Re: [Asterisk-Users] MusicOnHold
Hi you all,
Thanks for the help, got it working! The mpg123 in combination with the
mpg123 directory (executable MUST be in /usr/local/bin AND in /usr/bin)
was the problem that MOH was not working
Thanks!
Jeroen
Brian West wrote:
put mpg123 in /usr/bin
bkw
On Tue, 19 Aug 2003, Asterisk - linux - JVB wrote:
Yes I linked all the mp3 and mpg extensions with the mpg123 program
(/usr/local/bin) ... but still not able to get the music on hold playing
Getting curious now what I am doing wrong ...
Andrew Joakimsen wrote:
Did you remove the symlink for mpg123 - mpg321 and replace it with a
symlink to the correct location for mpg123? I have also noticed when
using the eStara softphone that if push to talk is enabled if you do
not press ctrl to "talk" you cannot hear the music on hold, as well as
some other oddities.
-Original Message-
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] *On Behalf Of *Asterisk
- linux - JVB
*Sent:* Tuesday, August 19, 2003 3:13 PM
*To:* [EMAIL PROTECTED]
*Subject:* Re: [Asterisk-Users] MusicOnHold
Andrew, thanks I already have got mpg123 installed and working.
However still got the MOH stuff up and running.
Got a feeling it has got something to do with the stottering audio
(see my other message on this list)
NOTICE[1116949808]: File res_musiconhold.c, Line 258
(monmp3thread): Request to schedule in the past?!?!
Any suggestions are welcome
Andrew Joakimsen wrote:
http://www.marko.net/asterisk/archives/0207/0097.html
-Original Message-
*From:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]*On Behalf Of *Asterisk
- linux - JVB
*Sent:* Tuesday, August 19, 2003 6:12 AM
*To:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
*Subject:* [Asterisk-Users] MusicOnHold
Does anybody know why I can NOT hear the MusicOnHold - using SJphone
on another PC in our network (normal playback is not a problem) .
See the * output and the line configured in extension.conf below (also
mp3player does not function)
Any suggestions?
*Asterisk output:*
*CLI -- Executing WaitMusicOnHold("SIP/jeroen-bf54", "30") in new
stack
-- Started music on hold, class 'default', on SIP/jeroen-bf54
-- Stopped music on hold on SIP/jeroen-bf54
-- Timeout on SIP/jeroen-bf54
-- Executing Goto("SIP/jeroen-bf54", "#|1") in new stack
-- Goto (default,#,1)
-- Sent into invalid extension '#' in context 'default' on
SIP/jeroen-bf54
-- Executing Playback("SIP/jeroen-bf54", "invalid") in new stack
-- Playing 'invalid'
*Extension.conf*
exten = 4000,1,WaitMusicOnHold,30
exten = 4001,1,mp3player(/var/lib/asterisk/mohmp3/sample-hold.mp3)
*musiconhold.conf*
[classes]
default = quietmp3:/var/lib/asterisk/mohmp3
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[Asterisk-Users] Limit Number of user in Conference
Hello, Is it possible to limit the number of user in a particular conference room? Foong
[Asterisk-Users] echo on the sip side
so i call from a sip phone (grandstream) to a cell via x100p PSTN side hears everything nice, no echo. on the SIP side I hear myself about .1 to .2 sec later... any thoughts on how to resolve this. mucho thanks to everyone that has been helpful :) john ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323
should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ 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] Where to find correct ver of OpenH323 PWLIB for Chan_h323
I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 04:53 PM Please respond to asterisk-users should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ 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] Where to find correct ver of OpenH323 PWLIB for Chan_h323
you can do cvs update -r v1_11_7 to get version 1.11.7 for openh323 Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:51 PM Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 04:53 PM Please respond to asterisk-users should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ 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] Where to find correct ver of OpenH323 PWLIB for Chan_h323
Thanks - because of my ignorance using the CVS archive - could you please give me the full command - thanks. Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 05:03 PM Please respond to asterisk-users you can do cvs update -r v1_11_7 to get version 1.11.7 for openh323 Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:51 PM Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 04:53 PM Please respond to asterisk-users should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ 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
[Asterisk-Users] SIP using which codec?
Is there a way to determine what codec the remote server wants to use in a SIP session for an incoming call by looking at something, possiby sip debug?
Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323
export CVSROOT=:pserver:[EMAIL PROTECTED]:/cvsroot/openh323 cvs login CVS password: press enter cd /root cvs checkout openh323 cd openh323 cvs update -r v1_11_7 I usually get the latest version then down grade to older version, If you know how to get the older version directly, let me know. Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 3:05 PM Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Thanks - because of my ignorance using the CVS archive - could you please give me the full command - thanks. Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 05:03 PM Please respond to asterisk-users you can do cvs update -r v1_11_7 to get version 1.11.7 for openh323 Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:51 PM Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 04:53 PM Please respond to asterisk-users should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP using which codec?
Is there a way to determine what codec the remote server wants to use in a SIP session for an incoming call by looking at something, possiby sip debug? Take a look in the archives this was covered a couple of days ago.. the command you are looking for is sip show channels.. and then look in the format column.. the formula for determining the format was posted in the previous discussion and i can't rememebr it off the top of my head.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit Number of user in Conference
Citeren Chee Foong [EMAIL PROTECTED]: Hello, Is it possible to limit the number of user in a particular conference room? Foong Hi, I think the easiest way is to create a counter that adds one when a user joins and subtracts when a user leaves or hangs up. A few simple AGI scripts could do it. Just make sure you lock the counter when updating so as to avoid miscounting when two channels join at the same time.. -- Best regards, Florian Overkamp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP using which codec?
I already tried that, it says unknown. I suspect it is requiring the G723 codec. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: Wednesday, August 20, 2003 3:28 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP using which codec? Is there a way to determine what codec the remote server wants to use in a SIP session for an incoming call by looking at something, possiby sip debug? Take a look in the archives this was covered a couple of days ago.. the command you are looking for is sip show channels.. and then look in the format column.. the formula for determining the format was posted in the previous discussion and i can't rememebr it off the top of my head.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] SIP using which codec?
At 7:27 AM + 8/20/03, WipeOut . wrote: Is there a way to determine what codec the remote server wants to use in a SIP session for an incoming call by looking at something, possiby sip debug? Take a look in the archives this was covered a couple of days ago.. the command you are looking for is sip show channels.. and then look in the format column.. the formula for determining the format was posted in the previous discussion and i can't rememebr it off the top of my head.. Later.. I think that the question is a bit more subtle than that. The question says wants to use, not does use. Currently, I think the only way you'll find this is with a SIP debug, looking at the SDP request. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] weird error message with zaptel
Hi, While trying to update latest CVS, during make install to zaptel, I got weird error message (down under). Anyone had same kind of problem? What would be the solution? -Johanna _ fi /sbin/depmod -a depmod: cannot read ELF header from /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/vmlinux-obj.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/asuscom.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/avm_a1.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/avm_a1p.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/avm_pci.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/bkm_a4t.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/bkm_a8.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/callc.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/diva.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/elsa.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/enternow_pci.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/gazel.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/hfc_pci.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/hfc_sx.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/hfcscard.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/isar.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/isdnl1.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/isdnl2.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/isdnl3.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/isurf.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/ix1_micro.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/l3_1tr6.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/l3dss1.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/l3ni1.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/lmgr.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/mic.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/niccy.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/nj_s.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/nj_u.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/q931.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/s0box.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/saphir.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/sedlbauer.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/sportster.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/tei.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/teleint.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/teles0.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/teles3.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/telespci.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/w6692.o make: [install] Error 1 (ignored) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G723 (was SIP using which codec?)
Actually I got it working right before I gave up (I had the wrong line in my config commented out) But now I get these messages when I try to playback a recording: NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable to find a path from GSM to G723 WARNING[16401]: File file.c, Line 722 (ast_streamfile): Unable to open transfer (format G723): No such file or directory WARNING[16401]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on SIP/packet8.net-dab9 for transfer And when I try to play music on hold: NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable to find a path from SLINR to G723 WARNING[16401]: File res_musiconhold.c, Line 421 (moh_alloc): Unable to set 'SIP/packet8.net-dab9' to signed linear format This is the missing link in my system, I greatly appreciate any help that can be provided. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Wednesday, August 20, 2003 4:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP using which codec? At 7:27 AM + 8/20/03, WipeOut . wrote: Is there a way to determine what codec the remote server wants to use in a SIP session for an incoming call by looking at something, possiby sip debug? Take a look in the archives this was covered a couple of days ago.. the command you are looking for is sip show channels.. and then look in the format column.. the formula for determining the format was posted in the previous discussion and i can't rememebr it off the top of my head.. Later.. I think that the question is a bit more subtle than that. The question says wants to use, not does use. Currently, I think the only way you'll find this is with a SIP debug, looking at the SDP request. JT ___ 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] Dialogic cards...
Are the dialogic DTI series cards supported in asterisk? I know there's standard API, but I don't know if it supports only the cards listed on the digium site, or if it will support ALL dialogic cards.. Sorry, I *AM* a newbie to this stuff, just trying to get my hands on a good card. Thanks.
[Asterisk-Users] snom100(with latest firmware) screeching noise when doing transfers,
Hello, I`ve upgraded my Snom 100 to the new version of firmware that is snom100-2.00n-SIP.bin, and they did fix the GSM, that is the nice news, it is very clear and nice almost indistinguishable from the G.711. But there still a problem, when doing transfers or for example diling the 500 ( demo iax connection to digium) there is a screeching noise and the sound of static. For example when dialing the 500, right after the words please wait a moment whlie I attempt to make a connection there`s this screech. Is this the asterisk issue or the Snom issue? I`ve tried this with couple month old CVS, and the todays version too Thnx -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue
Hello, I have problem setting up queue. Everything works nice, but I would like to have some kind of announcement while playing MusicOnHold. Is it possible? If yes how I can set it up. Bartek
Re: [Asterisk-Users] Limit Number of user in Conference
On Wed, 2003-08-20 at 02:55, Florian Overkamp wrote: Citeren Chee Foong [EMAIL PROTECTED]: Hello, Is it possible to limit the number of user in a particular conference room? Foong Hi, I think the easiest way is to create a counter that adds one when a user joins and subtracts when a user leaves or hangs up. A few simple AGI scripts could do it. Just make sure you lock the counter when updating so as to avoid miscounting when two channels join at the same time.. Problem with that is making sure that calls exit the conference and passes through your exit AGI. This is why I submitted a patch some time ago to modify MeetMeCount to set a variable to the number of users in the conference. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] weird error message with zaptel
On Wed, 2003-08-20 at 02:07, Johanna Kangas wrote: Hi, While trying to update latest CVS, during make install to zaptel, I got weird error message (down under). Anyone had same kind of problem? What would be the solution? OPEN EYES AND READ. Your problem is in hisax, not zaptel. -Johanna _ fi /sbin/depmod -a depmod: cannot read ELF header from /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/vmlinux-obj.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/asuscom.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/avm_a1.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/avm_a1p.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/avm_pci.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/bkm_a4t.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/bkm_a8.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/callc.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/diva.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/elsa.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/enternow_pci.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/gazel.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/hfc_pci.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/hfc_sx.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/hfcscard.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/isar.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/isdnl1.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/isdnl2.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/isdnl3.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/isurf.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/ix1_micro.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/l3_1tr6.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/l3dss1.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/l3ni1.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/lmgr.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/mic.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/niccy.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/nj_s.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/nj_u.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/q931.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/s0box.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/saphir.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/sedlbauer.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/sportster.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/tei.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/teleint.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/teles0.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/teles3.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/telespci.o depmod: *** Unresolved symbols in /lib/modules/2.4.20-8/kernel/drivers/isdn/hisax/w6692.o make: [install] Error 1 (ignored) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Virus propagation by asterisk usermember.
On Wed, 2003-08-20 at 00:50, Dave Cotton wrote: On Tue, 2003-08-19 at 22:42, Steven Critchfield wrote: So far I have received 43 since 3am till 3:45pm According to mails in the ser list it's there also, and around the same time of day. Sounds appropriate since I have received a bounce from iconnecthere customer support due to this luser. But let's not just have a go at the users, even the worm writers acknowledge the real culprit, quoted from a message inside BlasterA code. I just want to say LOVE YOU SAN!! billy gates why do you make this possible ? Stop making money and fix your software!! And do you expect a crack dealer to stop selling crack? This won't change till enough lusers are educated. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] reload not working
I upgraded to the latest CVS yesterday (and this morning again), and whenever I execute the reload command Asterisk seems to hang. While the current calls aren't dropped, no new calls can be made. The CLI isn't responding properly either. The only way to get going again is to exit the CLI and stop Asterisk and start again. Any comments? Thanks, Marcus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reload not working
yes start it with asterisk -gc watch and see what the error is. bkw On Wed, 20 Aug 2003, Marcus Adolfsson wrote: I upgraded to the latest CVS yesterday (and this morning again), and whenever I execute the reload command Asterisk seems to hang. While the current calls aren't dropped, no new calls can be made. The CLI isn't responding properly either. The only way to get going again is to exit the CLI and stop Asterisk and start again. Any comments? Thanks, Marcus ___ 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] Where to find correct ver of OpenH323 PWLIBfor Chan_h323
You can also check www.openh323.org/bin/ bkw On Wed, 20 Aug 2003, Steven Thomas wrote: Thanks - because of my ignorance using the CVS archive - could you please give me the full command - thanks. Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 05:03 PM Please respond to asterisk-users you can do cvs update -r v1_11_7 to get version 1.11.7 for openh323 Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:51 PM Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 04:53 PM Please respond to asterisk-users should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
It is possible to connect ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank to Asterisk ? Somebody offered me that hardware, but I do not know if thats good hardware for Asterisk. rgs, Bartosz
Re: [Asterisk-Users] G723 (was SIP using which codec?)
MOH requires that Asterisk transcodes (It also has to transcode to for PSTN calls and voicemail and playing any sound files). Asterisk can't transcode to or from G723. Nope. Doesn't work. May very well never work. Use a different codec. On Wed, 2003-08-20 at 03:47, Andrew Joakimsen wrote: Actually I got it working right before I gave up (I had the wrong line in my config commented out) But now I get these messages when I try to playback a recording: NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable to find a path from GSM to G723 WARNING[16401]: File file.c, Line 722 (ast_streamfile): Unable to open transfer (format G723): No such file or directory WARNING[16401]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on SIP/packet8.net-dab9 for transfer And when I try to play music on hold: NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable to find a path from SLINR to G723 WARNING[16401]: File res_musiconhold.c, Line 421 (moh_alloc): Unable to set 'SIP/packet8.net-dab9' to signed linear format This is the missing link in my system, I greatly appreciate any help that can be provided. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Wednesday, August 20, 2003 4:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP using which codec? At 7:27 AM + 8/20/03, WipeOut . wrote: Is there a way to determine what codec the remote server wants to use in a SIP session for an incoming call by looking at something, possiby sip debug? Take a look in the archives this was covered a couple of days ago.. the command you are looking for is sip show channels.. and then look in the format column.. the formula for determining the format was posted in the previous discussion and i can't rememebr it off the top of my head.. Later.. I think that the question is a bit more subtle than that. The question says wants to use, not does use. Currently, I think the only way you'll find this is with a SIP debug, looking at the SDP request. JT ___ 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 -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference call
Conference call problem - do not have any special hardware added to the system yet. Did the following: * Uncommented the ztdummy.c in the Makefile (/zaptel) - recompiled all * Extensions.conf exten = 2675,1,meetme,2675 * meetme.conf conf = 2675 When I dial 2675 I get the message That is not a valid conference number, please try again. I have read in the archives several people with the same problem but never found the solution for it... I do see that * gives the following messages *CLI WARNING[1239498032]: File app_meetme.c, Line 153 (build_conf): Unable to open pseudo channel Any ideas? Cheers - Jeroen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App Directory issues-again?
Hi, I've seen some postings on the Directory application, but haven't seen too many resolution postings. Has anyone experienced where the Directory app doesn't even answer when called? For example, using the config below, dialing 899 results in just a continual ringing sound. extensions.conf exten = 899,1,Directory(local) exten = 899,2,Hangup [local] exten = 8000,1,Dial(SIP/8000) ..various extensions defined voicemail.conf [local] 8000 = 1234,John Morris,[EMAIL PROTECTED] ...etc... Is there a config problem or are others having this issue too? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
The FXO ports will only allow you to connect phone lines, not actual phones, but since FXO ports are more expensive in general than FXS ones, it's likely you could find someone to trade. We probably should have a list dedicated to trading/selling/buying asterisk related hardware, but failing that i would suggest people just contact you off-list. Mark On Wed, 20 Aug 2003, Bartosz Jozwiak wrote: It is possible to connect ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank to Asterisk ? Somebody offered me that hardware, but I do not know if thats good hardware for Asterisk. rgs, Bartosz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference call
Jeroen wrote: Conference call problem - do not have any special hardware added to the system yet. Did the following: * Uncommented the ztdummy.c in the Makefile (/zaptel) - recompiled all [...] Any ideas? When you do an lsmod, is ztdummy listed? If you do a depmod -a is there any output, and if so, what is it? -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Virus propagation by asterisk usermember.
On Wed, 2003-08-20 at 14:36, Steven Critchfield wrote: And do you expect a crack dealer to stop selling crack? This won't change till enough lusers are educated. No Gates got of the hook last time, too political to discuss here but perhaps John Dvorak's article hits the nail on the head. http://www.pcmag.com/article2/0,4149,1224343,00.asp Could be a good earner. Here in France you cannot teach your kids to drive without the input of, and payment to, a driving school. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
On Wed, 2003-08-20 at 07:58, Mark Spencer wrote: The FXO ports will only allow you to connect phone lines, not actual phones, but since FXO ports are more expensive in general than FXS ones, it's likely you could find someone to trade. We probably should have a list dedicated to trading/selling/buying asterisk related hardware, but failing that i would suggest people just contact you off-list. Yeah, but will it work? What if he wants 24 port FXO, not FXS? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
I want to connect analog telephone lines only. The analog lines telecom gives you :) - Original Message - From: Steve Meyers [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 11:34 AM Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank On Wed, 2003-08-20 at 07:58, Mark Spencer wrote: The FXO ports will only allow you to connect phone lines, not actual phones, but since FXO ports are more expensive in general than FXS ones, it's likely you could find someone to trade. We probably should have a list dedicated to trading/selling/buying asterisk related hardware, but failing that i would suggest people just contact you off-list. Yeah, but will it work? What if he wants 24 port FXO, not FXS? Steve ___ 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] Conference call
Hi Almaw, The following: * Asterisk up running * lsmod - no ztdummy module loaded * depmod -a - no output So I tried to modprobe the ztdummy --- with result! Conference is running without problems .. do you knwo if there is a manual or something like that which summarises all these nice-2-know? (I am not a linux expert but is the module loaded automatically at the next startup?) [[EMAIL PROTECTED] root]# depmod -a [EMAIL PROTECTED] root]# modprobe ztdummy [EMAIL PROTECTED] root]# lsmod Module Size Used by Not tainted ztdummy 2548 0 (unused) zaptel 180032 0 [ztdummy] ppp_generic 2 0 [zaptel] slhc 6756 0 [ppp_generic] snd-pcm-oss 45284 0 (autoclean) (unused) snd-pcm 85280 0 (autoclean) [snd-pcm-oss] snd-page-alloc 9844 0 (autoclean) [snd-pcm] snd-timer 19556 0 (autoclean) [snd-pcm] snd-mixer-oss 16408 0 (autoclean) [snd-pcm-oss] snd 43076 0 (autoclean) [snd-pcm-oss snd-pcm snd-timer snd-mixer-oss] es1371 30824 1 (autoclean) gameport 3364 0 (autoclean) [es1371] ac97_codec 14568 0 (autoclean) [es1371] soundcore 6404 4 (autoclean) [snd es1371] parport_pc 19076 1 (autoclean) lp 8996 0 (autoclean) parport 37056 1 (autoclean) [parport_pc lp] autofs 13268 0 (autoclean) (unused) e100 54564 1 8139too 18120 1 mii 3976 0 [8139too] ipt_REJECT 3992 6 (autoclean) iptable_filter 2412 1 (autoclean) ip_tables 15096 2 [ipt_REJECT iptable_filter] sg 36524 0 (autoclean) sr_mod 18136 0 (autoclean) ide-scsi 12208 0 scsi_mod 107544 3 [sg sr_mod ide-scsi] ide-cd 35712 0 cdrom 33728 0 [sr_mod ide-cd] printer 8928 0 keybdev 2976 0 (unused) mousedev 5556 1 hid 22244 0 (unused) input 5856 0 [keybdev mousedev hid] usb-uhci 26412 0 [ztdummy] ehci-hcd 20072 0 (unused) usbcore 79040 1 [printer hid usb-uhci ehci-hcd] ext3 70784 2 jbd 51924 2 [ext3] [EMAIL PROTECTED] root]#
Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
Then yes, it will work and do what you're looking for it to do. On Wed, 20 Aug 2003, Bartosz Jozwiak wrote: I want to connect analog telephone lines only. The analog lines telecom gives you :) - Original Message - From: Steve Meyers [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 11:34 AM Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank On Wed, 2003-08-20 at 07:58, Mark Spencer wrote: The FXO ports will only allow you to connect phone lines, not actual phones, but since FXO ports are more expensive in general than FXS ones, it's likely you could find someone to trade. We probably should have a list dedicated to trading/selling/buying asterisk related hardware, but failing that i would suggest people just contact you off-list. Yeah, but will it work? What if he wants 24 port FXO, not FXS? Steve ___ 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 ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
Thanks for your help. Bart - Original Message - From: Steve Creel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 11:47 AM Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank Then yes, it will work and do what you're looking for it to do. On Wed, 20 Aug 2003, Bartosz Jozwiak wrote: I want to connect analog telephone lines only. The analog lines telecom gives you :) - Original Message - From: Steve Meyers [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 11:34 AM Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank On Wed, 2003-08-20 at 07:58, Mark Spencer wrote: The FXO ports will only allow you to connect phone lines, not actual phones, but since FXO ports are more expensive in general than FXS ones, it's likely you could find someone to trade. We probably should have a list dedicated to trading/selling/buying asterisk related hardware, but failing that i would suggest people just contact you off-list. Yeah, but will it work? What if he wants 24 port FXO, not FXS? Steve ___ 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 ___ Steve Creel[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] Hardware question
Hello, Again one more question about hardware. What could you suggest me to buy. I need hardware to connect let's say 4 analog lines. (FXO). This hardware should "talk" to Asterisk of course.. Thanks very much for some advices :) Bartek
[Asterisk-Users] Re: Asterisk diskless server, a web page with more info?
Hello, I've had quite a few requests for this info, so I thought I'd copy this to the list as well. Since I don't really monitor the list anymore, queries should be directed back to me if you have problems. On Wed, 2003-08-20 at 04:08, Sjur Eivind Usken wrote: Dear Ben, I saw your posting on the newslist about the diskless configuration you have done. Can you please let me know more details around it. I'm familiar how to boot a diskless pc from the ROM on the network cards, so I would like to know the following: In my case, I used network cards with Intel's PXE (Preboot eXecution Envirionment) boot loader. These cards have a built-in DHCP and TFTP client. You can use pxeboot (by H. Peter Anvin, the same person that wrote isolinux and syslinux), which can be found at http://syslinux.zytor.com. I then created a custom built kernel, compiled some static binaries, and created an initial ramdisk. This is documented elsewhere, but the quick steps are dd if=/dev/zero of=disk.img bs=1024 count=65536 /sbin/mke2fs disk.img # (you'll get a warning here that its not a block device, hit y to accept) mount -o loop disk.img /mnt THen copy your kernel and static binaries in there. Unmount, gzip -9 disk.img (to make for faster booting), drop the gzipped file in the tftp directory per the pxelinux documentation, configure dhcpd, and go! Ok, so there are a few more steps in there (like compile asterisk, create /etc /tmp and /var, configure syslog) but that should give you a start. I followed the existing diskless linux howto and the nfs-root linux howto, with heavy modifications. If you get seriously stuck, shoot me an email back and I'll send you a copy of my configuration files. When I get some time (don't hold your breath) I may put up a webpage detailing what I did with example configuration files. You use a pc with just 128mb memory, that you install a small linux distro. Is it possible to have a copy of this? I didn't make a distro per se, but once I got the image stable, I just copied it and changed some key /etc files and it was good to go. Since I use a cron job which calls lynx to read from a PHP/MySQL page to generate my configuration, there is very little to do per image. This was a key design requirement so that I could roll out 20 of these things in short order if necessary. Are any other files available, e.g. your booting configuration and the linux setup? I had a look at your website, but didn't find any information there. You had a look at my website? I wasn't aware I had one of those... See above comment, hopefully I can put something up for real one of these days. This could make asterisk easy to expand to a lot of branches, with a centralized configuration and easy redundancy. Yep! My design has solved this problem and has been stable for over a year. Its met my goals pretty well. Answers would be greatly appreciated! Good luck on your project! -- Ben Klang, KF4WBX signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] echo on the sip side
Did you enable echocancel and echocancelwhilebridged? Did you put them in the correct location in the zapata.conf ? It has to be before the channel statement (this is what threw me for a week) If you tail -f debug in the /var/log/asterisk you can watch the call and see if echo cancel was kicking in Lee - Original Message - From: John Brown [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:23 AM Subject: [Asterisk-Users] echo on the sip side so i call from a sip phone (grandstream) to a cell via x100p PSTN side hears everything nice, no echo. on the SIP side I hear myself about .1 to .2 sec later... any thoughts on how to resolve this. mucho thanks to everyone that has been helpful :) john ___ 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] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
Also dont forget FXO cards must be L2 with minimum REV K firmware to support Caller ID. see http://www.wwworks-inc.com/asterisk/ Then yes, it will work and do what you're looking for it to do. On Wed, 20 Aug 2003, Bartosz Jozwiak wrote: I want to connect analog telephone lines only. The analog lines telecom gives you :) - Original Message - From: Steve Meyers [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 11:34 AM Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank On Wed, 2003-08-20 at 07:58, Mark Spencer wrote: The FXO ports will only allow you to connect phone lines, not actual phones, but since FXO ports are more expensive in general than FXS ones, it's likely you could find someone to trade. We probably should have a list dedicated to trading/selling/buying asterisk related hardware, but failing that i would suggest people just contact you off-list. Yeah, but will it work? What if he wants 24 port FXO, not FXS? Steve ___ 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 ___ Steve Creel[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
RE: [Asterisk-Users] reload not working
Martin - et all, I'm having the same issue. I have a PRI T1 on a T100P with six 7940 Cisco Phones w/SIP load 4.4. What hardware do u have? The worst part is that my system will sometimes just busy out even if I do not issue a reload command! However if I issue reload it's a sure thing * will hang. I spoke w/[EMAIL PROTECTED] about this last week but we never resolved it and my system has been hung 3 times in two weeks. Each time it hangs I killall -9 asterisk, cvs update restart. He said he was working on the releasing of a PRI channel code last week and that the Telco and * may not be jiving correctly. I had the problem w/the 7/8/2003 cvs so the issue probably still exists. I'll report any findings from asterisk -gc as soon as I find any on my side, I just don't want to take down * during business hours. So tonight I'll restart w/asterisk -gc try and get to the bottom of this w/you. TL Here is what he [EMAIL PROTECTED] recommended: Well if you use sip phones you might want to have sip debug turned on. But if you have many SIP phones then you're going to have lots of SIP messages. The zap show channels output is broken. It might show some false messages. Instead do zap show channel chann_no If the PRI Flag is Call then this channel hasn't been cleared properly. If it's empty than it's ok. If the channel is not serviced by your telco it's going to be in Restarting state. Also make sure then when you restart asterisk that all the 23 channels get restarted at the very begining. regards Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Marcus Adolfsson Sent: Wednesday, August 20, 2003 9:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] reload not working I upgraded to the latest CVS yesterday (and this morning again), and whenever I execute the reload command Asterisk seems to hang. While the current calls aren't dropped, no new calls can be made. The CLI isn't responding properly either. The only way to get going again is to exit the CLI and stop Asterisk and start again. Any comments? Thanks, Marcus ___ 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] Hardware question
Digium makes a 4 port card. It'd be hard to get 4 lines with quicknet hardware. Bartosz Jozwiak wrote: Hello, Again one more question about hardware. What could you suggest me to buy. I need hardware to connect let's say 4 analog lines. (FXO). This hardware should talk to Asterisk of course.. Thanks very much for some advices :) Bartek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VAD (silence suppression) on Asterisk
Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman
Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIBfor Chan_h323
you can download current release tarballs from openh323.org. I just put them up yesterday Steven Thomas wrote: Thanks - because of my ignorance using the CVS archive - could you please give me the full command - thanks. Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 05:03 PM Please respond to asterisk-users you can do cvs update -r v1_11_7 to get version 1.11.7 for openh323 Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:51 PM Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 04:53 PM Please respond to asterisk-users should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is Asterisk ready for real use?
Okay, I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our traditional PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. It is *not* an option to purchase a VoIP system package from Cisco, 3com, etc. Installers are getting an enormous premium for this now (rough estimate, 20 extensions $40K (!)). I am this close to committing to a solution based on Asterisk PBX, PoE LAN switches, and VoIP phones. I am absolutely sure it is the right *long term* solution, but I don't know if it is ready for reliable daily usage. I've literally read the last year's worth of posts to asterisk-users to get a feel for the situation. Since you don't see posts of the form installed it, just working, no problems very often, you could get the opinion that everyone has problems since that is what the mailing list is for. So, I would like to hear from those out there that have a system as I've described above and tell me if I'm insane to commit this direction or whether it makes sense. For those of you who have done it, how much time did it take you to get the system running smoothly? PS: In case it matters, we're extremely Linux capable (we use it for our file serving, networking, and we built our own custom ERP on perl and mySQL, we also do embedded Linux in custom military robot controllers). -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite Build 1059 problems
Does anyone have X-Lite build 1059 working fully with Asterisk? The GSM Codec works very well now but we have problems when using G711 in that when I setup a ping between the two sites and then watch the latency, it steadily increases and starts at about 150ms and goes up to 2500ms within about 20 seconds. I have not investigated fully but I guess that its sending ever increasing size packets. When we use the GSM codec its fine. When we have used the G711 codec in previous releases of X-Lite, it's been fine. Also we get one way speech in the following circumstances :- SNOM 200 running 1.16w 4941 Grandstream Budgetone running 1.0.3.77 Calls between SNOM and Budgetone are fine in both directions. Call from X-Lite to Asterisk voicemail = Fine Call from X-Lite to either SNOM or Budgetone = Fine Call from SNOM to X-Lite = Call is delivered and answered OK but there is no speech path in either direction. If its any help one of our guys is on the end of a slow link (150ms) and the symptoms are the same but there is a 1 second audio burst from the X-Lite to SNOM just as the call is answered. Call from Budgetone to X-Lite = Call is delivered and answered OK but one way speech path. Budgetone to X-Lite OK but X-Lite to Budgetone, no audio. I do have call traces from X-Lite if anyone is interested but I would like to know if any other X-Lite users were seeing the same type of problems. Rgds, Stuart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] reload not working
I have three Cisco 7960 phones/SIP 5.3 using two Wildcard X100Ps and IAX service from Nufone. It worked fine on my earlier installed CVS from 6/10. I have not noticed any random hangs, altough it has only been running for two days. Thanks, Marcus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Lieberman Sent: Wednesday, August 20, 2003 11:21 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] reload not working Martin - et all, I'm having the same issue. I have a PRI T1 on a T100P with six 7940 Cisco Phones w/SIP load 4.4. What hardware do u have? The worst part is that my system will sometimes just busy out even if I do not issue a reload command! However if I issue reload it's a sure thing * will hang. I spoke w/[EMAIL PROTECTED] about this last week but we never resolved it and my system has been hung 3 times in two weeks. Each time it hangs I killall -9 asterisk, cvs update restart. He said he was working on the releasing of a PRI channel code last week and that the Telco and * may not be jiving correctly. I had the problem w/the 7/8/2003 cvs so the issue probably still exists. I'll report any findings from asterisk -gc as soon as I find any on my side, I just don't want to take down * during business hours. So tonight I'll restart w/asterisk -gc try and get to the bottom of this w/you. TL Here is what he [EMAIL PROTECTED] recommended: Well if you use sip phones you might want to have sip debug turned on. But if you have many SIP phones then you're going to have lots of SIP messages. The zap show channels output is broken. It might show some false messages. Instead do zap show channel chann_no If the PRI Flag is Call then this channel hasn't been cleared properly. If it's empty than it's ok. If the channel is not serviced by your telco it's going to be in Restarting state. Also make sure then when you restart asterisk that all the 23 channels get restarted at the very begining. regards Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Marcus Adolfsson Sent: Wednesday, August 20, 2003 9:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] reload not working I upgraded to the latest CVS yesterday (and this morning again), and whenever I execute the reload command Asterisk seems to hang. While the current calls aren't dropped, no new calls can be made. The CLI isn't responding properly either. The only way to get going again is to exit the CLI and stop Asterisk and start again. Any comments? Thanks, Marcus ___ 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] Is Asterisk ready for real use?
At 10:42 AM 8/20/2003 -0500, you wrote: I've literally read the last year's worth of posts to asterisk-users to get a feel for the situation. Since you don't see posts of the form installed it, just working, no problems very often, you could get the opinion that everyone has problems since that is what the mailing list is for. So, I would like to hear from those out there that have a system as I've described above and tell me if I'm insane to commit this direction or whether it makes sense. For those of you who have done it, how much time did it take you to get the system running smoothly? I'm in almost the same situation as you. However, I'm mostly worried that the customer service desk here will start to complain that they can't tell how many calls are in the queue any more (our current phone tells us how many calls are ringing, on hold, etc). Regardless, I'm very interested to hear your results as well as what others on the list say, and would like to stay in touch with you if you decide to move forward with Asterisk. Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] reload not working
What is your SIP registration timeout? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Marcus Adolfsson Sent: Wednesday, August 20, 2003 11:50 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] reload not working I have three Cisco 7960 phones/SIP 5.3 using two Wildcard X100Ps and IAX service from Nufone. It worked fine on my earlier installed CVS from 6/10. I have not noticed any random hangs, altough it has only been running for two days. Thanks, Marcus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Lieberman Sent: Wednesday, August 20, 2003 11:21 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] reload not working Martin - et all, I'm having the same issue. I have a PRI T1 on a T100P with six 7940 Cisco Phones w/SIP load 4.4. What hardware do u have? The worst part is that my system will sometimes just busy out even if I do not issue a reload command! However if I issue reload it's a sure thing * will hang. I spoke w/[EMAIL PROTECTED] about this last week but we never resolved it and my system has been hung 3 times in two weeks. Each time it hangs I killall -9 asterisk, cvs update restart. He said he was working on the releasing of a PRI channel code last week and that the Telco and * may not be jiving correctly. I had the problem w/the 7/8/2003 cvs so the issue probably still exists. I'll report any findings from asterisk -gc as soon as I find any on my side, I just don't want to take down * during business hours. So tonight I'll restart w/asterisk -gc try and get to the bottom of this w/you. TL Here is what he [EMAIL PROTECTED] recommended: Well if you use sip phones you might want to have sip debug turned on. But if you have many SIP phones then you're going to have lots of SIP messages. The zap show channels output is broken. It might show some false messages. Instead do zap show channel chann_no If the PRI Flag is Call then this channel hasn't been cleared properly. If it's empty than it's ok. If the channel is not serviced by your telco it's going to be in Restarting state. Also make sure then when you restart asterisk that all the 23 channels get restarted at the very begining. regards Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Marcus Adolfsson Sent: Wednesday, August 20, 2003 9:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] reload not working I upgraded to the latest CVS yesterday (and this morning again), and whenever I execute the reload command Asterisk seems to hang. While the current calls aren't dropped, no new calls can be made. The CLI isn't responding properly either. The only way to get going again is to exit the CLI and stop Asterisk and start again. Any comments? Thanks, Marcus ___ 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] Is Asterisk ready for real use?
Okay, I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our traditional PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. It is *not* an option to purchase a VoIP system package from Cisco, 3com, etc. Installers are getting an enormous premium for this now (rough estimate, 20 extensions $40K (!)). I am this close to committing to a solution based on Asterisk PBX, PoE LAN switches, and VoIP phones. I am absolutely sure it is the right *long term* solution, but I don't know if it is ready for reliable daily usage. I've literally read the last year's worth of posts to asterisk-users to get a feel for the situation. Since you don't see posts of the form installed it, just working, no problems very often, you could get the opinion that everyone has problems since that is what the mailing list is for. So, I would like to hear from those out there that have a system as I've described above and tell me if I'm insane to commit this direction or whether it makes sense. For those of you who have done it, how much time did it take you to get the system running smoothly? PS: In case it matters, we're extremely Linux capable (we use it for our file serving, networking, and we built our own custom ERP on perl and mySQL, we also do embedded Linux in custom military robot controllers). -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com Basically if you are as Linux capable as you say then Asterisk is most definately ready for real use in your situation.. I am nowhere near the all the Linux skills you mentioned and I am running 2 small Asterisk PBX's without too much hassle.. I have one with 3 extentions and the other currently with 5 extensions and counting... both are connected together via an IAX trunk.. As for the time frame.. you will really only need time to get your head around the config syntax and concepts.. after that you could setup a fresh PBX in a few hours.. but you will constantly be adding to it and tuning it.. :) So I would honestly say go for Asterisk... Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk ready for real use?
astman or gastman would tell you this info. And yes we us it in production right now. Works better than anything we have had previously. bkw On Wed, 20 Aug 2003, Ernest W. Lessenger wrote: At 10:42 AM 8/20/2003 -0500, you wrote: I've literally read the last year's worth of posts to asterisk-users to get a feel for the situation. Since you don't see posts of the form installed it, just working, no problems very often, you could get the opinion that everyone has problems since that is what the mailing list is for. So, I would like to hear from those out there that have a system as I've described above and tell me if I'm insane to commit this direction or whether it makes sense. For those of you who have done it, how much time did it take you to get the system running smoothly? I'm in almost the same situation as you. However, I'm mostly worried that the customer service desk here will start to complain that they can't tell how many calls are in the queue any more (our current phone tells us how many calls are ringing, on hold, etc). Regardless, I'm very interested to hear your results as well as what others on the list say, and would like to stay in touch with you if you decide to move forward with Asterisk. Thanks, --Ernest ___ 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] reload not working
Brian, == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Not found (No such file or directory) == Parsing '/etc/asterisk/rtp.conf': Not found (No such file or directory) == RTP Allocating from port range 5000 - 31000 -- Reloading module 'cdr_mysql.so' (MySQL CDR Backend) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so So in modem.conf I commented out: ; Modem Drivers to load ;driver=aopen Seems to work fine now! Thanks for your help. Marcus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, August 20, 2003 9:23 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] reload not working yes start it with asterisk -gc watch and see what the error is. bkw On Wed, 20 Aug 2003, Marcus Adolfsson wrote: I upgraded to the latest CVS yesterday (and this morning again), and whenever I execute the reload command Asterisk seems to hang. While the current calls aren't dropped, no new calls can be made. The CLI isn't responding properly either. The only way to get going again is to exit the CLI and stop Asterisk and start again. Any comments? Thanks, Marcus ___ 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] VAD (silence suppression) on Asterisk
Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman Somthing tells me that it is not supported but it is somthing that I would like to see supported as well.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI Question
Can you do remote loopup from your switches side ... and look asterisk's T1 and check if your transmission is ok ? regards Martin On Tue, 19 Aug 2003, Barry Porch wrote: Martin, Here is the trace you asked for. It's quite lengthy so I'm attaching it as a text file. The way I generated this output was to start up an instance of asterisk redirecting output to a text file. Then I connected in another terminal window as console and issued the debug command. I don't know if there's a better way to do this. Also I can successfully connect to the PRI port on my PBX with my T1/PRI tester (a Sunset T10) and I can also successfully connect to my Asterisk box and place an inbound call. I just can't connect the Asterisk box directly to my PBX. The PBX is a Mitel 3300, by the way. Thanks for your help! Barry -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: Monday, August 18, 2003 9:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PRI Question First of all you should have callprogress=no and immediate=no for any kind of a PRI. Also why is your d-channel going down ? Can you send a trace pri intense debug span 1 ? regards Martin On Mon, 18 Aug 2003, Barry Porch wrote: I managed to get Asterisk working with my PBX using T1, now I am moving on to trying to make PRI work. I have my zaptel.conf and zapata.conf configured as follows: Zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us Zapata.conf: [channels] transfer=yes immediate=yes callprogress=yes language=en context=default switchtype=national signalling=pri_net group=1 channel=1-23 I have a PRI port provisioned off my PBX to work with the above settings. When I start up Asterisk everything seems to come up (my PBX sees a D-channel) but I get constant output as follows: == D-Channel on span 1 up B-channel 2 restarted on span 1 B-channel 3 restarted on span 1 B-channel 4 restarted on span 1 B-channel 5 restarted on span 1 B-channel 6 restarted on span 1 B-channel 7 restarted on span 1 B-channel 8 restarted on span 1 B-channel 9 restarted on span 1 B-channel 10 restarted on span 1 B-channel 11 restarted on span 1 B-channel 12 restarted on span 1 B-channel 13 restarted on span 1 B-channel 14 restarted on span 1 B-channel 16 restarted on span 1 D-Channel on span 1 down D-Channel on span 1 up B-channel 2 successfully restarted on span 1 B-channel 3 successfully restarted on span 1 B-channel 4 successfully restarted on span 1 B-channel 5 successfully restarted on span 1 And so forth with the D-channel going up and down and various channels successfully restarting but inbound calls do not work. I have tried changing the PRI protocol, I have swapped the CPE and NET settings between the PBX and Asterisk and every combination gives me this same result. Any thoughts? 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
[Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway (SIP)
Is anyone out there using an AudioCodes MP108 8-Port FXO Analog Gateway (SIP) with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion before purchasing. Incidentally, if this works well it could be an option for those of you who have recently asked about putting multiple analog FXO ports into an asterisk server. Thanks, --Ernest W. Lessenger OACYS Technology ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VAD (silence suppression) on Asterisk
VAD is evil. I hate it. I find when we used it.. you keep asking people to repeat stuff all the time.. and it was just anoying. bkw On Wed, 20 Aug 2003, WipeOut . wrote: Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman Somthing tells me that it is not supported but it is somthing that I would like to see supported as well.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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 IAX trunking... DP cache?
I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like this: Internet | --- | | | Public PBX | NAT Firewall | || | Private PBX Phone Phone ... This way we can run private Asterisk PBXes behind NATting firewalls that register with a central public Asterisk PBX using IAX/IAX2. The phones can merrily run SIP to the local Private PBX without worrying about NAT headaches or outbound proxies. I've managed to get the PBXes to mutually register correctly: Public PBX: *CLI iax show users Username Secret Authen Def.Context A/C privatepbxrsadefault No Private PBX: *CLI iax show registry Host Username PerceivedRefresh State x.x.x.x:5036 privatepbx x.x.x.x:5036 60 Registered *CLI iax2 show registry Host Username PerceivedRefresh State x.x.x.x:4569 privatepbx x.x.x.x:4569 60 Registered On the Public PBX, I added: iax.conf: [privatepbx] type=friend host=dynamic ;trunk=yes ;-- doesn't work without a zap interface for timing context=outgoing auth=rsa inkeys=privatepbx outkeys=publicpbx qualify=yes extension.conf: [outgoing] include = iaxtel; IAXTEL include = fwd ; fwd.pulver.com include = iptel ; iptel.org include = sipphone ; SIPPhone.com include = commx ; CommunicationsXchange.com On the Private PBX, I've added what I think are the appropriate sections: iax.conf: [general] register = privatepbx:[EMAIL PROTECTED] extension.conf: [outgoing] switch = IAX/privatepbx:[EMAIL PROTECTED]/outgoing ;exten = s,1,Dial(IAX/privatepbx:[EMAIL PROTECTED]/outgoing,20,tr) Whenever I try to route an outbound call, however, I get the following errors on the private PBX: WARNING[15376]: File chan_iax.c, Line 4837 (find_cache): Unable to generate call for 'privatepbx:[EMAIL PROTECTED]/outgoing' WARNING[15376]: File chan_iax.c, Line 4957 (iax_exists): Unable to make DP cache WARNING[15376]: File chan_iax.c, Line 4837 (find_cache): Unable to generate call for 'privatepbx:[EMAIL PROTECTED]/outgoing' WARNING[15376]: File chan_iax.c, Line 4979 (iax_canmatch): Unable to make DP cache No IAX traffic appears to go out in response to this.. so I'm guessing I have another problem. Can anyone help point me in the right direction? -- - Ian C. Blenke [EMAIL PROTECTED] (This message bound by the following: http://www.nks.net/email_disclaimer.html) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX IAX trunking... DP cache?
I would use the latest CVS for one. And try again. bkw On Wed, 20 Aug 2003, Ian Blenke wrote: I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like this: Internet | --- | | | Public PBX | NAT Firewall | || | Private PBX Phone Phone ... This way we can run private Asterisk PBXes behind NATting firewalls that register with a central public Asterisk PBX using IAX/IAX2. The phones can merrily run SIP to the local Private PBX without worrying about NAT headaches or outbound proxies. I've managed to get the PBXes to mutually register correctly: Public PBX: *CLI iax show users Username Secret Authen Def.Context A/C privatepbxrsadefault No Private PBX: *CLI iax show registry Host Username PerceivedRefresh State x.x.x.x:5036 privatepbx x.x.x.x:5036 60 Registered *CLI iax2 show registry Host Username PerceivedRefresh State x.x.x.x:4569 privatepbx x.x.x.x:4569 60 Registered On the Public PBX, I added: iax.conf: [privatepbx] type=friend host=dynamic ;trunk=yes ;-- doesn't work without a zap interface for timing context=outgoing auth=rsa inkeys=privatepbx outkeys=publicpbx qualify=yes extension.conf: [outgoing] include = iaxtel ; IAXTEL include = fwd ; fwd.pulver.com include = iptel; iptel.org include = sipphone ; SIPPhone.com include = commx; CommunicationsXchange.com On the Private PBX, I've added what I think are the appropriate sections: iax.conf: [general] register = privatepbx:[EMAIL PROTECTED] extension.conf: [outgoing] switch = IAX/privatepbx:[EMAIL PROTECTED]/outgoing ;exten = s,1,Dial(IAX/privatepbx:[EMAIL PROTECTED]/outgoing,20,tr) Whenever I try to route an outbound call, however, I get the following errors on the private PBX: WARNING[15376]: File chan_iax.c, Line 4837 (find_cache): Unable to generate call for 'privatepbx:[EMAIL PROTECTED]/outgoing' WARNING[15376]: File chan_iax.c, Line 4957 (iax_exists): Unable to make DP cache WARNING[15376]: File chan_iax.c, Line 4837 (find_cache): Unable to generate call for 'privatepbx:[EMAIL PROTECTED]/outgoing' WARNING[15376]: File chan_iax.c, Line 4979 (iax_canmatch): Unable to make DP cache No IAX traffic appears to go out in response to this.. so I'm guessing I have another problem. Can anyone help point me in the right direction? -- - Ian C. Blenke [EMAIL PROTECTED] (This message bound by the following: http://www.nks.net/email_disclaimer.html) ___ 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] Where to find correct ver of OpenH323 PWLIBfor Chan_h323
I always keep known working code and libs at http://www.nufone.net/downloads Jeremy McNamara Steven Thomas wrote: Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ 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] Is Asterisk ready for real use?
Or you can jump on #asterisk bkw On Wed, 20 Aug 2003, John Brown wrote: We are getting ready to replace our old Panasonic PBX with an Asterisk system. I'd say its ready for prime time. THe other thing is to have a good consultant in your back pocket for those now how do I do this. I can strongly recommend John Todd for that role. His email is in the archives. On Wed, Aug 20, 2003 at 11:08:35AM -0500, Mike Ciholas wrote: On Wed, 20 Aug 2003, Ernest W. Lessenger wrote: At 10:42 AM 8/20/2003 -0500, you wrote: So, I would like to hear from those out there that have a system as I've described above and tell me if I'm insane to commit this direction or whether it makes sense. I'm in almost the same situation as you. However, I'm mostly worried that the customer service desk here will start to complain that they can't tell how many calls are in the queue any more (our current phone tells us how many calls are ringing, on hold, etc). Regardless, I'm very interested to hear your results as well as what others on the list say, and would like to stay in touch with you if you decide to move forward with Asterisk. Needs like yours are probably why you *should* choose Asterisk. Sounds like there outta be a way to do that. Right off, I can see keeping queue length counters and displaying them on the phone, or perhaps having them displayed on a dynamic web page visible to all. You could imagine doing things as sophisticated as your expected wait time is X, when the queue grows over X size, call other extensions, etc. I am very intrigued by the flexibility Asterisk offers, but I need to know that I can reliably just make calls at first. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ 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] Is Asterisk ready for real use?
On Wed, 20 Aug 2003, Mike Ciholas wrote: I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our traditional PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. Hedge your bets, pull two cables, and try asterisk. -- 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
Re: [Asterisk-Users] IAX IAX trunking... DP cache?
Brian West wrote: I would use the latest CVS for one. And try again. Unfortunately, I've tried numerous times to get a current CVS trunk snapshot to talk to *anything*, to no avail. Even getting my Grandstream phones to register with it was an apparent excersize in futility. Dropping back to 0.4.0 *immediately* worked with the same configs. I'll give it a go again with today's snapshot and see if I can get *anything* to work again. Is there any hope for a 0.5.0 release on the horizon? -- - Ian C. Blenke [EMAIL PROTECTED] (This message bound by the following: http://www.nks.net/email_disclaimer.html) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adtran TA 750
Hello, Does somebody knows how to connect Adtran Total Access to Asterisk, is it with T1 ? bart
Re: [Asterisk-Users] Is Asterisk ready for real use?
Mike Ciholas wrote: Okay, I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our traditional PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. It is *not* an option to purchase a VoIP system package from Cisco, 3com, etc. Installers are getting an enormous premium for this now (rough estimate, 20 extensions $40K (!)). I am this close to committing to a solution based on Asterisk PBX, PoE LAN switches, and VoIP phones. I am absolutely sure it is the right *long term* solution, but I don't know if it is ready for reliable daily usage. I've literally read the last year's worth of posts to asterisk-users to get a feel for the situation. Since you don't see posts of the form installed it, just working, no problems very often, you could get the opinion that everyone has problems since that is what the mailing list is for. So, I would like to hear from those out there that have a system as I've described above and tell me if I'm insane to commit this direction or whether it makes sense. For those of you who have done it, how much time did it take you to get the system running smoothly? PS: In case it matters, we're extremely Linux capable (we use it for our file serving, networking, and we built our own custom ERP on perl and mySQL, we also do embedded Linux in custom military robot controllers). For me it is ready for heavy use. I allready using it for 1 call center (call queues ...) and for the offise PBX. Now i'm waiting some hardware (channel banks ) to test it with 100+ lines (1 E1 Trunks and analog lines). Only thing that is, if you're are begginer asterisk user, you will need some more time to get whole picture and the features (3-4 days googling and reading mailing list archives) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VAD (silence suppression) on Asterisk
When I turn on VAD on cisco ATA186, asterisk shows: Aug 20 11:12:20 NOTICE[618513]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible RCF3389 defines Payload for Comfort Noise, that is used with VAD. So I turned it off on my endpoints (ATA186 and c827-4v) Eduardo On Wed, 20 Aug 2003 11:35:14 -0500 (CDT) Brian West [EMAIL PROTECTED] wrote: VAD is evil. I hate it. I find when we used it.. you keep asking people to repeat stuff all the time.. and it was just anoying. bkw On Wed, 20 Aug 2003, WipeOut . wrote: Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman Somthing tells me that it is not supported but it is somthing that I would like to see supported as well.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] G723 (was SIP using which codec?)
And if one cannot use a different codec? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, August 20, 2003 9:51 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G723 (was SIP using which codec?) MOH requires that Asterisk transcodes (It also has to transcode to for PSTN calls and voicemail and playing any sound files). Asterisk can't transcode to or from G723. Nope. Doesn't work. May very well never work. Use a different codec. On Wed, 2003-08-20 at 03:47, Andrew Joakimsen wrote: Actually I got it working right before I gave up (I had the wrong line in my config commented out) But now I get these messages when I try to playback a recording: NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable to find a path from GSM to G723 WARNING[16401]: File file.c, Line 722 (ast_streamfile): Unable to open transfer (format G723): No such file or directory WARNING[16401]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on SIP/packet8.net-dab9 for transfer And when I try to play music on hold: NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable to find a path from SLINR to G723 WARNING[16401]: File res_musiconhold.c, Line 421 (moh_alloc): Unable to set 'SIP/packet8.net-dab9' to signed linear format This is the missing link in my system, I greatly appreciate any help that can be provided. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Wednesday, August 20, 2003 4:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP using which codec? At 7:27 AM + 8/20/03, WipeOut . wrote: Is there a way to determine what codec the remote server wants to use in a SIP session for an incoming call by looking at something, possiby sip debug? Take a look in the archives this was covered a couple of days ago.. the command you are looking for is sip show channels.. and then look in the format column.. the formula for determining the format was posted in the previous discussion and i can't rememebr it off the top of my head.. Later.. I think that the question is a bit more subtle than that. The question says wants to use, not does use. Currently, I think the only way you'll find this is with a SIP debug, looking at the SDP request. JT ___ 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 -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ 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] Is Asterisk ready for real use?
I'm in almost the same situation as you. However, I'm mostly worried that the customer service desk here will start to complain that they can't tell how many calls are in the queue any more (our current phone tells us how many calls are ringing, on hold, etc). yea we ran into that as well, we used the manager interface to write the info out to a Matrix Orbital LCD display devices so the Customer Service reps see that info ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323.c
On Mon, 18 Aug 2003, Mark Spencer wrote:
It's up one directly. It just moved.
Run make in h323 then do make install on asterisk again.
On Mon, 18 Aug 2003, John Fortman wrote:
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp,
ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was
not created so no h323 support in asterisk.
Just wondering when to expect it again because I was stupid and didn't
make a backup of the asterisk code before wiping the directory for a
rebuild.
Has anyone gotten H.323 channel complied on Redhat 9.0? Every time I try,
I get a ton of errors from ptlib. I'm about ready to punt this sucker out
the door. I really like what I have seen out of asterisk so far...
Example of errors:
In file included from /usr/include/ptlib/contain.h:218,
from /usr/include/ptlib.h:137,
from ast_h323.h:29,
from ast_h323.cpp:27:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
PObject::BOOL'
/usr/include/ptlib/object.h:1214: parse error before `(' token
/usr/include/ptlib/object.h:1265: syntax error before `operator'
/usr/include/ptlib/object.h:1214: duplicate member `PObject::BOOL'
/usr/include/ptlib/object.h:1274: ISO C++ forbids defining types within
return
type
Any help would be appriciated, even if it's a recommendation to another
flavor of linux.
Thanks
Sean
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Re: [Asterisk-Users] IAX IAX trunking... DP cache?
On Wed, 2003-08-20 at 11:09, Ian Blenke wrote: Brian West wrote: I would use the latest CVS for one. And try again. Unfortunately, I've tried numerous times to get a current CVS trunk snapshot to talk to *anything*, to no avail. Even getting my Grandstream phones to register with it was an apparent excersize in futility. Dropping back to 0.4.0 *immediately* worked with the same configs. I'll give it a go again with today's snapshot and see if I can get *anything* to work again. Is there any hope for a 0.5.0 release on the horizon? I would also like to see a more structured release program. It's kind of scary to tell people that they should just use the latest CVS code. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk ready for real use?
On Wed, Aug 20, 2003 at 12:13:07PM -0500, Dave Weis wrote: On Wed, 20 Aug 2003, Mike Ciholas wrote: I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our traditional PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. Hedge your bets, pull two cables, and try asterisk. I always run an all Cat5 network for voice and data. I usually try to pull twice as many strands of Cat5 to a location as I expect to have devices plugged in. That way, you can just patch your voice or data circuits to anywhere. Moving an extension is just a patch cable change and move the phone to the new jack. In the companies I have worked with, the only constant in layout of users was the state of flux. RJ11 plugs work in RJ45 jacks most of the time. If you have instability, crimp an RJ45 on you phone cord. -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI Question
My switch doesn't let me set up a loop but I am confident that everything is OK at the T1 layer. I can connect via robbed bit to the Asterisk box with no problem. Also I can use my T1/PRI tester towards either systems and it works fine with PRI and I can place and receive calls. There seems to be some incompatibility between Mitel's PRI implementation and Asterisk. I have tried all supported flavors of PRI (ni2, dms100, 4ess, 5ess) with the same result. I have also tried changing pri_cpe vs pri_net between the 2 machines. Everything gives me the same result with the d channel going down and coming back up constantly. I have been running on a cvs from about 2 weeks ago so I also did a cvs update yesterday with no change. It appears that I may be out of luck connecting the Mitel directly to the Asterisk box via PRI. If someone has other experience I'd love to hear it. Has anyone else run into PRI implementations that they cannot make work with Asterisk? Barry -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 12:32 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] PRI Question Can you do remote loopup from your switches side ... and look asterisk's T1 and check if your transmission is ok ? regards Martin On Tue, 19 Aug 2003, Barry Porch wrote: Martin, Here is the trace you asked for. It's quite lengthy so I'm attaching it as a text file. The way I generated this output was to start up an instance of asterisk redirecting output to a text file. Then I connected in another terminal window as console and issued the debug command. I don't know if there's a better way to do this. Also I can successfully connect to the PRI port on my PBX with my T1/PRI tester (a Sunset T10) and I can also successfully connect to my Asterisk box and place an inbound call. I just can't connect the Asterisk box directly to my PBX. The PBX is a Mitel 3300, by the way. Thanks for your help! Barry -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: Monday, August 18, 2003 9:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PRI Question First of all you should have callprogress=no and immediate=no for any kind of a PRI. Also why is your d-channel going down ? Can you send a trace pri intense debug span 1 ? regards Martin On Mon, 18 Aug 2003, Barry Porch wrote: I managed to get Asterisk working with my PBX using T1, now I am moving on to trying to make PRI work. I have my zaptel.conf and zapata.conf configured as follows: Zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us Zapata.conf: [channels] transfer=yes immediate=yes callprogress=yes language=en context=default switchtype=national signalling=pri_net group=1 channel=1-23 I have a PRI port provisioned off my PBX to work with the above settings. When I start up Asterisk everything seems to come up (my PBX sees a D-channel) but I get constant output as follows: == D-Channel on span 1 up B-channel 2 restarted on span 1 B-channel 3 restarted on span 1 B-channel 4 restarted on span 1 B-channel 5 restarted on span 1 B-channel 6 restarted on span 1 B-channel 7 restarted on span 1 B-channel 8 restarted on span 1 B-channel 9 restarted on span 1 B-channel 10 restarted on span 1 B-channel 11 restarted on span 1 B-channel 12 restarted on span 1 B-channel 13 restarted on span 1 B-channel 14 restarted on span 1 B-channel 16 restarted on span 1 D-Channel on span 1 down D-Channel on span 1 up B-channel 2 successfully restarted on span 1 B-channel 3 successfully restarted on span 1 B-channel 4 successfully restarted on span 1 B-channel 5 successfully restarted on span 1 And so forth with the D-channel going up and down and various channels successfully restarting but inbound calls do not work. I have tried changing the PRI protocol, I have swapped the CPE and NET settings between the PBX and Asterisk and every combination gives me this same result. Any thoughts? 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G723 (was SIP using which codec?)
If you want to be able to use G723 from a legal standpoint you will have to license the codec from the current patent holders. The patent holder's price list can be found at http://www.dspg.com/technology/LicensePricing.html If you obtain a license to use G723 then Digium or the Asterisk user community might be able to assist you in adding the codec to Asterisk. I don't know how this would work from the technical standpoint. For G729, Voiceage (the patent holder for G729) provides a binary module with some horrible license agreement to add to Asterisk. You can buy G729 licenses direct from Digium for $10/channel. If you can't afford the US$30,000+ licensing fees for G723 and if you can't use G729 or any other codec then you really are out of luck. There are many people that want to use G723 (myself included), but I'm not going to spend that kind of money for four G729 channels. I'd be happy to pay $10/channel just like I have for the G729 license. On Wed, 2003-08-20 at 12:28, Andrew Joakimsen wrote: And if one cannot use a different codec? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, August 20, 2003 9:51 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G723 (was SIP using which codec?) MOH requires that Asterisk transcodes (It also has to transcode to for PSTN calls and voicemail and playing any sound files). Asterisk can't transcode to or from G723. Nope. Doesn't work. May very well never work. Use a different codec. On Wed, 2003-08-20 at 03:47, Andrew Joakimsen wrote: Actually I got it working right before I gave up (I had the wrong line in my config commented out) But now I get these messages when I try to playback a recording: NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable to find a path from GSM to G723 WARNING[16401]: File file.c, Line 722 (ast_streamfile): Unable to open transfer (format G723): No such file or directory WARNING[16401]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on SIP/packet8.net-dab9 for transfer And when I try to play music on hold: NOTICE[16401]: File channel.c, Line 1406 (ast_set_write_format): Unable to find a path from SLINR to G723 WARNING[16401]: File res_musiconhold.c, Line 421 (moh_alloc): Unable to set 'SIP/packet8.net-dab9' to signed linear format This is the missing link in my system, I greatly appreciate any help that can be provided. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Wednesday, August 20, 2003 4:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP using which codec? At 7:27 AM + 8/20/03, WipeOut . wrote: Is there a way to determine what codec the remote server wants to use in a SIP session for an incoming call by looking at something, possiby sip debug? Take a look in the archives this was covered a couple of days ago.. the command you are looking for is sip show channels.. and then look in the format column.. the formula for determining the format was posted in the previous discussion and i can't rememebr it off the top of my head.. Later.. I think that the question is a bit more subtle than that. The question says wants to use, not does use. Currently, I think the only way you'll find this is with a SIP debug, looking at the SDP request. JT ___ 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 -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange happenings
Just idly watching * in console mode and saw that someone from 50.49.54.102 tried to register with my *. whois gives:- OrgName:Internet Assigned Numbers Authority OrgID: IANA Address:4676 Admiralty Way, Suite 330 City: Marina del Rey StateProv: CA PostalCode: 90292-6695 Country:US NetRange: 50.0.0.0 - 50.255.255.255 CIDR: 50.0.0.0/8 NetName:RESERVED-50 NetHandle: NET-50-0-0-0-0 Parent: NetType:IANA Reserved Comment: RegDate: Updated:2002-08-23 OrgTechHandle: IANA-ARIN OrgTechName: Internet Corporation for Assigned Names and Number OrgTechPhone: +1-310-823-9358 OrgTechEmail: [EMAIL PROTECTED] ? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk ready for real use?
As for cables. Pull ONLY Cat5 or Cat5e as they can be used for either Ethernet OR voice. You can then use a plug pannel in the phone closet to route a spicif cable to either a voice or data switch. Is Asterick ready?? I'd say the software is but ONLY IF 1) You or someone you can depend on knows how to support it. 2) You have a test system on which to experiment and test out any changes/updates and that can serve as a hot backup The isues you will run into when deploying * are the same as with any other mission critical system. You want to do a LOT of testing and have backup plans for various kinds of failures. I'd certainly NOT want to install * on a one wheek schedule if I was just starting to look at it now. You would need to hire someonr who is already up to speed on it --- Anton Tinchev [EMAIL PROTECTED] wrote: Mike Ciholas wrote: Okay, I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our traditional PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX IAX trunking... DP cache?
I would also like to see a more structured release program. It's kind of scary to tell people that they should just use the latest CVS code. That's where consultants earn their money. They should be preforming some kind of quality control. You build the code, get it to work, test it and ONLY then install it at a customer's site for final testing. If you don't have a consultent then you do this kind of work yourself. You are right. It would be stupid to install a new untested CVS download on a working PBX system. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI Question
This may have nothing to do with it but have you verified your timing? Make sure one end of the T1 is using an internal clock and the other end is using timing off of the T1. Don Pobanz On Wednesday, August 20, 2003 12:37 PM, Barry Porch [SMTP:[EMAIL PROTECTED] wrote: My switch doesn't let me set up a loop but I am confident that everything is OK at the T1 layer. I can connect via robbed bit to the Asterisk box with no problem. Also I can use my T1/PRI tester towards either systems and it works fine with PRI and I can place and receive calls. There seems to be some incompatibility between Mitel's PRI implementation and Asterisk. I have tried all supported flavors of PRI (ni2, dms100, 4ess, 5ess) with the same result. I have also tried changing pri_cpe vs pri_net between the 2 machines. Everything gives me the same result with the d channel going down and coming back up constantly. I have been running on a cvs from about 2 weeks ago so I also did a cvs update yesterday with no change. It appears that I may be out of luck connecting the Mitel directly to the Asterisk box via PRI. If someone has other experience I'd love to hear it. Has anyone else run into PRI implementations that they cannot make work with Asterisk? Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VAD (silence suppression) on Asterisk
Thanks, that's the answer I was looking for. Do we know if VAD will ever be supported? I know some people don't like VAD and in my testing, how well VAD works depends on how well it was coded (and the hardware I suspect). I have seen very good and very bad implementations of VAD. I have a real need for VAD to work (for bandwidth reasons). So if VAD isn't currently in the Asterisk development schedule, can I request it be added? Thanks Lee Goodman - Original Message - From: Eduardo Goncalves [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 1:21 PM Subject: Re: [Asterisk-Users] VAD (silence suppression) on Asterisk When I turn on VAD on cisco ATA186, asterisk shows: Aug 20 11:12:20 NOTICE[618513]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible RCF3389 defines Payload for Comfort Noise, that is used with VAD. So I turned it off on my endpoints (ATA186 and c827-4v) Eduardo On Wed, 20 Aug 2003 11:35:14 -0500 (CDT) Brian West [EMAIL PROTECTED] wrote: VAD is evil. I hate it. I find when we used it.. you keep asking people to repeat stuff all the time.. and it was just anoying. bkw On Wed, 20 Aug 2003, WipeOut . wrote: Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman Somthing tells me that it is not supported but it is somthing that I would like to see supported as well.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] IAX to zaptel echo
Title: Message Hi all, I am experiencing a problem with the quality of the voice communication between an IAX based softphone (WinIAX) and an outside line through a FXO port or even with a regular analog phone connected to a FXS port. The party using the IAX softphone hears his own echo a plit of a second after speaking. The party on the analog end does not experience any echo. I tried to modify the KFLAG parameter in the Makefile of zaptel to change the algorithm without results. I also tried to modify the value of the echocancel paramter in zapata.conf (tried 32,64,yes/128,256) without success. Note the parameter echocancelwhenbridged is set to yes. On the other hand, when I establish a communication between two IAX softphones, I hear no echo. This leads me to suspect the cards I am using. I have one X100P and one TDM400P card with 2 modules enabled. I would like to know if anyone encoutered this problem and how to minimize/eliminate the echo, if possible. Also, we will eventually be using a PRI with a T100P or a T400P, would the echo problem still be present? Regards, Claude Klimos
Re: [Asterisk-Users] chan_h323.c
I have been using Slackware 9 with the sources for ptlib_unix, pwlib_unix,
openh323, asterisk, zaptel and libpri in /root/src
1) load all 6 packages (+ ohphone if you need an h323 client) from cvs to
/root/src
2) /root/src/pwlib: configure, make, make install, ldconfig (not all that
sure why, but Slackware requires ldconfig to be run)
3) /root/src/openh323: configure, make, make install, ldconfig
4) /root/src/zaptel: make, make install (reload and reconfigure your zaptel
card)
5) /root/src/libpri: make, make install (I don't have a PRI card so I don't
do anything here)
6) /root/src/asterisk/channels/h323:
- edit Makfile
- set PWLIBDIR = $(HOME)/src/pwlib
- set OPENH323DIR = $(HOME)/src/openh323
- make, make install (installs openh323.a) (make samples if you do not
have h323.conf in /etc/asterisk when done)
7) /root/src/asterisk: make, make install, make samples
8) asterisk -vvvc
- the last section should load chan_h323
I haven't had any problems compiling this from CVS for almost a month on at
least three different systems with some version of Slackware. I have had
problems with other things like transferring calls but that's a different
issue.
John.
- Original Message -
From: Sean Figgins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 20, 2003 12:22 PM
Subject: Re: [Asterisk-Users] chan_h323.c
On Mon, 18 Aug 2003, Mark Spencer wrote:
It's up one directly. It just moved.
Run make in h323 then do make install on asterisk again.
On Mon, 18 Aug 2003, John Fortman wrote:
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp,
ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was
not created so no h323 support in asterisk.
Just wondering when to expect it again because I was stupid and didn't
make a backup of the asterisk code before wiping the directory for a
rebuild.
Has anyone gotten H.323 channel complied on Redhat 9.0? Every time I try,
I get a ton of errors from ptlib. I'm about ready to punt this sucker out
the door. I really like what I have seen out of asterisk so far...
Example of errors:
In file included from /usr/include/ptlib/contain.h:218,
from /usr/include/ptlib.h:137,
from ast_h323.h:29,
from ast_h323.cpp:27:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
PObject::BOOL'
/usr/include/ptlib/object.h:1214: parse error before `(' token
/usr/include/ptlib/object.h:1265: syntax error before `operator'
/usr/include/ptlib/object.h:1214: duplicate member `PObject::BOOL'
/usr/include/ptlib/object.h:1274: ISO C++ forbids defining types within
return
type
Any help would be appriciated, even if it's a recommendation to another
flavor of linux.
Thanks
Sean
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Re: [Asterisk-Users] IAX IAX trunking... DP cache?
Chris Albertson wrote: I would also like to see a more structured release program. It's kind of scary to tell people that they should just use the latest CVS code. For testing and development, this isn't a bad thing - as long as the trunk codebase generally *compiles* and *runs* more often than not. I've had no problems compiling the latest snapshots, but running it seems to lead me down a path of frustration - nothing seemed to work as the 0.4.0 release did. For production, a stable release cycle really would be nice. Particularly with a patch history along the stable tree until a next release (0.4.1, 0.4.2, etc, until the 0.5.0 development tree is deemed adequate for a feature freeze to 0.5.0). That's where consultants earn their money. They should be preforming some kind of quality control. You build the code, get it to work, test it and ONLY then install it at a customer's site for final testing. If you don't have a consultent then you do this kind of work yourself. In the OpenSource world, consultants typically implement stable releases of packages - typically for bug reporting if for no other reason. It's difficult to explain to someone why your CVS checkout from 3 months ago exhibits some unexpected flaw - who out there is going to have the same snapshot installed to compare notes? You are right. It would be stupid to install a new untested CVS download on a working PBX system. Granted. It would also be stupid to dictate a code-fork of the Asterisk source base just to have a stable reference tree. I'm partial to the Mozilla approach. New development in trunk, branch stable releases with fixes rolled back into trunk as appropriate. How many folks are running 2.6.0preX series Linux kernels on their production servers? This isn't much different, really. I don't mind rolling in a half dozen backports to patch up a 2.4 tree to something I can use, and *depend* on. In the end, I'm really just happy to have something of Asterisk's quality in the OpenSource community. Despite a few minor quirks, 0.4.0 really does seem to work quite well. -- - Ian C. Blenke [EMAIL PROTECTED] (This message bound by the following: http://www.nks.net/email_disclaimer.html) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VAD (silence suppression) on Asterisk
Comfort Noise and VAD are diffrent things. bkw On Wed, 20 Aug 2003, Eduardo Goncalves wrote: When I turn on VAD on cisco ATA186, asterisk shows: Aug 20 11:12:20 NOTICE[618513]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible RCF3389 defines Payload for Comfort Noise, that is used with VAD. So I turned it off on my endpoints (ATA186 and c827-4v) Eduardo On Wed, 20 Aug 2003 11:35:14 -0500 (CDT) Brian West [EMAIL PROTECTED] wrote: VAD is evil. I hate it. I find when we used it.. you keep asking people to repeat stuff all the time.. and it was just anoying. bkw On Wed, 20 Aug 2003, WipeOut . wrote: Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman Somthing tells me that it is not supported but it is somthing that I would like to see supported as well.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] chan_h323.c
Great! Thanks for the recommendation. I'll beat on Redhat a little bit
longer, then try to load slackware and give that a whirl.
Thanks again.
Sean
On Wed, 20 Aug 2003, John Fortman wrote:
I have been using Slackware 9 with the sources for ptlib_unix, pwlib_unix,
openh323, asterisk, zaptel and libpri in /root/src
1) load all 6 packages (+ ohphone if you need an h323 client) from cvs to
/root/src
2) /root/src/pwlib: configure, make, make install, ldconfig (not all that
sure why, but Slackware requires ldconfig to be run)
3) /root/src/openh323: configure, make, make install, ldconfig
4) /root/src/zaptel: make, make install (reload and reconfigure your zaptel
card)
5) /root/src/libpri: make, make install (I don't have a PRI card so I don't
do anything here)
6) /root/src/asterisk/channels/h323:
- edit Makfile
- set PWLIBDIR = $(HOME)/src/pwlib
- set OPENH323DIR = $(HOME)/src/openh323
- make, make install (installs openh323.a) (make samples if you do not
have h323.conf in /etc/asterisk when done)
7) /root/src/asterisk: make, make install, make samples
8) asterisk -vvvc
- the last section should load chan_h323
I haven't had any problems compiling this from CVS for almost a month on at
least three different systems with some version of Slackware. I have had
problems with other things like transferring calls but that's a different
issue.
John.
- Original Message -
From: Sean Figgins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 20, 2003 12:22 PM
Subject: Re: [Asterisk-Users] chan_h323.c
On Mon, 18 Aug 2003, Mark Spencer wrote:
It's up one directly. It just moved.
Run make in h323 then do make install on asterisk again.
On Mon, 18 Aug 2003, John Fortman wrote:
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp,
ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was
not created so no h323 support in asterisk.
Just wondering when to expect it again because I was stupid and didn't
make a backup of the asterisk code before wiping the directory for a
rebuild.
Has anyone gotten H.323 channel complied on Redhat 9.0? Every time I try,
I get a ton of errors from ptlib. I'm about ready to punt this sucker out
the door. I really like what I have seen out of asterisk so far...
Example of errors:
In file included from /usr/include/ptlib/contain.h:218,
from /usr/include/ptlib.h:137,
from ast_h323.h:29,
from ast_h323.cpp:27:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
PObject::BOOL'
/usr/include/ptlib/object.h:1214: parse error before `(' token
/usr/include/ptlib/object.h:1265: syntax error before `operator'
/usr/include/ptlib/object.h:1214: duplicate member `PObject::BOOL'
/usr/include/ptlib/object.h:1274: ISO C++ forbids defining types within
return
type
Any help would be appriciated, even if it's a recommendation to another
flavor of linux.
Thanks
Sean
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[Asterisk-Users] RTP header compression?
I sent this to the asterisk-dev by accident... Original Message Follows Hi all, I have a couple questions about RTP header compression with Asterisk: 1) Has this been implemented before or is it included in the Asterisk package? 2) If the answer to (1) is no, is there an RTP stack that this can be logically implemented into? Where would that be? Thanks, Kevin _ bGet MSN 8/b and enjoy automatic e-mail virus protection. http://join.msn.com/?page=features/virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VAD (silence suppression) on Asterisk
On Wed, 20 Aug 2003 13:28:59 -0500 (CDT) Brian West [EMAIL PROTECTED] wrote: Comfort Noise and VAD are diffrent things. bkw Yeap. But most devices when uses VAD looks out for gaps in speech and replaces those gaps with comfort noise. :-) [ ]'s Eduardo On Wed, 20 Aug 2003, Eduardo Goncalves wrote: When I turn on VAD on cisco ATA186, asterisk shows: Aug 20 11:12:20 NOTICE[618513]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible RCF3389 defines Payload for Comfort Noise, that is used with VAD. So I turned it off on my endpoints (ATA186 and c827-4v) Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA-186 locking: implausible unlock method
For those of you wanting to salvage your Cisco ATA-186 after inadvertent locking, or after recovering your devices from a vendor who has locked them, here is a rainy-day project for you: http://www.sst.com/downloads/datasheet/S71077.pdf The above document gives exact specifications on the 4mb flash EEPROM that stores all program and configuration data on the ATA-186 (aka Komodo.) If you have a suitably delicate hand, a large amount of sophisticated equipment, and some good clamps, you might be able to erase those sections on the chip which contain the configuration data. Maybe. :) I would advise writing off a few ATA's while you try this, and I'd also write off a few weeks of time while you attempt it (unless, of course, you happen to be an unnamed three-letter US government agency who is good at this particular type of craft, and you happen to send me an anonymous email message with instructions for the home user. Hint hint.) Please aim your negative karma at Cisco for creating a piece of hardware that can be rendered useless with software. This is against all previous ideology of Cisco, and is a disturbing trend. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VAD (silence suppression) on Asterisk
Thanks, that's the answer I was looking for. Do we know if VAD will ever be supported? I know some people don't like VAD and in my testing, how well VAD works depends on how well it was coded (and the hardware I suspect). I have seen very good and very bad implementations of VAD. Hard to give an answer to your question without a crystal ball. :) VAD is actually useful in situations where you have a long, narrow bandwidth pipe and users are already anticipating poor voice quality. I could see it as a useful feature for long-haul VoIP traffic overseas or over exotic network stub connections. I have a real need for VAD to work (for bandwidth reasons). So if VAD isn't currently in the Asterisk development schedule, can I request it be added? Yes, you can. Go to http://bugs.digium.com/ and put in a feature request. To enhance the chances of your request actually being implemented, include pointers to the RFCs that cover VAD with RTP, or IEEE VAD specs, or whatever documentation you can find that might assist a programmer in gettin VAD implemented with the least amount of their effort. This does not guarantee VAD being added, but it helps more than a less-specific request might otherwise. JT Thanks Lee Goodman - Original Message - From: Eduardo Goncalves [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 1:21 PM Subject: Re: [Asterisk-Users] VAD (silence suppression) on Asterisk When I turn on VAD on cisco ATA186, asterisk shows: Aug 20 11:12:20 NOTICE[618513]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible RCF3389 defines Payload for Comfort Noise, that is used with VAD. So I turned it off on my endpoints (ATA186 and c827-4v) Eduardo On Wed, 20 Aug 2003 11:35:14 -0500 (CDT) Brian West [EMAIL PROTECTED] wrote: VAD is evil. I hate it. I find when we used it.. you keep asking people to repeat stuff all the time.. and it was just anoying. bkw On Wed, 20 Aug 2003, WipeOut . wrote: Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman Somthing tells me that it is not supported but it is somthing that I would like to see supported as well.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] X-Lite Build 1059 problems
I've had similar problems with X-Lite to X-lite calls. Sometimes you get a one second burst of audio. After putting the call on hold and then resuming the call the audio seemed to then work. I think there are definitely problems. -Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Hirst Sent: 20 August 2003 16:43 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X-Lite Build 1059 problems Does anyone have X-Lite build 1059 working fully with Asterisk? The GSM Codec works very well now but we have problems when using G711 in that when I setup a ping between the two sites and then watch the latency, it steadily increases and starts at about 150ms and goes up to 2500ms within about 20 seconds. I have not investigated fully but I guess that its sending ever increasing size packets. When we use the GSM codec its fine. When we have used the G711 codec in previous releases of X-Lite, it's been fine. Also we get one way speech in the following circumstances :- SNOM 200 running 1.16w 4941 Grandstream Budgetone running 1.0.3.77 Calls between SNOM and Budgetone are fine in both directions. Call from X-Lite to Asterisk voicemail = Fine Call from X-Lite to either SNOM or Budgetone = Fine Call from SNOM to X-Lite = Call is delivered and answered OK but there is no speech path in either direction. If its any help one of our guys is on the end of a slow link (150ms) and the symptoms are the same but there is a 1 second audio burst from the X-Lite to SNOM just as the call is answered. Call from Budgetone to X-Lite = Call is delivered and answered OK but one way speech path. Budgetone to X-Lite OK but X-Lite to Budgetone, no audio. I do have call traces from X-Lite if anyone is interested but I would like to know if any other X-Lite users were seeing the same type of problems. Rgds, Stuart ___ 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] Strange happenings
Just idly watching * in console mode and saw that someone from 50.49.54.102 tried to register with my *. whois gives:- OrgName:Internet Assigned Numbers Authority OrgID: IANA Address:4676 Admiralty Way, Suite 330 City: Marina del Rey StateProv: CA PostalCode: 90292-6695 Country:US NetRange: 50.0.0.0 - 50.255.255.255 CIDR: 50.0.0.0/8 NetName:RESERVED-50 NetHandle: NET-50-0-0-0-0 Parent: NetType:IANA Reserved Comment: RegDate: Updated:2002-08-23 OrgTechHandle: IANA-ARIN OrgTechName: Internet Corporation for Assigned Names and Number OrgTechPhone: +1-310-823-9358 OrgTechEmail: [EMAIL PROTECTED] ? -- Dave Cotton [EMAIL PROTECTED] 1) Using what protocol? 2) As John Brown mentions, do a tracroute to that IP from your location. My views show that the address you list is not in the global BGP tables. Spoofed single packets might hit your machines, but they have no way of getting back unless your local routers have static or locally learned routes to that destination. In short: don't worry about it, probably. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP dialtone?
Does http://www.voicepulse.com/ work with *? On Wed, 20 Aug 2003, John Todd wrote: At 3:20 PM -0500 8/20/03, Mike Ciholas wrote: Hi all, While pondering my choices for local dial tone service via a bunch of POTS lines or a T1, I began to wonder if perhaps there is another way. Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? I guess I am imagining a company that gateways between the PTSN and the internet backbone. Calls come in and get VoIP'ed and sent to me as packets, perhaps IAX, perhaps something else? First question: Does such a thing exist? Where? Yes. http://www.iconnecthere.com/ http://www.packet8.net/ http://www.nufone.net/ http://www.coloco.com/ (not obviously visible on the home page, but exists) http://www.voicepulse.com/ ...many others. Use your favorite search engine to look up SIP long distance providers. Some of the above (notably NuFone and Coloco) will provide IAX/IAX2 termination. Second question: Does it work? How well? Works great. I haven't made a long distance call on my PSTN line in months, and I spend pretty much all day on LD calls. Third question: Would you want it? Why? Yes. Cheap, portable, failure-tolerant. Note that your phone service suddenly becomes as (un)reliable as your Internet connectivity, so ensure that you have those bases covered through the normal methods such as multihoming, facility redundancy, MPLS, etc. I would also suggest you have multiple outbound VoIP providers, with automatic failover configured in your Asterisk server. This is easily done. Fourth question: How much $$$? As little as $.01 a minute anywhere in the US, and great international rates, depending on providers. Remember you can get multiple accounts, and send your calls to different providers based on static tables of who you think is cheapest for that dial prefix. To address your previous question of is it ready for prime time the answer is: For basic features, absolutely. I have several customers whose systems I have configured for their offices... and I haven't heard from them in MONTHS. The systems have had 100% uptime, handling calls from POTS and VoIP lines. For exotic features: maybe. There is a HUGE list of niggly little features that everyone is in love with in their particular PBX. Some of those features, Asterisk does exceedingly well, and others that are less frequently used, it does not. However, this situation is no different with Asterisk than with any other PBX system that you might evaluate, so all things being equal I'd say Asterisk is a LOT better than a proprietary solution since you can get under the hood yourself and fix things that might need to be updated. JT -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ 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] VoIP dialtone?
On Wed, 20 Aug 2003, John Todd wrote: At 3:20 PM -0500 8/20/03, Mike Ciholas wrote: Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? First question: Does such a thing exist? Where? Yes. Second question: Does it work? How well? Works great. I haven't made a long distance call on my PSTN line in months, and I spend pretty much all day on LD calls. I guess my question was a little deeper than that. Can I simply ditch the PTSN? I see that toll free inbound and LD outbound can be handled, can they handle inbound and local, too? Seems like we are very close to cutting the local phone company out of the loop! That would be so nice as trying to talk them about provisioning the lines is quite a chore. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk introductory talk: Portland, OR USA
Any chance of handouts, transcripts, or video being posted to your website soon? On Wed, 20 Aug 2003, John Todd wrote: For those of you that are in the Portland, Oregon area: I am giving a talk today on Asterisk at the PLUG Advanced Topics Meeting. Details below. JT From: Zot O'Connor [EMAIL PROTECTED] To: PLUG LIST [EMAIL PROTECTED], PLUG Announcement List [EMAIL PROTECTED] Organization: White Knight Hacklers Subject: [PLUG] 2nd Announcement: Advanced Linux SIG - 7PM Wednesday Aug 20th --Asterkisk PBX Reply-To: [EMAIL PROTECTED] Date: Mon, 18 Aug 2003 16:56:21 -0700 Please note the new location (JAX). John Todd will talk on Asterisk: Open-Source VoIP Telephony in 3 Beer's Time Or Less General guidelines will be: - Asterisk: introduction - Goals of the project - license notes from Digium - Two brief implementation notes: enterprise and home examples - what asterisk isn't - components: hardware and software - components: channels, configuration files, how it is a voice router - what is VoIP: SIP, H.323, MGCP - hardware: capi, modem, zap analog, zap digital - requirements to run Asterisk - applications: - answering machine - IVR - VoIP gateway to upstream providers - follow-me calling - call screening - voice interaction front-end to anything you want - outbound call generator - call center applications - toll avoidance trunking for switching platforms - etc. etc. etc. as time permits - gotchas, readmes, caveats about the system - resource list Time: 7pm Date: Wednesday Aug 20th, 2003 Location: Jax [Restaurant] 110 SW Yamhill St. Portland, OR On the MAX Line, next to Bally's Fitness. This is around the corner from Paddy's about There will be food, alcohol, and much technical discussion. Projector: I do need a project still. I plan on bringing the one that needs a power supply fix (really I am), so if you can bring a projector please email me. Thanks! -- Zot O'Connor http://www.ZotConsulting.com http://www.WhiteKnightHackers.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP dialtone?
Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? I guess I am imagining a company that gateways between the PTSN and the internet backbone. Calls come in and get VoIP'ed and sent to me as packets, perhaps IAX, perhaps something else? First question: Does such a thing exist? Where? Yes; Delta 3, Vonage, and a bunch of other companies that I can't rememeber off the top of my head do exactly this. /a ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP dialtone?
On Wed, 20 Aug 2003, Brian West wrote: I think NuFone can do what you need contact [EMAIL PROTECTED] I have inbound 800 service and outbound ld service with them.. works great. And for local service, you do what? -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP dialtone?
I guess my question was a little deeper than that. Can I simply ditch the PTSN? 911 is the sticking point. Most commercial VoIP services come with the disclaimer that they are *not* a primary line replacement, precisely because of the liability issues associated with providing emergency services. For example, a typical configuration would be to not care where the caller is from, and simply route calls according to the country and city code, as appropriate. If you dump 911 into such a system, it has no way to route you to an appropriate operator. Sitting around in Indiana talking to a 911 operator in Los Angeles generally does you very little good. That said, in controlled environments, some services are now offering VoIP primary line replacements. The only service I currently know that is doing so is Vonage (http://www.vonage.com/), and it is doing so only in very specific markets at the moment. Further, the handling of 911 in their system is sub-optimal[1], in as much as it doesn't dump you into the normal 911 queue, and the PSAP will not have any information about your location. In, say, a medical emergency, I would far prefer to be talking about the emergency itself than trying to spell the name of my street. Until this tiny, possibly life-or-death detail gets sorted out, I'm probably going to have at least one traditional phone line at all times. /a [1] See http://www.vonage.com/small_business/features_911.php ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP dialtone?
pipe my local CO line into my * box with an X100P bkw On Wed, 20 Aug 2003, Mike Ciholas wrote: On Wed, 20 Aug 2003, Brian West wrote: I think NuFone can do what you need contact [EMAIL PROTECTED] I have inbound 800 service and outbound ld service with them.. works great. And for local service, you do what? -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP dialtone?
On Wed, 20 Aug 2003, Adam Roach wrote: I guess my question was a little deeper than that. Can I simply ditch the PTSN? 911 is the sticking point. Ah. Until this tiny, possibly life-or-death detail gets sorted out, I'm probably going to have at least one traditional phone line at all times. Hmm, okay, so would it be possible to maintain *one* POTS line that is used if anyone dials 911 on their desk phone (set this up in * dial plan), then it connects to emergency services properly, and then use a VoIP dial tone provider for *everything* else? This assumes we are having only one emergency at a time! Now, if that is possible, how does the VoIP dial tone provider get my inbound local and toll calls? I would want my local phone number to work, of course. -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP dialtone?
At 04:48 PM 8/20/2003 -0500, you wrote: Hmm, okay, so would it be possible to maintain *one* POTS line that is used if anyone dials 911 on their desk phone (set this up in * dial plan), then it connects to emergency services properly, and then use a VoIP dial tone provider for *everything* else? This assumes we are having only one emergency at a time! Yes, that would work fine. Now, if that is possible, how does the VoIP dial tone provider get my inbound local and toll calls? I would want my local phone number to work, of course. You would need to redirect your local number to them. This ALWAYS assumes that the VoIP provider has a switch in your local CO or an agreement with someone who does. Vonage and Voicepulse, for example, do not have a presence in my area. I intend to maintain several POTS lines for incoming calls, and use a VoIP provider for all outgoing calls. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
Is this the Adtran 624 series channel bank? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Wednesday, August 20, 2003 9:55 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank Thanks for your help. Bart - Original Message - From: Steve Creel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 11:47 AM Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank Then yes, it will work and do what you're looking for it to do. On Wed, 20 Aug 2003, Bartosz Jozwiak wrote: I want to connect analog telephone lines only. The analog lines telecom gives you :) - Original Message - From: Steve Meyers [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 11:34 AM Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank On Wed, 2003-08-20 at 07:58, Mark Spencer wrote: The FXO ports will only allow you to connect phone lines, not actual phones, but since FXO ports are more expensive in general than FXS ones, it's likely you could find someone to trade. We probably should have a list dedicated to trading/selling/buying asterisk related hardware, but failing that i would suggest people just contact you off-list. Yeah, but will it work? What if he wants 24 port FXO, not FXS? Steve ___ 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 ___ Steve Creel[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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP dialtone?
On Wed, 20 Aug 2003, Ernest W. Lessenger wrote: At 04:48 PM 8/20/2003 -0500, you wrote: Now, if that is possible, how does the VoIP dial tone provider get my inbound local and toll calls? I would want my local phone number to work, of course. You would need to redirect your local number to them. This ALWAYS assumes that the VoIP provider has a switch in your local CO or an agreement with someone who does. Vonage and Voicepulse, for example, do not have a presence in my area. I intend to maintain several POTS lines for incoming calls, and use a VoIP provider for all outgoing calls. Oh well. I'm would expect no one would have presence here. This sounds so suboptimal, you have to provision *two* systems, one for inbound (local CO) and one for outbound (VoIP provider). Of course, the outbound can be just your internet connection, but this still seems annoying because most of the money is in the local CO service. Hmm, perhaps *all* incoming calls can be toll free? I would maintain the one local CO POTS line for 911 out bound, and then only use my toll free number for inbound. For the money I would save on local CO lines I can buy a *lot* of toll free minutes! Then the VoIP dial tone provider can route my toll free number to me over the internet. Presumably, then, there is no real limit on the number of lines coming in. It isn't hard coded like the CO lines are. This all seems pretty fanciful at the moment... -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware question
Here are some options: Digium X100P x 4 US$100 * 4 = US$400 well supported by asterisk manufacturer supports asterisk developers Deployed in lots of places with Asterisk Voicetronix OpenLine4 - US$500? Use 1 PCI card reported working with chan_vpb manufacturer supports linux Can be used with other software Voicetronix OpenSwitch 6(12) US$750?(US$1800?) Use 1 PCI card works with chan_vpb manufacturer supports linux Can be used with other software 6/12 ports can be jumpered in pairs(4s?) to be FXO or FXS. Channel Bank + T/E 100/400 P US$? use 1 PCI card On Wed, Aug 20, 2003 at 08:27:06AM -0700, Bruce Ferrell wrote: Digium makes a 4 port card. It'd be hard to get 4 lines with quicknet hardware. Bartosz Jozwiak wrote: Hello, Again one more question about hardware. What could you suggest me to buy. I need hardware to connect let's say 4 analog lines. (FXO). This hardware should talk to Asterisk of course.. Thanks very much for some advices :) Bartek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
