't remember exactly how it worked. I'm pretty sure you just entered the persons last name in by digits. -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Unable to use voicemail(Please suggestion)
Try with another codec different than G.723. Use GSM o G.711 for this. You could disable G.723 in your sip.conf disallow=all allow=gsm allow=alaw allow=ulaw Hope this helps, Gus - Original Message - From: Hachy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 12:32 AM Subject: [Asterisk-Users] Re: Unable to use voicemail(Please suggestion) Hello all I got call log from Asterisk. I call to ext1001 from ext1002. But could not leave a message in the voice mail. Please help me. -- Executing Dial(SIP/1002-8217, SIP/1001|20) in new stack -- Called 1001 -- SIP/1001-25ce is ringing -- Nobody picked up in 2 ms == Spawn extension (sip, 1001, 2) exited non-zero on 'SIP/1002-8217 Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go down. And Icould not leave a messages. Please teach me how to resolve this problem. Here is configuration of Asterisk and Xlite. #sip.conf in Asterisk [general] port=5060 bindaddr=0.0.0.0 nortifymimetype=text/plain allow=all [1001] type=friend username=1001 secret=1001 host=dynamic defaultip=192.168.0.1 mailbox=1001 context=sip canreinvite=no [1002] type=friend username=1002 secret=1002 host=dynamic defaultip=192.168.0.1 mailbox=1002 context=sip canreinvite=no #extensions.conf in Asterisk [general] static=yes writeprotect=no [glovals] CONSOLE=Console/dsp [sip] exten = 1001,1,Dial(SIP/1001,20) exten = 1001,2,Voicemail(u1001) exten = 1001,102,Voicemail(b1001) exten = 1001,103,Hungup exten = 1002,1,Dial(SIP/1001,20) exten = 1002,2,Voicemail(u1002) exten = 1002,102,Voicemail(b1002) exten = 1002,103,Hungup #voicemail.conf in Asterisk [local] 1001 = 1001,1001,mail address 1002 = 1002,1002,mail address #Create mailbox by addmailbox already. #Client configuration User Name1001 1002 Authorization User same as username PAssword 1001 1002 Domain/Realm 192.168.0.120 SIP Proxy192.168.0.120 Here is call flow on this test. (c)2003 Xten Networks Inc. All rights reserved. Private build: 1008 SIP: 192.168.0.125:5061 RTP: 192.168.0.125:8000 NAT: 210.253.186.126 PXY#0: 192.168.0.120:5060 RECEIVE 192.168.0.120:5060 NOTIFY sip:[EMAIL PROTECTED]:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.120:5060;branch=z9hG4bK375605f3 From: asterisk sip:[EMAIL PROTECTED];tag=as633f7afa To: sip:[EMAIL PROTECTED]:5061 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: text/plain Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 SEND 192.168.0.120:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:5061 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 26502 INVITE Content-Type: application/sdp Content-Length: 301 v=0 o=1002 22002568 22002568 IN IP4 192.168.0.125 s=X-Lite c=IN IP4 192.168.0.125 t=0 0 m=audio 8000 RTP/AVP 4 0 8 3 101 a=rtpmap:4 G723/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:126 x-pro-encrypted/8000 RECEIVE 192.168.0.120:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.125:5061 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961 To: sip:[EMAIL PROTECTED];tag=as08d3281f Call-ID: [EMAIL PROTECTED] CSeq: 26502 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=05d14468 Content-Length: 0 SEND 192.168.0.120:5060 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:5061 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961 To: sip:[EMAIL PROTECTED];tag=as08d3281f Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 26502 ACK Max-Forwards: 70 Content-Length: 0 SEND 192.168.0.120:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:5061 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 26503 INVITE Proxy-Authorization: Digest username=1002,realm=asterisk,nonce= 05d14468,response=8fb4b56e7dae5665a8ea56a34027be5f,uri=sip:[EMAIL PROTECTED] 168.0.120 Content-Type: application/sdp Content-Length: 301 v=0 o=1002 22002778 22002778 IN IP4 192.168.0.125 s=X-Lite c=IN IP4 192.168.0.125 t=0 0 m=audio 8000 RTP/AVP 4 0 8 3 101
[Asterisk-Users] IAX needs a zaptel device?
Hi All, I'm currently running Asterisk with SIP phones and an ISDN card using chan_capi. I've just started to use IAX (GSM codec)over the Internet and the sound is adequate. However, there is an occasional 'glitch' in the audio resulting in lost sound or distortion. Is the distortion because I'm using zaprtc for timing instead of a zaptel card, or is more likely to be due to lost packets? I guess I'm asking if its worth getting a X100P for timing? -Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX needs a zaptel device?
Hi iax doesn't use zaptel for timing. only iax2 uses it, but when using trunking=yes. (not your case,so) so the distortion could be caused by loss of packets. Matteo. Il mer, 2003-11-12 alle 15:31, nathan ha scritto: Hi All, I'm currently running Asterisk with SIP phones and an ISDN card using chan_capi. I've just started to use IAX (GSM codec)over the Internet and the sound is adequate. However, there is an occasional 'glitch' in the audio resulting in lost sound or distortion. Is the distortion because I'm using zaprtc for timing instead of a zaptel card, or is more likely to be due to lost packets? I guess I'm asking if its worth getting a X100P for timing? -Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 911 IAX(2): [EMAIL PROTECTED] - ext 911 Iaxtel: 1-700-56-62458 - ext 911 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Jitter Buffer on chan_sip
it's implemented on the zap side (which is now configurable with jitterbuffers=foo in zapata.conf. Mark On Wed, 12 Nov 2003, Matteo Brancaleoni wrote: mmmh... I'm not sure ig chan_sip has jitter buffer. I think that there isn't a jb in sip, but correct me if I'm wrong. Matteo. Il lun, 2003-11-10 alle 16:14, Andres ha scritto: Hi, I would like to test chan_sip with a bigger jitter buffer. Does anybody know where in the code this is defined? I looked through it but could not find where. If anybody else can find it please let me know. Regards, Andres ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 911 IAX(2): [EMAIL PROTECTED] - ext 911 Iaxtel: 1-700-56-62458 - ext 911 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Licence may be changing..
I'll try to call them tonight. Mark On Wed, 12 Nov 2003, costas wrote: I guess people are pissed off with them and are looking at the alternatives. I think they are charging too much money for it. Also they must compete against MS free Personal Server (SQL Server but not optimized) and PostgreSQL. -- Original Message -- From: WipeOut [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Wed, 12 Nov 2003 08:58:11 + I read on a site yesterday (wish I had saved it now.) that said that MySQL were re-visiting their new licence policy to make it possible for projects to use MySQL again.. Has anyone else seen this? This looks like good news, it means that the MySQL stuff may be able to be merged back into the main Asterisk source so we will not have to hassle with the addons anymore.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAX 0.93 with some sound improvements and not only...
Hi all, DIAX 0.9.3 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. Still just IAX1 is supported (for the moment). What's new in 0.9.3? - accept blank passwords; - accept for registration/calls host names, not only IP Address; - password no more displayed in clear in the registration window; - if no username is entered in the registration form, then 'guest' is used by default; - tooltip for memory buttons as full CallerID; - missed calls indicator display now the number of missed calls (in the tooltip too); - tooltip for tray icon is now You have X missed call(s); - A bunch of signal processing stuff from speex (newly implemented in the Steve K's iaxclient library, see http://iaxclient.sourceforge.net for more details): 1) A denoising filter: This is a very effective filter which reduces background noise; 2) New AGC: This AGC implementation from speex seems to work much better than the compander implementation used previously; 3) A new automatic silence detection based on speex' VAD. If you set the silence threshold to a positive number, you now get speex' VAD, instead of my hacky noise-threshold code; 4) Bias removal: Removes bias from input signals. This makes other DSP functions work correctly, including the VU-meter, which was often incorrect when bias was present; 5) Echo cancellation:. Speex echo is used (this cannot be changed for the moment). - Some bugs solved: - after clicking on VOL UP/DOWN only the left channel is heard; - tab stop in Registration form corrected Please send me you feedback especially on sound quality (noise, agc, etc) and bugs. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip: 401 unauthorized with xlite
Romulo,
Without a little more information this is not so
easy to solve. So let me see if I can go through a couple of scenarios and
see if we can figure out your particular problem.
If you have X-Lite behind a NAT router AND you are
not connecting to an * server or your * server is outside of that same NAT
router then your configuration should look like this.
[2203]
type=friend
username=2203
auth=md5
secret=1234
host=dynamic
nat=yes
reinvite=no
canreinvite=no
dissallow=all
allow=gsm
context=sip
If you are connecting to a * box and it is behind the same NAT router ie on
the same network. You configuration is a workable configuration.
[2203]
type=friend
username=2203
auth=md5
secret=1234
host=192.168.10.149; my machine that has xlite
reinvite=no
canreinvite=no
dissallow=all
allow=gsm
context=sip
Now here is were it gets a little tricky. If you have your X-Lite and
* behind the same NAT router and you think you can make this work with
externip=???.???.???.??? then good luck. I was not able to get this to
work with any SIP configuration. I tried every possible combination I
could think of at that point with no luck at all. Unless someone else in
this group has been able to make this work then to the best of my knowledge it
will not work.
Now on a completely different side note here. I have done this where
I can get X-Lite or X-Pro either one to connect okay sometimes and not others
(totally random) with success. Sometimes it would auth ok and sometimes it
would fail. No configuration change or anything and nothing I could link
to the issue. During a SIP debug session I would simply get SIP/2.0 401
Unauthorized messages sent to my client. I even went so far as to change
the chan_sip.c to print out the two compared md5 hashes that should have looked
exactly the same and they were not. I would have reported this as a bug
but I could find no one else to have this same issue that I was having. So
my suggestion to you is this. If you think your sip configuration is
correct then temporarily remove your secret with ;secret=1234 or remove the line
altogether then restart your * box and then see if it will log in. If it
does then you know you are having some sort of other problem whether it be the
same authentication issue I was having or something else altogether.
-Robert
- Original Message -
From: doracknz foi
mais uma
To: [EMAIL PROTECTED]
Sent: Tuesday, November 11, 2003 7:09 PM
Subject: [Asterisk-Users] sip: 401 unauthorized with
xlite
Hi there,
I have tried very hard to setup the x-lite with
asterisk, but until now i didn't get sucess. When i start the asterisk in debug
mode, i see the message: sip/2.0 401 unauthorized. I know that
this problem with authentication. I put in my sip.conf as below.
[2203]
type=friend
username=2203
auth=md5
secret=1234
reinvite=no
canreinvite=no
dissallow=all
allow=gsm
context= sip
host= 192.168.10.149 - my machine that have xlite
extension.conf
[sip]
exten = 2203,1,Dial(${Phone1})
I have read and read many message in list but i could found anyone that
explain in details how to setup this correct. My sip.conf i got from a example
in pdf how to setup x-lite with sip, but this think doesn't work in my
server.
Please, could someone help me how to do that...
thanks a lot,
Romulo
Yahoo!
Mail - 6MB, anti-spam e antivírus gratuito. Crie sua
conta agora!
Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and not only...
On Wed, 2003-11-12 at 15:07, Dan wrote:
DIAX 0.9.3 is available for download from the same place:
Hi Dan :)
Do you know if anyone has successfully run DIAX on Linux with Wine?
After installing the VB6 runtime DLL, I ran diax.exe and got
fixme:ole:CoRegisterMessageFilter stub
fixme:ole:OLEPictureImpl_Construct Unsupported type 3
fixme:ole:OLEPictureImpl_SaveAsFile (0x404068d0)-(0x40406bc8, 0,
(nil)), hacked stub.
fixme:ole:VarParseNumFromStr (L2,flags=8000,), partial stub!
fixme:ole:VarParseNumFromStr numparse: cDig=30, InFlags=954
fixme:ole:VarNumFromParseNum (..,dwVtBits=20,), partial stub!
fixme:ole:VarParseNumFromStr (L-99,flags=8000,), partial stub!
fixme:ole:VarParseNumFromStr numparse: cDig=30, InFlags=954
fixme:ole:VarNumFromParseNum (..,dwVtBits=20,), partial stub!
fixme:ole:OLEPictureImpl_FindConnectionPoint tried to find connection
point on {33ad4ed2-6699-11cf-b70c-00aa0060d393}?
and then a 'Runtime Error '6': Overflow' dialog with 'OK' ..
I don't know if any of these messages are even remotely useful, but I've
included them for completeness :)
Cheers,
Gavin.
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Re: [Asterisk-Users] Jitter Buffer on chan_sip
On Wednesday 12 November 2003 09:47, Mark Spencer wrote: it's implemented on the zap side (which is now configurable with jitterbuffers=foo in zapata.conf. Will this work on a SIP to SIP call? What does the parameter jitterbuffers=XXX represent? Is it memory allocation or milliseconds of voice? Thanks, Andres Mark On Wed, 12 Nov 2003, Matteo Brancaleoni wrote: mmmh... I'm not sure ig chan_sip has jitter buffer. I think that there isn't a jb in sip, but correct me if I'm wrong. Matteo. Il lun, 2003-11-10 alle 16:14, Andres ha scritto: Hi, I would like to test chan_sip with a bigger jitter buffer. Does anybody know where in the code this is defined? I looked through it but could not find where. If anybody else can find it please let me know. Regards, Andres ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 911 IAX(2): [EMAIL PROTECTED] - ext 911 Iaxtel: 1-700-56-62458 - ext 911 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.93 with some sound improvements andnot only...
Hi,
- Original Message -
From: Gavin Hamill [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 12, 2003 5:23 PM
Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements andnot
only...
Do you know if anyone has successfully run DIAX on Linux with Wine?
I have no idea.
After installing the VB6 runtime DLL, I ran diax.exe and got
fixme:ole:CoRegisterMessageFilter stub
fixme:ole:OLEPictureImpl_Construct Unsupported type 3
fixme:ole:OLEPictureImpl_SaveAsFile (0x404068d0)-(0x40406bc8, 0,
(nil)), hacked stub.
fixme:ole:VarParseNumFromStr (L2,flags=8000,), partial stub!
fixme:ole:VarParseNumFromStr numparse: cDig=30, InFlags=954
fixme:ole:VarNumFromParseNum (..,dwVtBits=20,), partial stub!
fixme:ole:VarParseNumFromStr (L-99,flags=8000,), partial stub!
fixme:ole:VarParseNumFromStr numparse: cDig=30, InFlags=954
fixme:ole:VarNumFromParseNum (..,dwVtBits=20,), partial stub!
fixme:ole:OLEPictureImpl_FindConnectionPoint tried to find connection
point on {33ad4ed2-6699-11cf-b70c-00aa0060d393}?
and then a 'Runtime Error '6': Overflow' dialog with 'OK' ..
I don't know if any of these messages are even remotely useful, but I've
included them for completeness :)
OS?
Someone else with this issue?
Thanks,
Dan
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[Asterisk-Users] Media Negotiation Failed
Title: Mensaje Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) --- Asterisk (public ip: 64.76.xx.xx) -- Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and whensend SIP to 3600, I have this error: v=0o=root 20045 20045 IN IP4 64.76.xx.xx - asterisk ip addresss=sessionc=IN IP4 64.76.xx.xx - asterisk ip address.t=0 0m=audio 15372 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16(no NAT) to 64.76.xx.xx:5060 - 3600 ip addressSip read: LISIP/2.0 400 Bad Request - 'Media Negotiation Failed'Via: SIP/2.0/UDP 64.76.xx.xx:5060;branch=z9hG4bK31ba01da - asterisk ip addressFrom: "1143724956" sip:[EMAIL PROTECTED];tag=as33c45436 - * ip addressTo: sip:[EMAIL PROTECTED] -3600 ip addressCall-ID: [EMAIL PROTECTED]Warning: 304 64.76.xx.xx:0 "Media Type(s) Unavailable" - 3600 ip addressCSeq: 102 INVITE then I have too another GW 5300, with same IOS and same config.. and with it, all work OK!!!... I don't understand what is the problem!!... IT WORKS OK!!!.. Cisco 5300 (public ip. 64.76.xx.xx) --- Asterisk (public ip: 64.76.xx.xx) -- Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) Some clue?
Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and not only...
-- Original Message -- From: Dan [EMAIL PROTECTED] Hi all, DIAX 0.9.3 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro Thank you for the update! I have the following problems with it! When exiting the program we get a General Protech error. Also when calling Zap ports it keeps ringing. From DIAX to Sip it works fine! It actually sound better then before! But I can not call it from SIP get Audio missmatch. I can call it from normal Zap ports! Hope this helps! Keep up the work! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT : For the SQL gurus..
Thanks everyone for your help on this.. For those who are interested I have done some speed tests on these two queries (below) on my server and the results are.. Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s Tried fulltext indexing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Media Negotiation Failed
Title: Mensaje Fijate en los 'voice codecs' de los dial-peers. - Original Message - From: Sebastian Nocetti To: [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 12:41 PM Subject: [Asterisk-Users] Media Negotiation Failed Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) --- Asterisk (public ip: 64.76.xx.xx) -- Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and whensend SIP to 3600, I have this error: v=0o=root 20045 20045 IN IP4 64.76.xx.xx - asterisk ip addresss=sessionc=IN IP4 64.76.xx.xx - asterisk ip address.t=0 0m=audio 15372 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16(no NAT) to 64.76.xx.xx:5060 - 3600 ip addressSip read: LISIP/2.0 400 Bad Request - 'Media Negotiation Failed'Via: SIP/2.0/UDP 64.76.xx.xx:5060;branch=z9hG4bK31ba01da - asterisk ip addressFrom: "1143724956" sip:[EMAIL PROTECTED];tag=as33c45436 - * ip addressTo: sip:[EMAIL PROTECTED] -3600 ip addressCall-ID: [EMAIL PROTECTED]Warning: 304 64.76.xx.xx:0 "Media Type(s) Unavailable" - 3600 ip addressCSeq: 102 INVITE then I have too another GW 5300, with same IOS and same config.. and with it, all work OK!!!... I don't understand what is the problem!!... IT WORKS OK!!!.. Cisco 5300 (public ip. 64.76.xx.xx) --- Asterisk (public ip: 64.76.xx.xx) -- Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) Some clue?
Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and not only...
Hi, - Original Message - From: Ariel Batista [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 5:50 PM Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and not only... . Thank you for the update! I have the following problems with it! When exiting the program we get a General Protech error. This is a known bug (see the help file) Hope to be solved when the IAX2 version will be available Also when calling Zap ports it keeps ringing. Try to put a line in extensions.conf before the dial one xxx,1,Answer xxx,2,Dial( It actually sound better then before! The noise (the microphone one especially when used on a notebook) must be drastically reduced now. But I can not call it from SIP get Audio missmatch. What type of SIP phone?... I have test it with CIsco 7960 and it works as expected.. Where did you gtet this message (on SIP phone or on DIAX)? Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT : For the SQL gurus..
On Wednesday 12 November 2003 10:01, Roy Sigurd Karlsbakk wrote: Thanks everyone for your help on this.. For those who are interested I have done some speed tests on these two queries (below) on my server and the results are.. Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s Tried fulltext indexing? Fulltext indexing won't get you anything, considering that these queries aren't searching for non-0-based-offsets in substrings. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and not only...
Hi Gavin
i have the same error when i try to run DIAX with Wine.
thanks
Dimitri
On Wednesday 12 November 2003 15:23, Gavin Hamill wrote:
On Wed, 2003-11-12 at 15:07, Dan wrote:
DIAX 0.9.3 is available for download from the same place:
Hi Dan :)
Do you know if anyone has successfully run DIAX on Linux with Wine?
After installing the VB6 runtime DLL, I ran diax.exe and got
fixme:ole:CoRegisterMessageFilter stub
fixme:ole:OLEPictureImpl_Construct Unsupported type 3
fixme:ole:OLEPictureImpl_SaveAsFile (0x404068d0)-(0x40406bc8, 0,
(nil)), hacked stub.
fixme:ole:VarParseNumFromStr (L2,flags=8000,), partial stub!
fixme:ole:VarParseNumFromStr numparse: cDig=30, InFlags=954
fixme:ole:VarNumFromParseNum (..,dwVtBits=20,), partial stub!
fixme:ole:VarParseNumFromStr (L-99,flags=8000,), partial stub!
fixme:ole:VarParseNumFromStr numparse: cDig=30, InFlags=954
fixme:ole:VarNumFromParseNum (..,dwVtBits=20,), partial stub!
fixme:ole:OLEPictureImpl_FindConnectionPoint tried to find connection
point on {33ad4ed2-6699-11cf-b70c-00aa0060d393}?
and then a 'Runtime Error '6': Overflow' dialog with 'OK' ..
I don't know if any of these messages are even remotely useful, but I've
included them for completeness :)
Cheers,
Gavin.
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Re: [Asterisk-Users] OT : For the SQL gurus..
Andy Powell wrote: Thanks everyone for your help on this.. For those who are interested I have done some speed tests on these two queries (below) on my server and the results are.. Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s OUCH! those times are lng! Andy _ I agree the first three are long, but the last one works out to just over 26000 queries per min.. I didn't think that was bad for a PII 350.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TAPI development
Has anyone ever worked opn TAPI stuff to make asterisk work with it ? I'm a Windoze C++ developer dig'n into asterisk (and linux at the same time)since a few months and i'm quite interested in creating a TAPI driver for asterisk. so if anybody did any research in that way please inform me. Also i've you think it's quite impossible to do it we can discuss our idea's Michael Devenijn DKMA bvba
Re: [Asterisk-Users] OT : For the SQL gurus..
At 11:07 AM 11/10/2003, you wrote:
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 (code field not indexed) = 47.183s
Query1 (code field indexed) = 45.731s
Query2 (code field not indexed) = 109.321s
Query2 (code field indexed) = 2.302s
Query2 has additional overhead in the script as well because it has to
itterate through the number and build up the query..
Query1 is far simpler to use in a script becasue the query does not have
to be built up..
Since you only need to do a simple lookup, why not either (a) build your
own db or (b) use berkely DB or some other fast database engine? Since all
you really need to do is a prefix search on a key:
struct node {
char num;
struct node* p0;
struct node* p1;
struct node* p2;
struct node* p3;
struct node* p4;
struct node* p5;
struct node* p6;
struct node* p7;
struct node* p8;
struct node* p9;
char* desc;
}
That's 48 bytes per record (not counting the description). Memory usage
will depend on how much data you need to store, but lookups would be O(k),
where k is the length of the key.
--Ernest
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[Asterisk-Users] Dial Plan Sequencing
I have an interesting dilemma with sequencing in the dialplan. Up to
now, I have assumed that the extensions in the dial plan were tested in
the order that they appear in extensions.conf. In other words, I have
the following fragment which was designed to dial toll free on the PSTN
and all other long distance on VoIP:
[longdistance]
include = local
;Handle local, etc first. (or so
I thought!)
exten = _91NXXNXX,1,Dial(${VPLSTRUNK}/${EXTEN:1});Dial long
distance through VoiP
exten = _91NXXNXX,2,Congestion
;OOPS! No lines available?
:
:
[local]
:
exten = _91800NXX,1,Dial(${PSTNTRUNK}/${EXTEN}) ; Long distance
toll free accessed through PSTN trunk interface
exten = _91800NXX,2,Congestion
exten = _91888NXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten = _91888NXX,2,Congestion
exten = _91877NXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten = _91877NXX,2,Congestion
exten = _91866NXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten = _91866NXX,2,Congestion
; The rest of the local definitions, etc
:
I expected that the _918 definitions would be tested first, followed
by the _91N definitions. Unfortunately, it appears as if the
definitions made using the include= operator are always tested last.
This means that the toll free numbers dialed by people in the
longdistance context are always routed over VoIP rather than PSTN
because they match the _91N pattern. While I can fix this with a
complicated set of conditionals or dial string patterns, I wonder if
anyone has found a more elegant solution, remembering that I want to
give some extensions access to only the local context, but still provide
toll free service for everyone (i.e, I don't want to move the _918
definitions into the longdistance context).
Stephen R. Besch
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Re: [Asterisk-Users] FreeBSD
It looks like my conversion of the STUN server to a GNU Autotools build system will go into Vovida.org's CVS system soon. My next task will be to do the same for Asterisk. Third task is to get Asterisk to use STUN. Back to BSD: I think GNU Autotools is the right way to fix this. But until then, Yes, you should try and make your BSD system as GNU-like as you can. Heck, I've got a GNU/Solaris system right here that I'm writting this with. Just put /usr/local/... first in your path and install the standard utilities there But as I said the long term fix is to use automake/autoconf to build custom Makefiles for your system. One more thing. Please people, if you post an error message that reads in part ...at line number NNN why don't you quote line number NNN? I assume you are posting in order to get help. Why not make it easy for people to help you. At least quote the line that is giving the error --- [EMAIL PROTECTED] wrote: I haven't looked @ Frrebsd support, but possibly using gmake will fix the problem pfor you? On Wed, Nov 12, 2003 at 03:29:58AM -0500, Andrew Joakimsen wrote: I am trying to get Asterisk to compile on FreeBSD 4.8. Per bug 389, BSD support should be in CVS. I have also tried applying the patch in bug 374, but always get these messages: click# make Makefile, line 21: Missing dependency operator Makefile, line 23: Need an operator Makefile, line 72: Missing dependency operator Makefile, line 74: Need an operator Makefile, line 76: Need an operator Makefile, line 116: Missing dependency operator Makefile, line 118: Need an operator Makefile, line 119: Missing dependency operator Makefile, line 121: Need an operator Makefile, line 149: Missing dependency operator Makefile, line 151: Need an operator Makefile, line 152: Missing dependency operator Makefile, line 154: Need an operator Makefile, line 155: Missing dependency operator Makefile, line 157: Need an operator Makefile, line 158: Need an operator Makefile, line 159: Need an operator Makefile, line 161: Missing dependency operator Makefile, line 163: Need an operator Makefile, line 164: Missing dependency operator Makefile, line 165: Missing dependency operator Makefile, line 167: Need an operator Makefile, line 168: Need an operator Makefile, line 175: Missing dependency operator Makefile, line 179: Need an operator Makefile, line 182: Need an operator Makefile, line 213: Missing dependency operator Makefile, line 214: Could not find .depend Makefile, line 215: Need an operator Makefile, line 233: Missing dependency operator Makefile, line 236: Need an operator Makefile, line 239: Need an operator make: fatal errors encountered -- cannot continue Any advice? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Licence may be changing..
--- Mark Spencer [EMAIL PROTECTED] wrote: I'll try to call them tonight. Mark On Wed, 12 Nov 2003, costas wrote: I guess people are pissed off with them and are looking at the alternatives. I think they are charging too much money for it. Also they must compete against MS free Personal Server (SQL Server but not optimized) and PostgreSQL. I think the best thing would be to keep DBMS-specific code out of the default Asterisk install. There are several other very good free SQL DBMSes. One of them is actually supported by one of the world's largest software companies, SAP. SAP and MySQL signed an agreement where MySQL will co-market SAPDB and the name will change toi MaxDB. MaxDB is be marketed as a step up from MySQL to an enterprize class DBMS. It will be interresting to see how the MySQL people will define MySQL, they surely will not try and tell people it is enterprize class. Perhaps they will be truthfull now and sell it as a light--weight product. Claiming integration with SAP would e a great marketing feature for Asterisk. everyone uses SAP. http://www.sapdb.org/7.4/sapdb_mysql.htm http://www.sapdb.org/ I use PostgreSQL and Oricle but have been following SAPDB for some time. -- Original Message -- From: WipeOut [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Wed, 12 Nov 2003 08:58:11 + I read on a site yesterday (wish I had saved it now.) that said that MySQL were re-visiting their new licence policy to make it possible for projects to use MySQL again.. Has anyone else seen this? This looks like good news, it means that the MySQL stuff may be able to be merged back into the main Asterisk source so we will not have to hassle with the addons anymore.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P random hangups.
I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogresssettingthem to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-workinclude = extensionssignalling = fxs_ksgroup = 1callgroup = 1pickupgroup = 1usecallerid = yescallerid = asreceivedhidecallerid = nocallwaiting = yescallwaitingcallerid = yesthreewaycalling = yestransfer = yesechocancel = yesechocancelwhenbridged = yesrxgain = 0.6txgain = 0.6immediate = nobusydetect = yescallprogress = yesmusiconhold = randomchannel = 1 context = inbound-homeinclude = extensionssignalling = fxs_ksgroup = 1callgroup = 1pickupgroup = 1usecallerid = yescallerid = asreceivedhidecallerid = nocallwaiting = yescallwaitingcallerid = yesthreewaycalling = yestransfer = yesechocancel = yesechocancelwhenbridged = yesrxgain = 0.6txgain = 0.6immediate = nobusydetect = yescallprogress = yesmusiconhold = randomchannel = 2 -Robert
[Asterisk-Users] Echo sometimes with TDM40B / X100P only
During calls using an extension off of the TDM40B out througha X100P I sometimes get a echo or cave sound if you will. It is random sometimes I have it sometimes not. Sometimes it starts with the beginning of a call sometimes you can be in the middle of a call and it starts. It only happens on the extension off of the TDM40B the caller on the X100P (outside line) does not get the same echo. My zapata.conf looks like. [channels]language = en ; TDM40B context = trusted signalling = fxo_ksgroup = 2channel = 3-6 ; X100P #1 context = inbound-1include = extensionssignalling = fxs_ksgroup = 1callgroup = 1pickupgroup = 1usecallerid = yescallerid = asreceivedhidecallerid = nocallwaiting = yescallwaitingcallerid = yesthreewaycalling = yestransfer = yesechocancel = yesechocancelwhenbridged = yesrxgain = 0.6txgain = 0.6immediate = nobusydetect = yescallprogress = yesmusiconhold = randomchannel = 1 # X100P #2 context = inbound-2include = extensionssignalling = fxs_ksgroup = 1callgroup = 1pickupgroup = 1usecallerid = yescallerid = asreceivedhidecallerid = nocallwaiting = yescallwaitingcallerid = yesthreewaycalling = yestransfer = yesechocancel = yesechocancelwhenbridged = yesrxgain = 0.6txgain = 0.6immediate = nobusydetect = yescallprogress = yesmusiconhold = randomchannel = 2
[Asterisk-Users] RE: Media Negotiation Failed
Codecs are g711ulaw, on both Cisco5300... Dial Peer config is showed below Los codecs que uso son G711ulaq, en los dos Cisco5300, te muestro los dialpeers... GW that not work - GW que no funciona translation-rule 1017 Rule 0 8002666333 1000 dial-peer voice 1016 voip destination-pattern 8002666333 translate-outgoing called 1017 session protocol sipv2 session target ipv4:64.76.xx.xx --- IP DE ASTERISK. dtmf-relay h245-alphanumeric codec g711ulaw no vad GW that work - GW que funciona translation-rule 7 Rule 0 ^3104 1000 Rule 1 ^3105 1000 dial-peer voice 7 voip destination-pattern 310[4-5] translate-outgoing called 7 session protocol sipv2 session target ipv4:64.76.xx.xx IP DE ASTERISK. dtmf-relay h245-alphanumeric codec g711ulaw no vad -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Miércoles, 12 de Noviembre de 2003 02:12 p.m. Para: [EMAIL PROTECTED] Asunto: Asterisk-Users digest, Vol 1 #1869 - 11 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: DIAX 0.93 with some sound improvements and not only... (Ariel Batista) 2. Re: OT : For the SQL gurus.. (Roy Sigurd Karlsbakk) 3. Re: Media Negotiation Failed (CW_ASN - Gus) 4. Re: DIAX 0.93 with some sound improvements and not only... (Dan) 5. Re: OT : For the SQL gurus.. (Tilghman Lesher) 6. Re: DIAX 0.93 with some sound improvements and not only... (reseaux) 7. Re: OT : For the SQL gurus.. (WipeOut) 8. Re: OT : For the SQL gurus.. (WipeOut) 9. TAPI development (Michael Devenijn) 10. Re: OT : For the SQL gurus.. (Ernest W. Lessenger) 11. Dial Plan Sequencing (Stephen R. Besch) --__--__-- Message: 1 Date: Wed, 12 Nov 2003 10:50:05 -0500 From: Ariel Batista [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and not only... Reply-To: [EMAIL PROTECTED] -- Original Message -- From: Dan [EMAIL PROTECTED] Hi all, DIAX 0.9.3 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro Thank you for the update! I have the following problems with it! When exiting the program we get a General Protech error. Also when calling Zap ports it keeps ringing. From DIAX to Sip it works fine! It actually sound better then before! But I can not call it from SIP get Audio missmatch. I can call it from normal Zap ports! Hope this helps! Keep up the work! --__--__-- Message: 2 Date: Wed, 12 Nov 2003 17:01:10 +0100 (CET) From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT : For the SQL gurus.. Reply-To: [EMAIL PROTECTED] Thanks everyone for your help on this.. For those who are interested I have done some speed tests on these two queries (below) on my server and the results are.. Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s Tried fulltext indexing? --__--__-- Message: 3 From: CW_ASN - Gus [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Media Negotiation Failed Date: Wed, 12 Nov 2003 13:01:29 -0300 Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_0061_01C3A91D.16D9E3F0 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable MensajeFijate en los 'voice codecs' de los dial-peers. - Original Message -=20 From: Sebastian Nocetti=20 To: [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 12:41 PM Subject: [Asterisk-Users] Media Negotiation Failed Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) --- Asterisk (public ip: = 64.76.xx.xx) -- Cisco 3600 (public ip: 64.76.xx.xx , same network than = * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, = asterisk plays a welcome message and resend call to Cisco 3600 that have = 4 analog lines connected... but after cisco play welcome message and = when send SIP to 3600, I have this error: v=3D0 o=3Droot 20045 20045 IN IP4 64.76.xx.xx - asterisk ip address s=3Dsession c=3DIN IP4 64.76.xx.xx - asterisk ip address. t=3D0 0 m=3Daudio 15372 RTP/AVP 0 101 a=3Drtpmap:0 PCMU/8000 a=3Drtpmap:101 telephone-event/8000 a=3Dfmtp:101 0-16 (no NAT) to 64.76.xx.xx:5060 - 3600 ip address Sip read: LI SIP/2.0 400 Bad Request -
RE: [Asterisk-Users] OT : For the SQL gurus..
Hey, -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of WipeOut Sent: Wednesday, November 12, 2003 10:28 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT : For the SQL gurus.. Andy Powell wrote: Thanks everyone for your help on this.. For those who are interested I have done some speed tests on these two queries (below) on my server and the results are.. Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s OUCH! those times are lng! Andy _ I agree the first three are long, but the last one works out to just over 26000 queries per min.. I didn't think that was bad for a PII 350.. :) Later.. I disagree with your disagreement :P We have to keep in mind the big picture. We are providing dial tone. I don't want to have to wait an extra 2.302 seconds for my call to be set up. Also, think of the big organizations: if you have 200 phone calls, and you have each one take even a couple of seconds extra, you are going to have to add more lines... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial Plan Sequencing
Hello. I have never run into this problem. What I would do is inserted
below:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen R. Besch
Sent: Wednesday, November 12, 2003 11:06 AM
To: asterisk users list
Subject: [Asterisk-Users] Dial Plan Sequencing
I have an interesting dilemma with sequencing in the dialplan. Up to
now, I have assumed that the extensions in the dial plan were tested
in
the order that they appear in extensions.conf. In other words, I have
the following fragment which was designed to dial toll free on the
PSTN
and all other long distance on VoIP:
[longdistance]
; include = local
;Handle local, etc first. (or
so
I thought!)
exten = _91NXXNXX,1,Dial(${VPLSTRUNK}/${EXTEN:1});Dial
long
distance through VoiP
exten = _91NXXNXX,2,Congestion
;OOPS! No lines available?
:
:
[reallongdistance]
include = local
include = longdistance
[local]
:
exten = _91800NXX,1,Dial(${PSTNTRUNK}/${EXTEN}) ; Long
distance
toll free accessed through PSTN trunk interface
exten = _91800NXX,2,Congestion
exten = _91888NXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten = _91888NXX,2,Congestion
exten = _91877NXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten = _91877NXX,2,Congestion
exten = _91866NXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten = _91866NXX,2,Congestion
; The rest of the local definitions, etc
:
I expected that the _918 definitions would be tested first, followed
by the _91N definitions. Unfortunately, it appears as if the
definitions made using the include= operator are always tested last.
This means that the toll free numbers dialed by people in the
longdistance context are always routed over VoIP rather than PSTN
because they match the _91N pattern. While I can fix this with a
complicated set of conditionals or dial string patterns, I wonder if
anyone has found a more elegant solution, remembering that I want to
give some extensions access to only the local context, but still
provide
toll free service for everyone (i.e, I don't want to move the _918
definitions into the longdistance context).
Stephen R. Besch
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Whadda ya think?
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Re: [Asterisk-Users] OT : For the SQL gurus..
Hi David, For those who are interested I have done some speed tests on these Test script of 1000 quieries.. Query2 (code field indexed) = 2.302s OUCH! those times are lng! I agree the first three are long, but the last one works out to just over 26000 queries per min.. I didn't think that was bad for a PII [..] picture. We are providing dial tone. I don't want to have to wait an extra 2.302 seconds for my call to be set up. Also, think of the big organizations: if you have 200 phone calls, and you have each one take even a couple of seconds extra, you are going to have to add more lines... you might want to re-read the results: 1000 queries = 2.302s For me this looks like 2ms per query. Maybe WipeOut can confirm the information (one way or another) -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hello Gavin,
Sorry for so long time in my reply but I was very busy on other tasks.
I attached to this message my working test files for mgcp.
Best regards,
Daniel
Daniel ANDRE wrote:
Gavin Hamill a crit:
On Tue, 2003-11-04 at 10:14, Daniel ANDRE wrote:
Hullo Daniel :)
Can I request that you post the pertinent parts of your config to the
list, since I'm sure I'm not the only one who would benefit from a set
of known-working configs for these phones.
I will make some clean-up in my files and post them in a day or two. I
am not fully satisfied with my conf for now but it may help you.
Daniel
Personally, I'm on the verge of buying some SwissVoice handsets, simply
because the mix of feature-set, price, and build quality seems to be
untouchable.
The GrandStreams are about the same price, but the build quality looks
cheap and plastic - the IP10 actually looks like a business telephone.
Cheers,
Gavin.
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--
Daniel ANDRE (mailto:[EMAIL PROTECTED])
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com
--
Daniel ANDRE (mailto:[EMAIL PROTECTED])
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if stati=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=yes
[globals]
dan = sip/p-dan.phone.iris-tech.fr
swiss1 = mgcp/aaln/[EMAIL PROTECTED]
swiss2 = mgcp/aaln/[EMAIL PROTECTED]
;
;MACRO
;
[macro-apl1]
exten = s,1,Dial(${ARG1},30,Ttmr)
;#
[SIP]
;#
include = ent
[local]
include = ent
;
[default]
include = ent
[ent]
exten = 111,1,Macro(apl1,${swiss1})
exten = 112,1,Macro(apl1,${swiss2})
exten = 326,1,Macro(apl1,${dan})
;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 192.168.10.254
[192.168.10.11]
host = 192.168.10.10
nat = no
disallow = all
allow = g711
allow = alaw
line = aaln/1
canreinvite = yes
[192.168.10.10]
threewaycalling=yes
transfer=yes
callwaiting=yes
callwaitingcallerid=yes
context=local
host = 192.168.10.10
nat=no
callerid = John 92
line = aaln/1
callgroup=0
cancallforward=yes
transfer=yes
line = aaln/1
Re: [Asterisk-Users] OT : For the SQL gurus..
Thanks everyone for your help on this.. For those who are interested I have done some speed tests on these two queries (below) on my server and the results are.. Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s OUCH! those times are lng! Can I have a copy of this database? It'd be cool to see what can be done to tune it :) roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT : For the SQL gurus..
David Gomillion wrote: Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s I disagree with your disagreement :P We have to keep in mind the big picture. We are providing dial tone. I don't want to have to wait an extra 2.302 seconds for my call to be set up. Also, think of the big organizations: if you have 200 phone calls, and you have each one take even a couple of seconds extra, you are going to have to add more lines... David, Please read again.. it was 2.3s for 1000 queries being run in a loop one after the other.. that means about 0.0023s for 1.. so you would get your dialtone pretty quick!!.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Document Control System?
Steven Critchfield wrote: Are you trolling here, or are you just clueless about the people who will be helping contribute to your documentation? I'm sure I am not the only one here that goes weeks on end without touching windows. Screw Word and its largely bloated file formats. Unfortunately, we do live in a world where some of us are forced to use Windows. I was not trying to imply that everyone must use Word (or even myself), but simply was meaning that I need something that will ALSO run on Windows. Whether this is a PHP based web interface, using RTF documents with CVS (I don't remember mentioning I was totally opposed to CVS) or whatever the case may be. Sometimes I think you are a very bitter man, just for the sake of being bitter :) -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1-700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Document Control System?
The new OpenOffice works very well now and is completley cross platform. It also allows one to save in any of a serval file formats. I've been using it to produce HTML, PDF and plain text format copies of documentation. and I can run this same Open Office suite on Solaris, Linux and Windows. --- Leif Madsen [EMAIL PROTECTED] wrote: Steven Critchfield wrote: Are you trolling here, or are you just clueless about the people who will be helping contribute to your documentation? I'm sure I am not the only one here that goes weeks on end without touching windows. Screw Word and its largely bloated file formats. Unfortunately, we do live in a world where some of us are forced to use Windows. I was not trying to imply that everyone must use Word (or even myself), but simply was meaning that I need something that will ALSO run on Windows. Whether this is a PHP based web interface, using RTF documents with CVS (I don't remember mentioning I was totally opposed to CVS) or whatever the case may be. Sometimes I think you are a very bitter man, just for the sake of being bitter :) -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1-700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zultys.
Title: Zultys. Is anyone familiar with http://www.zultys.com/index.htm. Do they use Asterisk? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
[Asterisk-Users] D Channel Bonding
Title: D Channel Bonding Are the Digium T1/E1 cards capable of D channel bonding for PRI? As in one D channel can service two more PRI lines? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
[Asterisk-Users] Canadian VoIP termination?
Hi, Does anyone know of Canadian VoIP termination providers? I have Canadian customers and would like to provide Canadian dial in and dial out (canadian callerid). Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan Sequencing
David Gomillion wrote:
Hello. I have never run into this problem. What I would do is inserted
below:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen R. Besch
Sent: Wednesday, November 12, 2003 11:06 AM
To: asterisk users list
Subject: [Asterisk-Users] Dial Plan Sequencing
I have an interesting dilemma with sequencing in the dialplan. Up to
now, I have assumed that the extensions in the dial plan were tested
in
the order that they appear in extensions.conf. In other words, I have
the following fragment which was designed to dial toll free on the
PSTN
and all other long distance on VoIP:
[longdistance]
; include = local
;Handle local, etc first. (or
so
I thought!)
exten = _91NXXNXX,1,Dial(${VPLSTRUNK}/${EXTEN:1});Dial
long
distance through VoiP
exten = _91NXXNXX,2,Congestion
;OOPS! No lines available?
:
:
[reallongdistance]
include = local
include = longdistance
[local]
:
exten = _91800NXX,1,Dial(${PSTNTRUNK}/${EXTEN}) ; Long
distance
toll free accessed through PSTN trunk interface
exten = _91800NXX,2,Congestion
exten = _91888NXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten = _91888NXX,2,Congestion
exten = _91877NXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten = _91877NXX,2,Congestion
exten = _91866NXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten = _91866NXX,2,Congestion
; The rest of the local definitions, etc
:
I expected that the _918 definitions would be tested first, followed
by the _91N definitions. Unfortunately, it appears as if the
definitions made using the include= operator are always tested last.
This means that the toll free numbers dialed by people in the
longdistance context are always routed over VoIP rather than PSTN
because they match the _91N pattern. While I can fix this with a
complicated set of conditionals or dial string patterns, I wonder if
anyone has found a more elegant solution, remembering that I want to
give some extensions access to only the local context, but still
provide
toll free service for everyone (i.e, I don't want to move the _918
definitions into the longdistance context).
Stephen R. Besch
___
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Whadda ya think?
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Worked like a charm! Thanks
Stephen R. Besch
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Re: [Asterisk-Users] OT : For the SQL gurus - performance testing
Chris Albertson wrote: Testing a querry by doing 2000 identical querries and then deviding the total by 2000 is not a valid way to measure the time to do one querry. The result will appear to be as much as 100X or even more to fast. The reason is: 1) Operating system will have cached the exact disk sectors required resulting in zero disk access time to 1999 of the querries. 2) If not the above the DBMS will have it's own cache 3) Some DBMSes will cache the querry plan so even the internal time to process the SQL into a list of lower level actions will be reduced for 1999 of the querries. A more realistic test would use multiple processes to 1) Do normal server stuff to keep the OS-level caches flushed 2) Do background writes to the DBMS, say loggig CDR data at a realistic rate 3) The test program that does 2000 __random__ test querries with a small, realistic delay between each. Next you'd devide the total time by 2000 and then subtract the one or two second delay you introduced out. I'll bet a beer the result is longer than 2ms. An even more realistic test would run four or five copies of step #3 above concurently. In the real world with MySQL the biggest constraint of performance and scaleability is due to table locking and a very simple test will ignore this single largest factor. I agree with you entirely.. If I was trying to get a real world TPC measurement.. The comparison was to compare the speed of one query against another.. They were both run in the same way on the same hardware so both would have gained or lost based on the same factors.. I really wasn't concerned with the actual total transaction time it at this point, that will come later and will involve a number of queries to complete the operation, it was more to determine which single query completed faster.. but thanks for the advice on a testing procedure it will be usedful later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] D Channel Bonding
On Wed, 12 Nov 2003, Ray Burkholder wrote: Are the Digium T1/E1 cards capable of D channel bonding for PRI? As in one D channel can service two more PRI lines? NFAS? Not that I know of. -- 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
[Asterisk-Users] SoftFax question
Hi, I am looking at using the softfax that Steve Underwood has developed. It's very straight forward when you assign an extension for the fax. A function that several pbx's has is that they listen for the 'faxtone' for 5 seconds after 'answer' in the menu where you can enter your local extension number, it's normally done in parallel with the DTMF detection. I think that the logical solution would be if the DTMF mask given to the DTMFdetector could had a digit for fax or if there was a 'background' function that we could check on with IfFaxGoTo(xxx). I haven't been able to google any function in '*' that would help us with this so that's why I try the list in case I (hopefully) have overlooked something. The above function would be nice since you could share the same access number for phone and fax (like the old autofax switches). Secondly when people mistakenly queues a fax for you main access number it would just be dropped into the 'faxbox' instaedt of calling you 10 times over the next 20 minutes. Freddi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Document Control System?
Chris Albertson wrote: The new OpenOffice works very well now and is completley cross platform. It also allows one to save in any of a serval file formats. I've been using it to produce HTML, PDF and plain text format copies of documentation. and I can run this same Open Office suite on Solaris, Linux and Windows. I, unfortunately live in a mostly WIN environment, but I have also migrated to OpenOffice from MS WinWord for several reasons. First, it does a better job handling graphics (most of the time), second, it's just as flexible and easier to use for formatted documents, and third, it's not MS. Here's a fun experiment for those of you whose favorite document preparation tool is still MS WinWord. Take any Word format document, preferably one that is over 100K in size. Load it into OpenOffice. The file translation is not perfect, but it's pretty good and getting better. Now save the file in OpenOffice format. Finally look at the comparative file sizes. If your experience is anything like mine, I suspect that you will be surprised. Oh, by the way, did I mention that it is free!! Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TAPI development
Actually, you may want to make your TSP use the Manager interface. Not ALL of the TAPI primitives are supported, but such is the case with most PBXs. Better yet, you can alter the PBX source to add additional events/commands that can be written into your TSP. If you need the TSP/MSP (or TAPI/WAV) type of functionality, you will need to look at wrapping AIX, SIP, MGCP, or H323 into an MSP or WAV device. AIX would most likely be the easiest. You would want to support multiple 3rd party pseudo-UAs on a single PC (i.e. multiple virtual channels). A closed-source PBX call the IP Office offered by Avaya uses TAPI 2.x/3.x to provide 1st and 3rd party control as well as media interfaces. I am under NDA so I can't go into detail. Suffice it to say that what you want to do can in fact be done, given enough effort. I have a client that is interested in a TSAPI wrapper for Asterisk. If somebody has JTAPI we could have a large portion of the CTI universe covered. Regards, Steven -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florian Overkamp Sent: Wednesday, November 12, 2003 1:24 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TAPI development Hi, Citeren Michael Devenijn [EMAIL PROTECTED]: Has anyone ever worked opn TAPI stuff to make asterisk work with it ? I'm a Windoze C++ developer dig'n into asterisk (and linux at the same time) since a few months and i'm quite interested in creating a TAPI driver for asterisk. so if anybody did any research in that way please inform me. Also i've you think it's quite impossible to do it we can discuss our idea's I guess it would be possible, but it more or less depends on what you'd want to do with it. Technically you should be able to make a virtual CAPI interface that links to an asterisk (IAX) account or perhaps simply some XMLRPC server (as in, VoIP calling from the PC or instructing asterisk to connect the phone on my desk to a certain number, straight out of outlook or similar) Although I have briefly looked through TAPI I am not familiar enough with its capacities. Any kind of useable interface for CTI with asterisk would serve a great purpose, but would - I expect - be very customer-specific... -- Best regards, Florian Overkamp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT : For the SQL gurus..
That's an awful lot of assumptions, my friend. What we care about is how long it takes to get the FIRST result, not all of them together. I mean, I only need to call ONE number, not 1000... This comes to a statement of optimal. What is optimal? Optimal with respect to what??? We want something that adds as close to 0 to the time to connect a call. If your database is not in memory at the time (swapped out, whatever) then there is overhead that you have to consider. No, it's not technically part of the database's time, but that is inconsequential. That's time a user will have to wait. And then what happens when reads are non-sequential? Is the database going to be local, or over a network? How long does a timeout take if the DB goes down? How are you going to deal with rebuilding indexes or getting backed up (i.e. lookups are a bit slower when the DBMS is working on other stuff)? I guess the point is this: if you're not careful in your implementation, this feature could be very dangerous. It's a fundamental issue with using a database with a real-time system. Web traffic hitting a database is great because it doesn't matter if a response takes 2 ms or 200 ms. And people are used to having to click refresh if it doesn't pop up in about 30 seconds. People expect more from voice. Having said that, I would like to see a database schema to define the dialplan, voicemail, features, etc. Then, it would be nice for * to read the info it needs and put it in a more efficient data structure, perhaps on a timed basis (i.e. every hour or so). This would give us the niceness of a PHP interface for configuration while giving us the quick response we currently enjoy when making phone calls. It would also allow the creation of one master dial plan that could then be spread across servers (more) easily, moved around, a server replaced, etc. And we could give Suzzy Secretary the rights to add new extensions or reset voicemail passwords for one subset of extensions, but not delete extensions. As a first step, I am working to engineer a way to intelligently define a database and create a script to write all of the little .conf files. I have no code yet; it's still cooking in the old brain. It has to be well thought out for it to be really useful. I'm writing a design document to formalize what each piece is going to do. Thoughts? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of WipeOut Sent: Wednesday, November 12, 2003 12:25 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT : For the SQL gurus.. David Gomillion wrote: Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s I disagree with your disagreement :P We have to keep in mind the big picture. We are providing dial tone. I don't want to have to wait an extra 2.302 seconds for my call to be set up. Also, think of the big organizations: if you have 200 phone calls, and you have each one take even a couple of seconds extra, you are going to have to add more lines... David, Please read again.. it was 2.3s for 1000 queries being run in a loop one after the other.. that means about 0.0023s for 1.. so you would get your dialtone pretty quick!!.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Global configuration question
Hi there, I'm new to Asterisk. Installed, configured, but not really used it yet... I'm considering some investment on mounting a small network for voice phones, say 20 to 30 terminals. What hardware should I use for the telephones ? IP Phones seam too expensive and I'm sure they do a lot of things that are not needed in Asterisk context... Are there other hardware solutions? What about USB phones connected to personal computers? Anything out there? Thanks to all for this briliant product -- Sergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftFax question
On Wed, 12 Nov 2003, Freddi Hansen wrote: Hi, I am looking at using the softfax that Steve Underwood has developed. It's very straight forward when you assign an extension for the fax. A function that several pbx's has is that they listen for the 'faxtone' for 5 seconds after 'answer' in the menu where you can enter your local extension number, it's normally done in parallel with the DTMF detection. I think that snip You want a fax extension: exten=fax,1,Blah() A google for 'fax extension' turns up the announcement of this feature here: http://lists.digium.com/pipermail/asterisk-users/2002-October/005414.html ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftFax question
Quoting Freddi Hansen [EMAIL PROTECTED]: I haven't been able to google any function in '*' that would help us with this so that's why I try the list in case I (hopefully) have overlooked something. Just take a look at fax extenstion which basically does what you want. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial Plan Sequencing
Here is what I have which uses IAXTEL for 800 calling and VOIP for long
distance with a fall back to my PSTN line. I don't have any issues as far
as 1800 numbers being grabbed before the long distance numbers. My internal
context is for all extensions inside the house, whereas the fax doesn't use
VOIP at all. Don't know if this will help or not:
[longdistance]
exten = _1NX,1,Dial(${VP_CONN}/[EMAIL PROTECTED])
exten = _1NX,2,Dial(${OUTGOING}/${EXTEN})
exten = _1NX,3,Congestion
[longdistance-novoip]
exten = _1NX,1,Dial(${OUTGOING}/${EXTEN})
exten = _1NX,2,Congestion
[tollfree]
exten = _1800NXX,1,Dial(${IAXTEL}/[EMAIL PROTECTED])
exten = _1800NXX,2,Dial(${OUTGOING}/${EXTEN})
exten = _1888NXX,1,Dial(${IAXTEL}/[EMAIL PROTECTED])
exten = _1888NXX,2,Dial(${OUTGOING}/${EXTEN})
exten = _1877NXX,1,Dial(${IAXTEL}/[EMAIL PROTECTED])
exten = _1877NXX,2,Dial(${OUTGOING}/${EXTEN})
exten = _1866NXX,1,Dial(${IAXTEL}/[EMAIL PROTECTED])
exten = _1866NXX,2,Dial(${OUTGOING}/${EXTEN})
[tollfree-iax]
exten = _91800NXX,1,Dial(${IAXTEL}/${EXTEN:[EMAIL PROTECTED])
exten = _91888NXX,1,Dial(${IAXTEL}/${EXTEN:[EMAIL PROTECTED])
exten = _91877NXX,1,Dial(${IAXTEL}/${EXTEN:[EMAIL PROTECTED])
exten = _91866NXX,1,Dial(${IAXTEL}/${EXTEN:[EMAIL PROTECTED])
[local]
include = emergency
include = operator
include = info
exten = _703NXX,1,Dial(${OUTGOING}/${EXTEN})
exten = _202NXX,1,Dial(${OUTGOING}/${EXTEN})
exten = _301NXX,1,Dial(${OUTGOING}/${EXTEN})
exten = _571NXX,1,Dial(${OUTGOING}/${EXTEN})
[emergency]
exten = 911,1,Dial(${OUTGOING}/${EXTEN})
[operator]
exten = 0,1,Dial(${OUTGOING}/${EXTEN})
[info]
exten = 411,1,Dial(${OUTGOING}/${EXTEN})
[fax]
include = house
include = local
include = longdistance-novoip
include = international-novoip
[internal]
include = house
include = local
include = iaxtel
include = fwd-out
include = iconnect
include = tollfree
include = longdistance
include = international
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[Asterisk-Users] Sipura / Handytone / Cisco
Could anybody shed some light in which device they would use in this situation: Remote office PBX's to be connected via a) Cisco ATA-186 or b) Sipura SPA-2000 or c) Grandstream HT-ATA-286 to go via the net to an * box. Pros / Cons for each device would be appreciated! Thanks Kris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Soft fax (rxfax) 8 byte output problem resolved?
I have read all the mailing list posts regarding rxfax receiving a fax and outputing an 8 byte tif file (tif header only). This is the problem I can't seem to get past. Has anyone out there also had this problem and found some workaround for it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan Sequencing
That is just beautiful...
Would you mind if it got into the wiki, or onto a webpage here or there?
-
Andrew Thompson
- Original Message -
From: Carlton J. O'Riley [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 12, 2003 3:48 PM
Subject: RE: [Asterisk-Users] Dial Plan Sequencing
Here is what I have which uses IAXTEL for 800 calling and VOIP for long
distance with a fall back to my PSTN line. I don't have any issues as far
as 1800 numbers being grabbed before the long distance numbers. My internal
context is for all extensions inside the house, whereas the fax doesn't use
VOIP at all. Don't know if this will help or not:
[longdistance]
exten = _1NX,1,Dial(${VP_CONN}/[EMAIL PROTECTED])
exten = _1NX,2,Dial(${OUTGOING}/${EXTEN})
exten = _1NX,3,Congestion
[longdistance-novoip]
exten = _1NX,1,Dial(${OUTGOING}/${EXTEN})
exten = _1NX,2,Congestion
[tollfree]
exten = _1800NXX,1,Dial(${IAXTEL}/[EMAIL PROTECTED])
exten = _1800NXX,2,Dial(${OUTGOING}/${EXTEN})
exten = _1888NXX,1,Dial(${IAXTEL}/[EMAIL PROTECTED])
exten = _1888NXX,2,Dial(${OUTGOING}/${EXTEN})
exten = _1877NXX,1,Dial(${IAXTEL}/[EMAIL PROTECTED])
exten = _1877NXX,2,Dial(${OUTGOING}/${EXTEN})
exten = _1866NXX,1,Dial(${IAXTEL}/[EMAIL PROTECTED])
exten = _1866NXX,2,Dial(${OUTGOING}/${EXTEN})
[tollfree-iax]
exten = _91800NXX,1,Dial(${IAXTEL}/${EXTEN:[EMAIL PROTECTED])
exten = _91888NXX,1,Dial(${IAXTEL}/${EXTEN:[EMAIL PROTECTED])
exten = _91877NXX,1,Dial(${IAXTEL}/${EXTEN:[EMAIL PROTECTED])
exten = _91866NXX,1,Dial(${IAXTEL}/${EXTEN:[EMAIL PROTECTED])
[local]
include = emergency
include = operator
include = info
exten = _703NXX,1,Dial(${OUTGOING}/${EXTEN})
exten = _202NXX,1,Dial(${OUTGOING}/${EXTEN})
exten = _301NXX,1,Dial(${OUTGOING}/${EXTEN})
exten = _571NXX,1,Dial(${OUTGOING}/${EXTEN})
[emergency]
exten = 911,1,Dial(${OUTGOING}/${EXTEN})
[operator]
exten = 0,1,Dial(${OUTGOING}/${EXTEN})
[info]
exten = 411,1,Dial(${OUTGOING}/${EXTEN})
[fax]
include = house
include = local
include = longdistance-novoip
include = international-novoip
[internal]
include = house
include = local
include = iaxtel
include = fwd-out
include = iconnect
include = tollfree
include = longdistance
include = international
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[Asterisk-Users] Group dial codes ?(Newbie question)
Hi All, Using asterisk and extension.conf can i make a group dial code ? Like this. Ie. Let's say i have a group called directors. Only People in this group can dial to a external number like 800. How can i make this possible in asterisk ? Thanks alot ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Group dial codes ?(Newbie question)
Sure. Make two contexts like these: [peons] exten = 1,1,TakeOutTheTrash() exten = 2,1,WashTheDishes() exten = 3,1,CleanTheToilet() [rulers] include = peons exten = 800,1,FeedMeGrapes() This way the rulers have everything the peons have and then a little more. Then use your sip.conf or your zapata.conf to assign some phones to the ruler context and others to the peon context. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Carlos Arnt Sent: Wednesday, November 12, 2003 2:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Group dial codes ?(Newbie question) Hi All, Using asterisk and extension.conf can i make a group dial code ? Like this. Ie. Let's say i have a group called directors. Only People in this group can dial to a external number like 800. How can i make this possible in asterisk ? Thanks alot ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 488 not acceptable here message
Title: 488 not acceptable here message I'm creating a test environment for Asterisk. I have Asterisk running on a PC with only a NIC card, No FXO, FXS, TDM cards. I have two Cisco 7960 phones setup for SIP. Within Asterisk, the SIP SHOW PEERS, shows the phones. They don't appear under SIP SHOW REGISTRY. When I call phone 2 from phone 1, I get a message stating it is from Phone 2, stating, Got SIP Response 488 Not Acceptable Here back from 167.131.14.26. Can someone point me in the direction to look for the trouble?
[Asterisk-Users] menu prompts and voice mail greetings.
What program do you use to record menu prompts and voice mail greetings we tried windows recorder and it kept telling us bad file format. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Global configuration question
Not to incite a flame war, but the only phone I like is the Cisco 7960. If you look hard enough you can find them off-lease for around $235 with a power brick. Cisco just came out with some new firmware that adds features previously sorely missed (like call forwarding). Once you get used to being able to search the company directory, get weather information, and pull up stock quotes all from the handset you will never want to go back. This assumes you know how to have a little fun with xml but fun it is. Take my advice cautiously though because the only other IP phone I have experience with is the Grandstream BudgetTone which I regret ever purchasing. If you really want to go cheap, buy a $200 channel bank on ebay and put in analog phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sérgio Bernardo Sent: Wednesday, November 12, 2003 1:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Global configuration question Hi there, I'm new to Asterisk. Installed, configured, but not really used it yet... I'm considering some investment on mounting a small network for voice phones, say 20 to 30 terminals. What hardware should I use for the telephones ? IP Phones seam too expensive and I'm sure they do a lot of things that are not needed in Asterisk context... Are there other hardware solutions? What about USB phones connected to personal computers? Anything out there? Thanks to all for this briliant product -- Sergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] menu prompts and voice mail greetings.
I prefer to record them right over the phone. Set up a macro like this
[macro-recordsound]
;${ARG1} - Sound filename
exten = s,1,record(${ARG1}:gsm,3)
exten = s,2,playback(${ARG1})
exten = s,3,playback(vm-goodbye)
exten = s,4,hangup
and then in your main context do something like this
;Temporary recording options
exten = 150,1,Macro(recordsound,main-menu-announcement)
Good luck
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Larry D.
Black
Sent: Wednesday, November 12, 2003 2:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] menu prompts and voice mail greetings.
What program do you use to record menu prompts and voice mail greetings
we tried windows recorder and it kept telling us bad file format.
Thanks.
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[Asterisk-Users] Zap timeout not occurring
Good day, I am trying to setup an outbound dial plan which will time out if no answer. Using a X100P with the following dial command : exten = 101,3,Dial(Zap/1/3036972357,5); try the desk line - fail to step 104 It dials out successfully, but never times out. I have a basic Zapata config : group = 1context = RedRockWeblanguage = ensignalling = fxs_ksusecallerid = yeshidecallerid = noechocancel = yesechocancelwhenbridged = noimmediate = nochannel = 1-2 Suggestions? Thanks! Tom [EMAIL PROTECTED]www.tellink-corp.com303-697-2357303-697-3103 fax
RE: [Asterisk-Users] Canadian VoIP termination?
By the end of next week, we'll be able to offer IAX2 service for Vancouver, Toronto, Hamilton, Montreal. End of this month or so: Calgary, Edmonton, Ottawa, Winnipeg. Sometime in December: Windsor, Kitchener and London. By mid next week, Charlotte NC should be on line. Other centers, as listed at: http://voice.oneunified.net/coverageareas.html will be available as needed. All with local inbound/outbound with DID service. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dana Martens Sent: November 12, 2003 14:41 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Canadian VoIP termination? Hi, Does anyone know of Canadian VoIP termination providers? I have Canadian customers and would like to provide Canadian dial in and dial out (canadian callerid). Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Brazilian VOIP Terminator
I am looking for a brazilian VOIP terminator for the states of Sao Paulo, Parana, Para and Bahia. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 strange error
Hi! I have this test configuration: Cisco7940(SIP)-*-GnuGK(H.323)-ATA186(H.323) When I do call from ATA to 7940, everything is OK (exept volume level, but it is not seriously). But when I try to call from 7940 to ATA, I got a strange error: =*= In CreateRealTimeLogicalChannel for call 21165 -- externalIpAddress: 10.253.1.253 -- externalPort: 15462 -- SessionID: 1 -- Direction: IsTransmitter -- Sending SETUP message -- Received RELEASE COMPLETE message... -- Sending RELEASE COMPLETE -- Call to ip$10.253.1.254:1720 aborted, insufficient bandwidth. == H.323 Connection deleted. What is insufficient bandwidth? There is none about bandwidth in source code of H.323 modules :-( Of course, in GnuGK bandwidth control is turned off and everything is going on on almost free LAN. -- WBR, Max Tulyev (MT6561-RIPE, 2:463/[EMAIL PROTECTED]) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] menu prompts and voice mail greetings.
Did you record the messages as gsm format? - Original Message - From: Larry D. Black [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 6:33 PM Subject: [Asterisk-Users] menu prompts and voice mail greetings. What program do you use to record menu prompts and voice mail greetings we tried windows recorder and it kept telling us bad file format. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap timeout not occurring
On Wednesday, November 12, 2003 3:47 PM, Tom Weeks [SMTP:[EMAIL PROTECTED] wrote: Good day, I am trying to setup an outbound dial plan which will time out if no answer. Using a X100P with the following dial command : exten = 101,3,Dial(Zap/1/3036972357,5) ; try the desk line - fail to step 104 That is right, it does not time out and never will correctly since the X100P 'seizes' the line from the phone company immediately upon dialing so the X100P does not know whether the far end phone is ringing, is busy or if someone has answered. I believe it would take getting some type of digital interface (T1 or ISDN) in order to have far end answer supervision. Don Pobanz It dials out successfully, but never times out. I have a basic Zapata config : group = 1 context = RedRockWeb language = en signalling = fxs_ks usecallerid = yes hidecallerid = no echocancel = yes echocancelwhenbridged = no immediate = no channel = 1-2 Suggestions? Thanks! Tom [EMAIL PROTECTED] www.tellink-corp.com 303-697-2357 303-697-3103 fax File: ATT00015.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Licence may be changing..
On Wed, Nov 12, 2003 at 09:22:56AM -0800, Chris Albertson wrote: ... There are several other very good free SQL DBMSes. One of them is actually supported by one of the world's largest software companies, SAP. SAP and MySQL signed an agreement where MySQL will co-market SAPDB and the name will change toi MaxDB. MaxDB is be marketed as a step up from MySQL to an enterprize class DBMS. It will be interresting to see how the MySQL people will define MySQL, they surely will not try and tell people it is enterprize class. Perhaps they will be truthfull now and sell it as a light--weight product. At the Portland Open Source Conference, the MySQL guys basically called the SAPdb crap. They were going to take some of the good aspects of it an put it into MySQL. They said the SAPdb was not extensible in the form it is now. Of course, they never actually called it crap that is my summary of the conversation that I heard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Licence may be changing..
--- PJ Welsh [EMAIL PROTECTED] wrote: On Wed, Nov 12, 2003 at 09:22:56AM -0800, Chris Albertson wrote: ... There are several other very good free SQL DBMSes. One of them is actually supported by one of the world's largest software companies, SAP. SAP and MySQL signed an agreement where MySQL will co-market SAPDB and the name will change toi MaxDB. MaxDB is be marketed as a step up from MySQL to an enterprize class DBMS. It will be interresting to see how the MySQL people will define MySQL, they surely will not try and tell people it is enterprize class. Perhaps they will be truthfull now and sell it as a light--weight product. At the Portland Open Source Conference, the MySQL guys basically called the SAPdb crap. They were going to take some of the good aspects of it an put it into MySQL. They said the SAPdb was not extensible in the form it is now. Of course, they never actually called it crap that is my summary of the conversation that I heard. I don't know much about SAPDB except that if you buy an SAP system and don't have a preferred DBMS, like Oracle or something you get SAPDB But I do know, first hand, that if you need extensiblity then PostgreSQL is the way to go. It's designed like Asterisk in that users can add their own funtions and extend the SQL with new data types and operators. But I didn't intend to start a DBMS war. Just wanted to point out that IMO it's best to a favor one over another. Best to remain independent. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Licence may be changing..
Prehaps a novel thought but what about ODBC for asterisk? Isn't that the whole idea of standards and such, stop adding support for every db and just have odbc? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Unable to use voicemail(Thanks)
Thank you Gus I found some mistake in extension.conf exten = 1001,2.Voicemail(u1001) This must be change to ,2,Voicemail(. Now I use some hard phone, so I would better try another codec. Thanks for your good advice Try with another codec different than G.723. Use GSM o G.711 for this. You could disable G.723 in your sip.conf disallow=all allow=gsm allow=alaw allow=ulaw Hope this helps, Gus - Original Message - From: Hachy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 12:32 AM Subject: [Asterisk-Users] Re: Unable to use voicemail(Please suggestion) Hello all I got call log from Asterisk. I call to ext1001 from ext1002. But could not leave a message in the voice mail. Please help me. -- Executing Dial(SIP/1002-8217, SIP/1001|20) in new stack -- Called 1001 -- SIP/1001-25ce is ringing -- Nobody picked up in 2 ms == Spawn extension (sip, 1001, 2) exited non-zero on 'SIP/1002-8217 Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go down. And Icould not leave a messages. Please teach me how to resolve this problem. Here is configuration of Asterisk and Xlite. #sip.conf in Asterisk [general] port=5060 bindaddr=0.0.0.0 nortifymimetype=text/plain allow=all [1001] type=friend username=1001 secret=1001 host=dynamic defaultip=192.168.0.1 mailbox=1001 context=sip canreinvite=no [1002] type=friend username=1002 secret=1002 host=dynamic defaultip=192.168.0.1 mailbox=1002 context=sip canreinvite=no #extensions.conf in Asterisk [general] static=yes writeprotect=no [glovals] CONSOLE=Console/dsp [sip] exten = 1001,1,Dial(SIP/1001,20) exten = 1001,2,Voicemail(u1001) exten = 1001,102,Voicemail(b1001) exten = 1001,103,Hungup exten = 1002,1,Dial(SIP/1001,20) exten = 1002,2,Voicemail(u1002) exten = 1002,102,Voicemail(b1002) exten = 1002,103,Hungup #voicemail.conf in Asterisk [local] 1001 = 1001,1001,mail address 1002 = 1002,1002,mail address #Create mailbox by addmailbox already. #Client configuration User Name1001 1002 Authorization User same as username PAssword 1001 1002 Domain/Realm 192.168.0.120 SIP Proxy192.168.0.120 Here is call flow on this test. (c)2003 Xten Networks Inc. All rights reserved. Private build: 1008 SIP: 192.168.0.125:5061 RTP: 192.168.0.125:8000 NAT: 210.253.186.126 PXY#0: 192.168.0.120:5060 RECEIVE 192.168.0.120:5060 NOTIFY sip:[EMAIL PROTECTED]:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.120:5060;branch=z9hG4bK375605f3 From: asterisk sip:[EMAIL PROTECTED];tag=as633f7afa To: sip:[EMAIL PROTECTED]:5061 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: text/plain Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 SEND 192.168.0.120:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:5061 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 26502 INVITE Content-Type: application/sdp Content-Length: 301 v=0 o=1002 22002568 22002568 IN IP4 192.168.0.125 s=X-Lite c=IN IP4 192.168.0.125 t=0 0 m=audio 8000 RTP/AVP 4 0 8 3 101 a=rtpmap:4 G723/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:126 x-pro-encrypted/8000 RECEIVE 192.168.0.120:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.125:5061 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961 To: sip:[EMAIL PROTECTED];tag=as08d3281f Call-ID: [EMAIL PROTECTED] CSeq: 26502 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=05d14468 Content-Length: 0 SEND 192.168.0.120:5060 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:5061 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961 To: sip:[EMAIL PROTECTED];tag=as08d3281f Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 26502 ACK Max-Forwards: 70 Content-Length: 0 SEND 192.168.0.120:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.125:5061 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5061 Call-ID: [EMAIL PROTECTED] CSeq: 26503 INVITE Proxy-Authorization: Digest username=1002,realm=asterisk,nonce=
Re: [Asterisk-Users] * VOIP Terminator
I don't believe that this is particularly relevant to the Asterisk software -- perhaps another list can be created for discussion of what commercial services may or may not exist? On Thu, Nov 13, 2003 at 08:02:12AM +, Isamar Maia wrote: I am looking for a brazilian VOIP terminator for the states of Sao Paulo, Parana, Para and Bahia. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distintive Ring on x100p
http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk cvs ebuilds for gentoo portage system
any other gentoo users out there? I installed from cvs earlier today, and then figured, hell I might as well have some portage scripts to do it for me. so, a set of cvs ebuilds for zaptel, zapata, libpri and asterisk: http://bugs.gentoo.org/show_bug.cgi?id=33345 share and enjoy ++dg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LCR for i4l (least cost routing)?
Returning to the original question of this thread: Have you ever looked at an LCR implementation instead of building your own Db? I know that i4l is not so popular around here, but still this might be of interest: http://www.isdn4linux.de/faq/i4lfaq-3.html#ss3.26 3.26 feature_lcr: Can isdn4linux do least cost routing (LCR)? Yes, this feature is now being supported by isdnlog. What it does is that it allows isdnlog to choose your telephone provider when placing a call through your ISDN card, depending on the time of day and the current rate information. Since isdnlog 4.16 an external script is called (if configured) to change various ISP settings (e.g. DNS lookup, proxy setup,...). Note: the ABC-extensions (s. docu_abc) must be installed. Also, isdnlog should always be running (otherwise your dialout will be delayed by 3 seconds). If the ABC-extensions are not installed, isdnlog prints hints to the log file, which provider would have been chosen. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Licence may be changing..
Thu, 13 Nov 2003, Fresno CA, Adam Hart, spoke these words: Prehaps a novel thought but what about ODBC for asterisk? Isn't that the whole idea of standards and such, stop adding support for every db and just have odbc? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ODBC is not well supported on UNIX. You have to have the connector software, as well as the ODBC libs. -- Robert G. Werner [EMAIL PROTECTED] x5204, ICQ #311363925 Udall's Fourth Law: Any change or reform you make is going to have consequences you don't like. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] vm email notifications
On my asterisk server I have placed valid email addresses in the voicemail.conf file as to allow mailbox users to receive message notification. My problem is it appears that the messages are attempting to be sent but instead they are bouncing with a fatal error message like the one below: (reason: 550 [PERMFAIL] yahoo.com requires valid sender) First of all this is not the whole message but I think the pertinent part of it. What's really odd is that I can send mail out fine using pine, nothing bounces, also on my other asterisk service at another location this feature works fine. I use sendmail as my MTA. Is there anybody who has had a similar problem or who might be able to give me a suggestion as to how to correct this problem. Thanks AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Group dial codes ?(Newbie question)
Thanks David, I had a an excellent laugh at this (actually my first of the day, which makes it worth twice as much) ;- Philipp [peons] exten = 1,1,TakeOutTheTrash() exten = 2,1,WashTheDishes() exten = 3,1,CleanTheToilet() [rulers] include = peons exten = 800,1,FeedMeGrapes() ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pick up ringing exten
I have ZAP channels.. so i add the lines at zapata.conf and it does not work. When i dial *8 it return me a busy tone. my zapta.conf is..context=home group=2 pickupgroup=2 signalling=fxo_ks channel=2-3 callerid=FIJO 200 channel=3 callerid=INALAMBRICO 100 channel=2 Rich Adamson wrote: Is it possible with Asterisk to pick up ringing extension from other extension? So I do not have to run to other desk to pick up the phone. Sure, just add callgroup=2 pickupgroup=2 to each extension definition in sip.conf as an example. Dial *8 to pick up that ringing extn. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pause after dialed option
Hi guys I've set up a layered menu system on one of my asterisk servers where there is a main menu and several submenus; one for each department. Each menu plays a background intro message giving its various options. My problem is when I'm in the main menu and press the option to go to one of the submenus there seems to be a 5-8 second pause before it plays the background of the submenu. Is there any way that I can eliminate this pause? I do not have the problem if I dial a Zap channel or one of the voicemail boxes. It seems to connect to them immediately. Thanks a bunch. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * VOIP Terminator
Our forums are opened for anykind of discussions, including business opportunities, termination providers list, etc. Let's move some stuff to forums, so people who is interested can find and exchange information. Please join, at least we will empty this mailing list from some busienss discussions, because some people are just technical and they are not interested in any kind of commercial info so let's move on. We are also will be discussing commercial implementations for Asterisk projects. Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 8:05 PM Subject: Re: [Asterisk-Users] * VOIP Terminator I don't believe that this is particularly relevant to the Asterisk software -- perhaps another list can be created for discussion of what commercial services may or may not exist? On Thu, Nov 13, 2003 at 08:02:12AM +, Isamar Maia wrote: I am looking for a brazilian VOIP terminator for the states of Sao Paulo, Parana, Para and Bahia. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX channel and transfering calls
Hi again,
I'm attempting to figure out how to transfer calls from an IAX client. I
have read and seen on the list where if you put a ,t at the end of the
dial portion in the extensions.conf file that you should be able to use
the # to park and transfer calls. I have not found this to be the case. I
have tried it several different ways and I can't seem to get it to work.
Can anyone send me a sample of what this line should look like? Also, I'm
having a bit of confusion as to exactly where in the extensions.conf file
this should go. I'm currently trying it by putting it in the line that
passes the dialout to NuFone in my extensions.conf file. However I am
unsure if this will still allow me to park calls or even transfer calls if
they are incoming from other IAX clients or from another context.
Currently, my dial line looks like so:
exten = _91NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:1}
I tried putting the comma t after the closing bracket and before the
closing bracket. No success in either case. Any suggestions?
AJ
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RE: [Asterisk-Users] pause after dialed option
Without looking at your extensions.conf I can only guess that maybe the first digit(s) of your exten aren't unique and asterisk is waiting for a digit timeout. You can shorten your timeout or make your extensions unique. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 6:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pause after dialed option Hi guys I've set up a layered menu system on one of my asterisk servers where there is a main menu and several submenus; one for each department. Each menu plays a background intro message giving its various options. My problem is when I'm in the main menu and press the option to go to one of the submenus there seems to be a 5-8 second pause before it plays the background of the submenu. Is there any way that I can eliminate this pause? I do not have the problem if I dial a Zap channel or one of the voicemail boxes. It seems to connect to them immediately. Thanks a bunch. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pause after dialed option
Ok thanks. I'll try to shorten the digit timeout. On Wed, 12 Nov 2003, David Carr wrote: Without looking at your extensions.conf I can only guess that maybe the first digit(s) of your exten aren't unique and asterisk is waiting for a digit timeout. You can shorten your timeout or make your extensions unique. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 6:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pause after dialed option Hi guys I've set up a layered menu system on one of my asterisk servers where there is a main menu and several submenus; one for each department. Each menu plays a background intro message giving its various options. My problem is when I'm in the main menu and press the option to go to one of the submenus there seems to be a 5-8 second pause before it plays the background of the submenu. Is there any way that I can eliminate this pause? I do not have the problem if I dial a Zap channel or one of the voicemail boxes. It seems to connect to them immediately. Thanks a bunch. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pause after dialed option
I had experienced this problem before. I found this to be related to 2 items. Firstly, try not to use the s,1 starting each submenu. Secondly, if there are more than 20 sub menus, you will get this delay problem. Why I do not know. I reordered and regrouped and the problem disappeared. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Carr Sent: Thursday, 13 November 2003 1:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] pause after dialed option Without looking at your extensions.conf I can only guess that maybe the first digit(s) of your exten aren't unique and asterisk is waiting for a digit timeout. You can shorten your timeout or make your extensions unique. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 6:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pause after dialed option Hi guys I've set up a layered menu system on one of my asterisk servers where there is a main menu and several submenus; one for each department. Each menu plays a background intro message giving its various options. My problem is when I'm in the main menu and press the option to go to one of the submenus there seems to be a 5-8 second pause before it plays the background of the submenu. Is there any way that I can eliminate this pause? I do not have the problem if I dial a Zap channel or one of the voicemail boxes. It seems to connect to them immediately. Thanks a bunch. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pause after dialed option
So what do you use instead of s,1? My s extensions set things like response timeout, digit timeout, etc. Thanks again. AJ On Thu, 13 Nov 2003, Master Abi wrote: I had experienced this problem before. I found this to be related to 2 items. Firstly, try not to use the s,1 starting each submenu. Secondly, if there are more than 20 sub menus, you will get this delay problem. Why I do not know. I reordered and regrouped and the problem disappeared. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Carr Sent: Thursday, 13 November 2003 1:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] pause after dialed option Without looking at your extensions.conf I can only guess that maybe the first digit(s) of your exten aren't unique and asterisk is waiting for a digit timeout. You can shorten your timeout or make your extensions unique. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 6:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pause after dialed option Hi guys I've set up a layered menu system on one of my asterisk servers where there is a main menu and several submenus; one for each department. Each menu plays a background intro message giving its various options. My problem is when I'm in the main menu and press the option to go to one of the submenus there seems to be a 5-8 second pause before it plays the background of the submenu. Is there any way that I can eliminate this pause? I do not have the problem if I dial a Zap channel or one of the voicemail boxes. It seems to connect to them immediately. Thanks a bunch. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI Functions
Does anyone know where I can get a list of ADSI functions.. Example *70 (No Call Waiting), Flash = Flash, Hold = ??? Thank you, -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] vm email notifications
If you send out via pine.. Who are you sending the mail out as... Also if * sends the mail out who is it sending it out as? Example if you host file only has loopback with localhost then it might be sent out as [EMAIL PROTECTED] And if Yahoo can resolve that domain it wont accept the email..(helps to prevent against spam) Hope this helps. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Posted At: Wednesday, November 12, 2003 8:29 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] vm email notifications Subject: [Asterisk-Users] vm email notifications On my asterisk server I have placed valid email addresses in the voicemail.conf file as to allow mailbox users to receive message notification. My problem is it appears that the messages are attempting to be sent but instead they are bouncing with a fatal error message like the one below: (reason: 550 [PERMFAIL] yahoo.com requires valid sender) First of all this is not the whole message but I think the pertinent part of it. What's really odd is that I can send mail out fine using pine, nothing bounces, also on my other asterisk service at another location this feature works fine. I use sendmail as my MTA. Is there anybody who has had a similar problem or who might be able to give me a suggestion as to how to correct this problem. Thanks AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pause after dialed option
Use like this... [mainmenu] exten = s,1,Goto(sales|100|1) exten = s,2,Goto(support|200|1) [sales] exten = 100,1,Answer ; Answer the line exten = 100,2,DigitTimeout,5 ; Maximum Timeout between digits exten = 100,3,ResponseTimeout,10 ; Maximum Timeout awaiting response exten = 100,4,BackGround,mainmenu; Play Main Menu [support] exten = 200,1,Answer ; Answer the line exten = 200,2,DigitTimeout,5 ; Maximum Timeout between digits exten = 200,3,ResponseTimeout,10 ; Maximum Timeout awaiting response exten = 200,4,BackGround,mainmenu; Play Main Menu .. etc, etc Also, I don't think putting digit timeouts are always required, but I did find Answer is a fairly safe bet. Try and use s extension is a minimum. Master -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, 13 November 2003 2:11 PM To: Master Abi Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] pause after dialed option So what do you use instead of s,1? My s extensions set things like response timeout, digit timeout, etc. Thanks again. AJ On Thu, 13 Nov 2003, Master Abi wrote: I had experienced this problem before. I found this to be related to 2 items. Firstly, try not to use the s,1 starting each submenu. Secondly, if there are more than 20 sub menus, you will get this delay problem. Why I do not know. I reordered and regrouped and the problem disappeared. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Carr Sent: Thursday, 13 November 2003 1:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] pause after dialed option Without looking at your extensions.conf I can only guess that maybe the first digit(s) of your exten aren't unique and asterisk is waiting for a digit timeout. You can shorten your timeout or make your extensions unique. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 6:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pause after dialed option Hi guys I've set up a layered menu system on one of my asterisk servers where there is a main menu and several submenus; one for each department. Each menu plays a background intro message giving its various options. My problem is when I'm in the main menu and press the option to go to one of the submenus there seems to be a 5-8 second pause before it plays the background of the submenu. Is there any way that I can eliminate this pause? I do not have the problem if I dial a Zap channel or one of the voicemail boxes. It seems to connect to them immediately. Thanks a bunch. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] vm email notifications
Well in pine I'm sending it out as [EMAIL PROTECTED] In asterisk in the actual voicemail.conf file I set the From field to a valid user name like [EMAIL PROTECTED] However for the loopback I have several names like local host.localdomain and myhost.mydomain.com which actually is probably unresolvable. AJ On Wed, 12 Nov 2003, PBX wrote: If you send out via pine.. Who are you sending the mail out as... Also if * sends the mail out who is it sending it out as? Example if you host file only has loopback with localhost then it might be sent out as [EMAIL PROTECTED] And if Yahoo can resolve that domain it wont accept the email..(helps to prevent against spam) Hope this helps. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Posted At: Wednesday, November 12, 2003 8:29 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] vm email notifications Subject: [Asterisk-Users] vm email notifications On my asterisk server I have placed valid email addresses in the voicemail.conf file as to allow mailbox users to receive message notification. My problem is it appears that the messages are attempting to be sent but instead they are bouncing with a fatal error message like the one below: (reason: 550 [PERMFAIL] yahoo.com requires valid sender) First of all this is not the whole message but I think the pertinent part of it. What's really odd is that I can send mail out fine using pine, nothing bounces, also on my other asterisk service at another location this feature works fine. I use sendmail as my MTA. Is there anybody who has had a similar problem or who might be able to give me a suggestion as to how to correct this problem. Thanks AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA 2000 and 404 not found
I have a Sipura SPA2000 2 line SIP FXS box with line 1 on port 5060 and line 2 on 5061. The SPA2000 is on IP address 192.168.17.6, and the asterisk box is on 102.168.17.2. Both SPA2000 ports(5060 and 5061) share the same IP address. Every minute I repeatedly get the following output: SIP Debugging Enabled 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.17.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK60fe7596 From: asterisk sip:[EMAIL PROTECTED];tag=as1cf7898d To: sip:192.168.17.6 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.17.6:5060 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.17.6:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK0178ca1c From: asterisk sip:[EMAIL PROTECTED];tag=as6a42fcc6 To: sip:192.168.17.6:5061 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.17.6:5061 Sip read: SIP/2.0 404 Not Found To: sip:192.168.17.6 From: asterisk sip:[EMAIL PROTECTED];tag=as1cf7898d Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK60fe7596 Server: Sipura/SPA2000-1.0.9 Content-Length: 0 8 headers, 0 lines Sip read: SIP/2.0 404 Not Found To: sip:192.168.17.6:5061 From: asterisk sip:[EMAIL PROTECTED];tag=as6a42fcc6 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.17.2:5060;branch=z9hG4bK0178ca1c Server: Sipura/SPA2000-1.0.9 Content-Length: 0 8 headers, 0 lines *CLI sip no debug SIP Debugging Disabled Here's what's in sip.conf: [general] port=5060 bindaddr=192.168.17.2 tos=lowdelay disallow=all allow=ulaw context=default ; ; SIP Entry for sipura line 1 ; This phone is allowed to dial extensions and local phone numbers ; [101] type=friend host=dynamic context=house-toll reinvite=no canreinvite=no qualify=300 secret=xx callerid=Sipura Line 1 101 username=101 mailbox=101 ; Sample for sipura line 2 ; This phone is allowed to dial extensions and local phone numbers ; [102] type=friend host=dynamic context=house-toll reinvite=no canreinvite=no qualify=300 secret=yy callerid=Sipura Line 2 102 username=102 mailbox=102 nat=0 Note that 192.168.17.6:5061 seems to have a problem with 404 not found, wheras 192.168.17.6:5060 does not. Could Asterisk be getting confused about a device with two ports sharing the same IP address? I don't seem to be seeing any traffic being logged from the SPA2000 to Asterisk; it all seems to be going from 192.168.17.2 to 192.168.17.6. If anyone could shed some light on what is going on here it would be sincerely appreciated. Steve Rodgers San Diego, CA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
