Re: [Asterisk-Users] TDM
Connect to the Asterisk console with verbose turned on and try to dial. Post that output. Curt Shaffer wrote: This is not [EMAIL PROTECTED] it is asterisk with FreePBX only. Yes the phone line is connected to the right port. No luck. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Saturday, May 27, 2006 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Steve Totaro wrote: Is your machine seeing the card? /var/log/messages? Are you loading the zaptel drivers? modprobe zaptel, modprobe wctdm? Would he get the ztcfg message if it were not? Is the phone line plugged into the correct jack? With only one module installed, the other three jacks lead to nowhere. Also this seems to be [EMAIL PROTECTED] from the references, so perhaps there is a context issue that the configuration files address. AAH can really lead one down the garden path! John Novack Curt Shaffer wrote: The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include zapata_additional.conf context=from-zaptel signalling=fxs_ks faxdetect=incoming usecallerid=asreceived echocancel=yes callprogress=no busydetect=no echocancelwhenbridged=no echotraining=800 group=0 channel=1 When I dial in Asterisk does not even show an initiation of the call. When I dial out on that trunk I get all circuits busy. Ztcfg -vvv shows the following ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Any help would be appreciated. Curt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FreeBSD Digium g.729 codec seg faults on rev 30652
Greetings- Was running the Digium FreeBSD g.729 codec until recently when the latest Asterisk bits were obtained via svn: svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk This produced: Checked out revision 30652 This on FreeBSD 6.1-RELEASE Attempting to start asterisk it returns: == Registered custom function URIENCODE [codec_g729a.so]May 27 13:29:59 WARNING[71884]: loader.c:728 __load_resource: missing mod_data for codec_g729a.so Segmentation fault (core dumped) This codec was licensed through Digium and, while listed as 'unsupported', worked very well. Any way Digium will release a G.729 codec for FreeBSD which is useable with rev 30652 ? regards -kim -- w8hdkim er.. gmail ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling a person over Internet
Hello Michiel, * Michiel van Baak [EMAIL PROTECTED] [27-05-06 17:15]: You have to do a couple of things: 1. Open your firewall so it allows the protocol you want to use. ok, that should be easy. 2. Configure asterisk to accept guest calls 3. Configure asterisk to ring some phones when someone dials your domain. and how is this working? Is the person who want call me dial [EMAIL PROTECTED] How can ppl reach me if they only use SIP? If there is a site or howto etc. available it would be a pleasure for my to get something to read :) Remember, only ppl with voip can reach you this way. Normal landline phones can only reach you when you have a landline connected to a tdm card or if you connect with a voip provider. I have a ISDN card in my PC which is working perfectly. Thx for answers. Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] My Call drop after 60 to 63 Seconds!!
Dear all, I have an Asterisk box running [EMAIL PROTECTED] 2.7 . and A2billing. when my asterisk box dial using dialcommand_param=|45|HL(%timeout%:61000:3) its working fine . but when i use dialcommand_param=|45|L(%timeout%) call got drop after 62 seconds. i used this same setting into my other two Asterisk boxes and those r working fine . but now i am trying this into a new Dell PC (GX series) and facing that strange problem. i think its something related to my PC . or Asterisk setting. Could anyone guide me where to look for thanks Salaque -- Got my Gmail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling a person over Internet
On 13:19, Sun 28 May 06, Matthias Fechner wrote: Hello Michiel, 2. Configure asterisk to accept guest calls 3. Configure asterisk to ring some phones when someone dials your domain. and how is this working? Is the person who want call me dial [EMAIL PROTECTED] How can ppl reach me if they only use SIP? In sip.conf you have to allow guest calls (read the comments in the default conf) the context = line in globals puts those calls in that context in extensions.conf in extensions.conf define the names/numbers ppl can use to reach you. mine has: michiel = Dial(SCCP/office) (of course that's a simplified version) That way ppl can reach me on: [EMAIL PROTECTED] have fun -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER qualify
Hi! I know that is not SER discuss, but probably some of you faced with the same problem: to detect trunk status (ok/unreachable) in *, it's a must, to set qualify=yes as * connecting to SER, it's not replying to qualify messages, so even i can use it well without qualify, with qualify it's says unreachable immediately. What i have to set in SER to reply? Thanks! -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy Signals
I think asterisk dropping you to s-BUSY, s-CONGESTED, s-UNREACHABLE
priority, better have a look there
(you can play a busy tone, or playback(called-party-is-busy))
A few employees have noticed some problem here and there when trying to
make outgoing phone calls. After it happens, they try again, and are
able to call through.
The dial plan for outbound calling looks like below. Which I know they
are getting to the Congestion part (which explains the busy) but what I
can't seem to figure out is the cause for why they are getting sent there.
exten = s,1,SetCallerID(${ARG1})
exten = s,2,Wait(2)
exten = s,3,Dial(${TRUNK1}/${ARG2})
exten = s,4,Congestion(10)
exten = s,104,Congestion(10)
The log for a call looked like this
May 26 12:21:08 VERBOSE[6997] logger.c: -- Channel 0/4, span 1 got
hangup request
May 26 12:21:08 VERBOSE[16613] logger.c: -- Zap/4-1 is circuit-busy
May 26 12:21:08 VERBOSE[16613] logger.c: -- Hungup 'Zap/4-1'
May 26 12:21:08 VERBOSE[16613] logger.c: == Everyone is busy/congested
at this time (1:0/1/0)
My question is it asterisk having an issue with the PRI or is the PRI
really reporting the number is busy. I know one case like this I was
calling home, and which when I got through to them, they were not even
on the phone. Are there any tests that I can run on the T1 card in the
server to the PRI? Any suggestions would be helpful.
Kevin
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WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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RE: [Asterisk-Users] TDM
Here is the output from a dial when starting asterisk with -v. The 1NXXNXX is actually the number not those characters FYI. Thanks -- Executing Macro(SIP/103-a555, dialout-trunk|1|1NXXNXX||) in new stack -- Executing GotoIf(SIP/103-a555, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/103-a555, user-callerid) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing GotoIf(SIP/103-a555, 0?start) in new stack -- Executing Set(SIP/103-a555, REALCALLERIDNUM=103) in new stack -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, AMPUSER=103) in new stack -- Executing Set(SIP/103-a555, AMPUSERCIDNAME=103) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing Set(SIP/103-a555, CALLERID(all)=103 103) in new stack -- Executing NoOp(SIP/103-a555, Using CallerID 103 103) in new stack -- Executing Macro(SIP/103-a555, record-enable|103|OUT) in new stack -- Executing GotoIf(SIP/103-a555, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/103-a555, recordingcheck|20060528-110627|1148832387.1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060528-110627|1148832387.1: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/103-a555, No recording needed) in new stack -- Executing Macro(SIP/103-a555, outbound-callerid|1) in new stack -- Executing GotoIf(SIP/103-a555, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, USEROUTCID=) in new stack -- Executing Set(SIP/103-a555, EMERGENCYCID=) in new stack -- Executing Set(SIP/103-a555, TRUNKOUTCID=) in new stack -- Executing GotoIf(SIP/103-a555, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf(SIP/103-a555, 1?usercid) in new stack -- Goto (macro-outbound-callerid,s,13) -- Executing GotoIf(SIP/103-a555, 1?report) in new stack -- Goto (macro-outbound-callerid,s,15) -- Executing NoOp(SIP/103-a555, CallerID set to 103 103) in new stack -- Executing Set(SIP/103-a555, GROUP()=OUT_1) in new stack -- Executing GotoIf(SIP/103-a555, 0?108) in new stack -- Executing Set(SIP/103-a555, DIAL_NUMBER=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/103-a555, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(SIP/103-a555, OUTNUM=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/103-a555, 0?16) in new stack -- Executing Dial(SIP/103-a555, ZAP/g0/1NXXNXX|120|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(SIP/103-a555, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/103-a555, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(SIP/103-a555, outisbusy|) in new stack -- Executing Playback(SIP/103-a555, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/103-a555, pls-try-call-later) in new stack -- Playing 'pls-try-call-later' (language 'en') == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' in macro 'outisbusy' == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, May 28, 2006 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Connect to the Asterisk console with verbose turned on and try to dial. Post that output. Curt Shaffer wrote: This is not [EMAIL PROTECTED] it is asterisk with FreePBX only. Yes the phone line is connected to the right port. No luck. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Saturday, May 27, 2006 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Steve Totaro wrote: Is your machine seeing the card? /var/log/messages? Are you loading the zaptel drivers? modprobe zaptel, modprobe wctdm? Would he get the ztcfg message if it were not? Is the phone line plugged into the correct jack? With only one module installed, the other three jacks lead to nowhere. Also this seems to be [EMAIL PROTECTED] from the references, so perhaps
RE: [Asterisk-Users] TDM
Here is the output from a dial when starting asterisk with -v. The 1NXXNXX is actually the number not those characters FYI. Thanks -- Executing Macro(SIP/103-a555, dialout-trunk|1|1NXXNXX||) in new stack -- Executing GotoIf(SIP/103-a555, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/103-a555, user-callerid) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing GotoIf(SIP/103-a555, 0?start) in new stack -- Executing Set(SIP/103-a555, REALCALLERIDNUM=103) in new stack -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, AMPUSER=103) in new stack -- Executing Set(SIP/103-a555, AMPUSERCIDNAME=103) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing Set(SIP/103-a555, CALLERID(all)=103 103) in new stack -- Executing NoOp(SIP/103-a555, Using CallerID 103 103) in new stack -- Executing Macro(SIP/103-a555, record-enable|103|OUT) in new stack -- Executing GotoIf(SIP/103-a555, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/103-a555, recordingcheck|20060528-110627|1148832387.1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060528-110627|1148832387.1: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/103-a555, No recording needed) in new stack -- Executing Macro(SIP/103-a555, outbound-callerid|1) in new stack -- Executing GotoIf(SIP/103-a555, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, USEROUTCID=) in new stack -- Executing Set(SIP/103-a555, EMERGENCYCID=) in new stack -- Executing Set(SIP/103-a555, TRUNKOUTCID=) in new stack -- Executing GotoIf(SIP/103-a555, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf(SIP/103-a555, 1?usercid) in new stack -- Goto (macro-outbound-callerid,s,13) -- Executing GotoIf(SIP/103-a555, 1?report) in new stack -- Goto (macro-outbound-callerid,s,15) -- Executing NoOp(SIP/103-a555, CallerID set to 103 103) in new stack -- Executing Set(SIP/103-a555, GROUP()=OUT_1) in new stack -- Executing GotoIf(SIP/103-a555, 0?108) in new stack -- Executing Set(SIP/103-a555, DIAL_NUMBER=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/103-a555, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(SIP/103-a555, OUTNUM=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/103-a555, 0?16) in new stack -- Executing Dial(SIP/103-a555, ZAP/g0/1NXXNXX|120|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(SIP/103-a555, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/103-a555, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(SIP/103-a555, outisbusy|) in new stack -- Executing Playback(SIP/103-a555, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/103-a555, pls-try-call-later) in new stack -- Playing 'pls-try-call-later' (language 'en') == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' in macro 'outisbusy' == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, May 28, 2006 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Connect to the Asterisk console with verbose turned on and try to dial. Post that output. Curt Shaffer wrote: This is not [EMAIL PROTECTED] it is asterisk with FreePBX only. Yes the phone line is connected to the right port. No luck. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Saturday, May 27, 2006 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Steve Totaro wrote: Is your machine seeing the card? /var/log/messages? Are you loading the zaptel drivers? modprobe zaptel, modprobe wctdm? Would he get the ztcfg message if it were not? Is the phone line plugged into the correct jack? With only one module installed, the other three jacks lead to nowhere. Also this seems to be [EMAIL PROTECTED] from the references, so perhaps
Re: [Asterisk-Users] FreeBSD Digium g.729 codec seg faults on rev 30652
Kim Culhan wrote: Was running the Digium FreeBSD g.729 codec until recently when the latest Asterisk bits were obtained via svn: svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk MAYBE it is the same problem: http://lists.digium.com/pipermail/asterisk-users/2006-April/147577.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 or asterisk
If someone here happens to have a mpg123 binary compiled for Centos 43 in a Pentium Dual Core, let me know. Somehow mpg123 cant compile. [EMAIL PROTECTED] mpg123-0.59r]# make linux make CC=gcc LDFLAGS= \ OBJECTS='decode_i386.o dct64_i386.o decode_i586.o \ audio_oss.o term.o' \ CFLAGS='-DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX \ -DREAD_MMAP -DOSS -DTERM_CONTROL\ -Wall -O2 -m486 \ -fomit-frame-pointer -funroll-all-loops \ -finline-functions -ffast-math' \ mpg123-make make[1]: Entering directory `/root/sources/mpg123/mpg123-0.59r' make[2]: Entering directory `/root/sources/mpg123/mpg123-0.59r' as -o decode_i586.o decode_i586.s decode_i586.s: Assembler messages: decode_i586.s:44: Error: suffix or operands invalid for `push' decode_i586.s:45: Error: suffix or operands invalid for `push' decode_i586.s:46: Error: suffix or operands invalid for `push' decode_i586.s:47: Error: suffix or operands invalid for `push' decode_i586.s:67: Error: suffix or operands invalid for `push' decode_i586.s:70: Error: suffix or operands invalid for `push' decode_i586.s:81: Error: suffix or operands invalid for `push' decode_i586.s:83: Error: suffix or operands invalid for `push' decode_i586.s:86: Error: suffix or operands invalid for `push' decode_i586.s:161: Error: suffix or operands invalid for `pop' decode_i586.s:211: Error: suffix or operands invalid for `pop' decode_i586.s:296: Error: suffix or operands invalid for `pop' decode_i586.s:315: Error: suffix or operands invalid for `pop' decode_i586.s:316: Error: suffix or operands invalid for `pop' decode_i586.s:317: Error: suffix or operands invalid for `pop' decode_i586.s:318: Error: suffix or operands invalid for `pop' make[2]: *** [decode_i586.o] Error 1 make[2]: Leaving directory `/root/sources/mpg123/mpg123-0.59r' make[1]: *** [mpg123-make] Error 2 make[1]: Leaving directory `/root/sources/mpg123/mpg123-0.59r' make: *** [linux] Error 2 [EMAIL PROTECTED] mpg123-0.59r]# On 5/27/06, Steve Totaro [EMAIL PROTECTED] wrote: Please let us know your results. I cannot really test this in production system since it is a $16,000/hr call center. I was using madplay but it was crashing and creating zombie processes, I figured native was not the way to go since all of the different audio streams. Mpg123 works perfectly for me under a load of sixty channels, I can confirm that for sure. Thanks, Steve Erick Perez wrote: Interesting. So, i will have to test then... On 5/27/06, Steve Totaro [EMAIL PROTECTED] wrote: In my very limited testing of native, each channel was receiving a different stream (each caller heard something different). Under a high volume of calls, which is going to hurt performance more? Transcoding MP3s but sending a single stream or separate streams per call under native? When I say high, I mean 1,000+ calls. Thanks, Steve Erick Perez wrote: Thanks to all. Native format will be. On 5/27/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: Vahan Yerkanian wrote: Erick Perez wrote: should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. mpg123 was used back when asterisk didn't have native format support. If you are expecting heavy load, the native format is the way to go. You might decide not to use mp3 format at all, recompressing your MoH files using sox to the formats you gonna use, such as .al, .ul, .gsm, or leave it at .sln to cut the decoding leg only. Heh, damn this GPRS connection. In order to pass the time while downloading messages I reply before they are all in, and yet by the time I have received all the messages I note that your question has already been answered! :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama
Re: [Asterisk-Users] TDM
It looks OK. Try editing extensions.conf and add an extension in a context that will included when you dial. Try something like this exten = 123,1,Dial(ZAP/g0/1NXXNXX) The open the console and dial 123. This will bypass any funky dialplan issues with FreePBX. If it works, then obviously something is not right in FreePBX. If it doesnt' then that indicates your configuration files need tweaking. Thanks, Steve Curt Shaffer wrote: Here is the output from a dial when starting asterisk with -v. The 1NXXNXX is actually the number not those characters FYI. Thanks -- Executing Macro(SIP/103-a555, dialout-trunk|1|1NXXNXX||) in new stack -- Executing GotoIf(SIP/103-a555, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/103-a555, user-callerid) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing GotoIf(SIP/103-a555, 0?start) in new stack -- Executing Set(SIP/103-a555, REALCALLERIDNUM=103) in new stack -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, AMPUSER=103) in new stack -- Executing Set(SIP/103-a555, AMPUSERCIDNAME=103) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing Set(SIP/103-a555, CALLERID(all)=103 103) in new stack -- Executing NoOp(SIP/103-a555, Using CallerID 103 103) in new stack -- Executing Macro(SIP/103-a555, record-enable|103|OUT) in new stack -- Executing GotoIf(SIP/103-a555, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/103-a555, recordingcheck|20060528-110627|1148832387.1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060528-110627|1148832387.1: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/103-a555, No recording needed) in new stack -- Executing Macro(SIP/103-a555, outbound-callerid|1) in new stack -- Executing GotoIf(SIP/103-a555, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, USEROUTCID=) in new stack -- Executing Set(SIP/103-a555, EMERGENCYCID=) in new stack -- Executing Set(SIP/103-a555, TRUNKOUTCID=) in new stack -- Executing GotoIf(SIP/103-a555, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf(SIP/103-a555, 1?usercid) in new stack -- Goto (macro-outbound-callerid,s,13) -- Executing GotoIf(SIP/103-a555, 1?report) in new stack -- Goto (macro-outbound-callerid,s,15) -- Executing NoOp(SIP/103-a555, CallerID set to 103 103) in new stack -- Executing Set(SIP/103-a555, GROUP()=OUT_1) in new stack -- Executing GotoIf(SIP/103-a555, 0?108) in new stack -- Executing Set(SIP/103-a555, DIAL_NUMBER=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/103-a555, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(SIP/103-a555, OUTNUM=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/103-a555, 0?16) in new stack -- Executing Dial(SIP/103-a555, ZAP/g0/1NXXNXX|120|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(SIP/103-a555, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/103-a555, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(SIP/103-a555, outisbusy|) in new stack -- Executing Playback(SIP/103-a555, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/103-a555, pls-try-call-later) in new stack -- Playing 'pls-try-call-later' (language 'en') == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' in macro 'outisbusy' == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, May 28, 2006 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Connect to the Asterisk console with verbose turned on and try to dial. Post that output. Curt Shaffer wrote: This is not [EMAIL PROTECTED] it is asterisk with FreePBX only. Yes the phone line is connected to the right port. No luck. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Saturday, May 27, 2006 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion
[Asterisk-Users] Calls connected, but no audio
Using sip connections some peers are not able to transmit or recieve audio. All peers are setup the same aside from the NAT settings. The call will go through, called device will ring, but when it answers there is no audio connection. From the callee, they will not here the rings, only silence when they dial the phone. The kicker is that sometimes it will work, and other times it will not. Miles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls connected, but no audio
You need to describe your NAT setup more. One thing to try is to set qualify to yes or a short number. Essentially a keepalive for any routers in the middle. If you have multiple phones behind a remote NAT, make sure they are using different ports. Miles Scruggs wrote: Using sip connections some peers are not able to transmit or recieve audio. All peers are setup the same aside from the NAT settings. The call will go through, called device will ring, but when it answers there is no audio connection. From the callee, they will not here the rings, only silence when they dial the phone. The kicker is that sometimes it will work, and other times it will not. Miles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 or asterisk
I can't imagine why Gnu's assembler compiler wouldn't work for i586 instructions, but does mpg123 perhaps need NASM instead? No guarantees, but I was toying with installing a CentOS 4.3 box today anyway, maybe I'll see if I can get it to compile for me and let you know if and how. I don't have a dual-core Pentium, but in a Google search I found folks with this same behavior on Athlon, Opteron, P3 and P4 architectures, so I'm thinking it's some kind of missing package/dependency/library or similar. Diamon Erick Perez wrote: If someone here happens to have a mpg123 binary compiled for Centos 43 in a Pentium Dual Core, let me know. Somehow mpg123 cant compile. [EMAIL PROTECTED] mpg123-0.59r]# make linux make CC=gcc LDFLAGS= \ OBJECTS='decode_i386.o dct64_i386.o decode_i586.o \ audio_oss.o term.o' \ CFLAGS='-DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX \ -DREAD_MMAP -DOSS -DTERM_CONTROL\ -Wall -O2 -m486 \ -fomit-frame-pointer -funroll-all-loops \ -finline-functions -ffast-math' \ mpg123-make make[1]: Entering directory `/root/sources/mpg123/mpg123-0.59r' make[2]: Entering directory `/root/sources/mpg123/mpg123-0.59r' as -o decode_i586.o decode_i586.s decode_i586.s: Assembler messages: decode_i586.s:44: Error: suffix or operands invalid for `push' decode_i586.s:45: Error: suffix or operands invalid for `push' decode_i586.s:46: Error: suffix or operands invalid for `push' decode_i586.s:47: Error: suffix or operands invalid for `push' decode_i586.s:67: Error: suffix or operands invalid for `push' decode_i586.s:70: Error: suffix or operands invalid for `push' decode_i586.s:81: Error: suffix or operands invalid for `push' decode_i586.s:83: Error: suffix or operands invalid for `push' decode_i586.s:86: Error: suffix or operands invalid for `push' decode_i586.s:161: Error: suffix or operands invalid for `pop' decode_i586.s:211: Error: suffix or operands invalid for `pop' decode_i586.s:296: Error: suffix or operands invalid for `pop' decode_i586.s:315: Error: suffix or operands invalid for `pop' decode_i586.s:316: Error: suffix or operands invalid for `pop' decode_i586.s:317: Error: suffix or operands invalid for `pop' decode_i586.s:318: Error: suffix or operands invalid for `pop' make[2]: *** [decode_i586.o] Error 1 make[2]: Leaving directory `/root/sources/mpg123/mpg123-0.59r' make[1]: *** [mpg123-make] Error 2 make[1]: Leaving directory `/root/sources/mpg123/mpg123-0.59r' make: *** [linux] Error 2 [EMAIL PROTECTED] mpg123-0.59r]# On 5/27/06, Steve Totaro [EMAIL PROTECTED] wrote: Please let us know your results. I cannot really test this in production system since it is a $16,000/hr call center. I was using madplay but it was crashing and creating zombie processes, I figured native was not the way to go since all of the different audio streams. Mpg123 works perfectly for me under a load of sixty channels, I can confirm that for sure. Thanks, Steve Erick Perez wrote: Interesting. So, i will have to test then... On 5/27/06, Steve Totaro [EMAIL PROTECTED] wrote: In my very limited testing of native, each channel was receiving a different stream (each caller heard something different). Under a high volume of calls, which is going to hurt performance more? Transcoding MP3s but sending a single stream or separate streams per call under native? When I say high, I mean 1,000+ calls. Thanks, Steve Erick Perez wrote: Thanks to all. Native format will be. On 5/27/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: Vahan Yerkanian wrote: Erick Perez wrote: should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. mpg123 was used back when asterisk didn't have native format support. If you are expecting heavy load, the native format is the way to go. You might decide not to use mp3 format at all, recompressing your MoH files using sox to the formats you gonna use, such as .al, .ul, .gsm, or leave it at .sln to cut the decoding leg only. Heh, damn this GPRS connection. In order to pass the time while downloading messages I reply before they are all in, and yet by the time I have received all the messages I note that your question has already been answered! :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth
Re: [Asterisk-Users] Polycom 600 presence indication on *LED*?
Jerry Jones wrote: Create a contact entry with their extension and enable buddy watch on it It will then show up on an unused line key On May 27, 2006, at 3:26 PM, Faris Raouf wrote: I've somehow managed to battle may way through hinting issues with type=peer type=friend and various other oddities and now have presence working correctly on my Polycom 600 and Grandstream GXP-2000 phones. However, on the Polycom I have to press the Buddies softkey in order to see if an extension with a hints priority is in use or not. I've spent all day going through google and my local archive of the mailing list, and from what I can see it appears that I should be able to set up one of the 6 line keys on the left of the phone to somehow show presence indications. But I simply cannot figure out how. I only know how to configure a line key as, well, a line key (i.e. mapped to a particular SIP registration in sip.conf). What do I need to do in order to get a nice LED or something to flash or light up or whatever on the phone to show that a particular extension on another phone is in use? This is driving me totally insane. Any help would be appreciated! Thanks, Faris. It will then show up on an unused line key That was the key! I had all my line keys in use. I got rid of one that I didn't really need, and BOOM! It works. Thanks! Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] TDM
What if you try Zap instead of ZAP for channel name? []'s MM -Original Message- From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Sun, 28 May 2006 13:33:46 -0400 Delivered: Sun, 28 May 2006 14:28:38 Subject:[Asterisk-Users] TDM It looks OK. Try editing extensions.conf and add an extension in a context that will included when you dial. Try something like this exten = 123,1,Dial(ZAP/g0/1NXXNXX) The open the console and dial 123. This will bypass any funky dialplan issues with FreePBX. If it works, then obviously something is not right in FreePBX. If it doesnt' then that indicates your configuration files need tweaking. Thanks, Steve Curt Shaffer wrote: Here is the output from a dial when starting asterisk with -v. The 1NXXNXX is actually the number not those characters FYI. Thanks -- Executing Macro(SIP/103-a555, dialout-trunk|1|1NXXNXX||) in new stack -- Executing GotoIf(SIP/103-a555, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/103-a555, user-callerid) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing GotoIf(SIP/103-a555, 0?start) in new stack -- Executing Set(SIP/103-a555, REALCALLERIDNUM=103) in new stack -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, AMPUSER=103) in new stack -- Executing Set(SIP/103-a555, AMPUSERCIDNAME=103) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing Set(SIP/103-a555, CALLERID(all)=103 103) in new stack -- Executing NoOp(SIP/103-a555, Using CallerID 103 103) in new stack -- Executing Macro(SIP/103-a555, record-enable|103|OUT) in new stack -- Executing GotoIf(SIP/103-a555, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/103-a555, recordingcheck|20060528-110627|1148832387.1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060528-110627|1148832387.1: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/103-a555, No recording needed) in new stack -- Executing Macro(SIP/103-a555, outbound-callerid|1) in new stack -- Executing GotoIf(SIP/103-a555, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, USEROUTCID=) in new stack -- Executing Set(SIP/103-a555, EMERGENCYCID=) in new stack -- Executing Set(SIP/103-a555, TRUNKOUTCID=) in new stack -- Executing GotoIf(SIP/103-a555, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf(SIP/103-a555, 1?usercid) in new stack -- Goto (macro-outbound-callerid,s,13) -- Executing GotoIf(SIP/103-a555, 1?report) in new stack -- Goto (macro-outbound-callerid,s,15) -- Executing NoOp(SIP/103-a555, CallerID set to 103 103) in new stack -- Executing Set(SIP/103-a555, GROUP()=OUT_1) in new stack -- Executing GotoIf(SIP/103-a555, 0?108) in new stack -- Executing Set(SIP/103-a555, DIAL_NUMBER=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/103-a555, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(SIP/103-a555, OUTNUM=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/103-a555, 0?16) in new stack -- Executing Dial(SIP/103-a555, ZAP/g0/1NXXNXX|120|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(SIP/103-a555, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/103-a555, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(SIP/103-a555, outisbusy|) in new stack -- Executing Playback(SIP/103-a555, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/103-a555, pls-try-call-later) in new stack -- Playing 'pls-try-call-later' (language 'en') == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' in macro 'outisbusy' == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, May 28, 2006 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Connect to the Asterisk console
[Asterisk-Users] SIP and sound breaking
Hi, is there any way to increse the buffer or something to make SIP connections sound better? When I make the calls with Asterisk as a SIP client (through sip.voipbuster.com) the sound quality is poor - constantly breaking (there are few occasional seconds when the sound is OK)- but with any other SIP client on the same network and through sip.voipbuster.com sound is allways OK. Matic ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analogue phone w/ TDM400
Hi, I'm running * 1.2.7.1 on Red Hay 9.0 w/ a TDM 400 2 x FXS, 1 x FXO. I'm using a VTech cordless that makes three short beeps when someone another extension is picked up, presumably this lets you know if someone is trying to listen in.. Everything works, except the VTech now makes the three beeps everytime you try to use it, even if another extension is in use. It seems as though the VTech phone trhinks the TDM400 is another extension - and i guess i can kinda see its point :) It becomes annoying though because I have to put up with these 3 beeps everytime I answer the phone or check messages. The beeps seem to mess up DTMF occassionally too, messing me up when I'm entering my password and so on. Anyone know how to defeat this behaviour? I didn't have this problem on older versions of Asterisl (i.e. 1.1) Thanks, H ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
On Fri, 26 May 2006, Guido Hecken wrote: We had the same problems with some cheap LevelOne Switches. The Snoms rebooted during a call, calls dropped etc. Replacing the switches was the solution. A switch should NEVER cause ANY device to lockup, ever. Period. If a phone locks up / reboots due to something a switch sends, then the phone is faulty. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
On Fri, 26 May 2006, Rich Adamson wrote: Dr. Michael J. Chudobiak wrote: Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. Just curious - what configuration issues did you have in mind? A partial list of issues that we've seen in the last 12 years include: - auto negotiation of duplex settings (mismatch) - spanning tree disabling ports for first 30 seconds after any link state change (some attached devices don't like that) - spanning tree loops that end up isolating devices from the backbone (spanning tree is usually implemented by the manufacture by default) - various switch manufacturers have licensed/implemented cisco's discovery protocol, and the user doesn't realize some equipment attached to such ports actually use the cdp data to change port configuration, while other devices might barf on those packets. - assumptions that all switches operate at wire speeds and buffer packets (eg, no such thing as a switch buffer; packets will be dropped under high load conditions) - distributing vlans across multiple switches where assumptions are made relative to what happens when two or more vlans are transporting traffic volumes that when combined exceed a trunk's port speed (eg, don't forget about broadcast storms). - switch forwarding tables that are too small (eg, workgroup switches) and the table fills, essentially turning the switch into a hub - bad assumptions relative to rate limiting broadcast and multicast packets, and how that impacts normal traffic. - etc, etc. If any of these issues makes a _phone reboot or lockup_ then that is a serious flaw with the phone. I migh expect a cheapy grandstream to have issues but expensive snom should really do better. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
On Fri, 26 May 2006, Remco Barende wrote: There is just no valid reason why the phone would need to lockup or reboot even if the network connection would be problematic, no matter what. That is just poor design, not a feature. I agree 100%. No device should ever lockup or reboot due to a marginal connection. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analogue phone w/ TDM400
Sure that's not the message waiting stuttering indicator? Terrelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hugolivude Sent: Sunday, May 28, 2006 12:40 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Analogue phone w/ TDM400 Hi, I'm running * 1.2.7.1 on Red Hay 9.0 w/ a TDM 400 2 x FXS, 1 x FXO. I'm using a VTech cordless that makes three short beeps when someone another extension is picked up, presumably this lets you know if someone is trying to listen in.. Everything works, except the VTech now makes the three beeps everytime you try to use it, even if another extension is in use. It seems as though the VTech phone trhinks the TDM400 is another extension - and i guess i can kinda see its point :) It becomes annoying though because I have to put up with these 3 beeps everytime I answer the phone or check messages. The beeps seem to mess up DTMF occassionally too, messing me up when I'm entering my password and so on. Anyone know how to defeat this behaviour? I didn't have this problem on older versions of Asterisl (i.e. 1.1) Thanks, H ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hook into authentication
Hi, to increase the security for remote extensions I would like to limit a sip-peer to a specific MAC address. Is it possible to hook into the authentication mechanism in asterisk and allow/deny incoming registrations? cheers urban ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling a person over Internet
On 5/28/06, Matthias Fechner [EMAIL PROTECTED] wrote: If there is a site or howto etc. available it would be a pleasure formy to get something to read :) Go to www.voip-info.org you'll find all you need to know and more. -- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Radius Module
Hi List, I'm looking for a Asterisk radius module ... Anybody has one ? Thanks, Oliver Oliver VermeulenWorld Venture Group Telecom Corporate Address:Str Avionului Nr 35/bl16J/3Bucharest, 014333 RomaniaOffice: +(40)21-569-4700Office2: +(40)31-860-0030Fax: +(40)31-860-0031USA DID: +1 (305)722-1457BE DID: +(32)9-395-5620UK DID:+(44)870-478-8896SIP : [EMAIL PROTECTED]msn: [EMAIL PROTECTED]http://www.wvg-tele.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy Signals
I'd change s,104 to something along the lines of a playback for debugging purposes just to be sure, but it looks as though all of your channels are busy. The way I am reading that is that you have 4 voice channels on your PRI, is that correct? Could you already have 4 simultaneous calls going on when this call is placed?
Other than that, the only thing I can think of is that you are getting an all circuits are busy from your carrier. Haven't yet encountered this, so I don't know how this is reported on a PRI.
I guess one more thing, are you sure you are dialing the correct number?
Maybe one more item, and this goes along with the first. Is TRUNK1 is defined as a group? I ask this because if not, then I believe Asterisk will use only the defined channel and not go to the next available channel. That being the case, you cannot have more than 1 simultaneous call.
On 5/26/06, Kevin Smith [EMAIL PROTECTED] wrote:
Hey everyone,A few employees have noticed some problem here and there when trying tomake outgoing phone calls. After it happens, they try again, and are
able to call through.The dial plan for outbound calling looks like below. Which I know theyare getting to the Congestion part (which explains the busy) but what Ican't seem to figure out is the cause for why they are getting sent there.
exten = s,1,SetCallerID(${ARG1})exten = s,2,Wait(2)exten = s,3,Dial(${TRUNK1}/${ARG2})exten = s,4,Congestion(10)exten = s,104,Congestion(10)The log for a call looked like this
May 26 12:21:08 VERBOSE[6997] logger.c: -- Channel 0/4, span 1 gothangup requestMay 26 12:21:08 VERBOSE[16613] logger.c: -- Zap/4-1 is circuit-busyMay 26 12:21:08 VERBOSE[16613] logger.c: -- Hungup 'Zap/4-1'
May 26 12:21:08 VERBOSE[16613] logger.c: == Everyone is busy/congestedat this time (1:0/1/0)My question is it asterisk having an issue with the PRI or is the PRIreally reporting the number is busy. I know one case like this I was
calling home, and which when I got through to them, they were not evenon the phone. Are there any tests that I can run on the T1 card in theserver to the PRI? Any suggestions would be helpful.Kevin
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Re: [Asterisk-Users] Busy Signals
You could try pri debug span 1 and watch for anything that looks strange.
Lacy Moore - Aspendora wrote:
I'd change s,104 to something along the lines of a playback for
debugging purposes just to be sure, but it looks as though all of your
channels are busy. The way I am reading that is that you have 4 voice
channels on your PRI, is that correct? Could you already have 4
simultaneous calls going on when this call is placed?
Other than that, the only thing I can think of is that you are getting
an all circuits are busy from your carrier. Haven't yet encountered
this, so I don't know how this is reported on a PRI.
I guess one more thing, are you sure you are dialing the correct number?
Maybe one more item, and this goes along with the first. Is TRUNK1 is
defined as a group? I ask this because if not, then I believe
Asterisk will use only the defined channel and not go to the next
available channel. That being the case, you cannot have more than 1
simultaneous call.
On 5/26/06, *Kevin Smith* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hey everyone,
A few employees have noticed some problem here and there when
trying to
make outgoing phone calls. After it happens, they try again, and are
able to call through.
The dial plan for outbound calling looks like below. Which I know they
are getting to the Congestion part (which explains the busy) but
what I
can't seem to figure out is the cause for why they are getting
sent there.
exten = s,1,SetCallerID(${ARG1})
exten = s,2,Wait(2)
exten = s,3,Dial(${TRUNK1}/${ARG2})
exten = s,4,Congestion(10)
exten = s,104,Congestion(10)
The log for a call looked like this
May 26 12:21:08 VERBOSE[6997] logger.c: -- Channel 0/4, span 1 got
hangup request
May 26 12:21:08 VERBOSE[16613] logger.c: -- Zap/4-1 is
circuit-busy
May 26 12:21:08 VERBOSE[16613] logger.c: -- Hungup 'Zap/4-1'
May 26 12:21:08 VERBOSE[16613] logger.c: == Everyone is
busy/congested
at this time (1:0/1/0)
My question is it asterisk having an issue with the PRI or is the PRI
really reporting the number is busy. I know one case like this I was
calling home, and which when I got through to them, they were not even
on the phone. Are there any tests that I can run on the T1 card in the
server to the PRI? Any suggestions would be helpful.
Kevin
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Lacy Moore
Aspendora, Inc.
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Re: [Asterisk-Users] asterisk silence suppression?
Hi, Has there been any improvements to this patch?, what is its state now?. Has anybody tested this?. Any results? I tried the link, seems the site is not up. Where can I download the patch from? -Vij On 3/3/06, Juan Salas [EMAIL PROTECTED] wrote: I will try to test your adaptation. How I congfigureto enable VAD? Regards Jsalas -Mensaje original-De: Moises Silva [mailto:[EMAIL PROTECTED]]Enviado el: Friday, February 17, 2006 11:26 AMPara: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] asterisk silence suppression? The patch you saw is not for the stable branch. Salu2 Jsalas Right, but try using this, i adapted it, no guarantees, i have not made tests, just modified it to apply properly, it would be great if some one can test it:http://chewbacca.ivsol.net/asterisk-1.2.1- silence-suppression-4.patchRegards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk registers but won't complete calls.
Hello,I work for a company that is experimenting with the implementation of Asterisk. We have a VoIP provider that is giving us a demo account with 200 minutes on it. We can register with their service but cannot complete calls with Asterisk. We can use a Grandstream GXP-2000 with the supplied registration info and it does work. We can also register Asterisk with FWD and dial the FWD loopback and it works. We can dial other extensions within Asterisk. The provider said that it appears that our usersname is comming over as the extensions after the / in the register line.We are running Asterisk 1.2.6 and the provider is running Tekelec 9000 switches.Thanks in advance. Be a chatter box. Enjoy free PC-to-PC calls with Yahoo! Messenger with Voice.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk registers but won't complete calls.
We need your dialplan and output from the console to help. Thanks, Steve Totaro Steven Haldeman wrote: Hello, I work for a company that is experimenting with the implementation of Asterisk. We have a VoIP provider that is giving us a demo account with 200 minutes on it. We can register with their service but cannot complete calls with Asterisk. We can use a Grandstream GXP-2000 with the supplied registration info and it does work. We can also register Asterisk with FWD and dial the FWD loopback and it works. We can dial other extensions within Asterisk. The provider said that it appears that our usersname is comming over as the extensions after the / in the register line. We are running Asterisk 1.2.6 and the provider is running Tekelec 9000 switches. Thanks in advance. Be a chatter box. Enjoy free PC-to-PC calls http://us.rd.yahoo.com/mail_us/taglines/postman12/*http://us.rd.yahoo.com/evt=39663/*http://messenger.yahoo.com with Yahoo! Messenger with Voice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Radius Module
How about this one? http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth --VoIP StreetDID origination serviceswith support you can count on!http://www.VoIPstreet.com - Original Message - From: Oliver Vermeulen To: asterisk-users@lists.digium.com Cc: 'Commercial and Business-Oriented Asterisk Discussion' Sent: Sunday, May 28, 2006 4:09 PM Subject: [Asterisk-Users] Asterisk Radius Module Hi List, I'm looking for a Asterisk radius module ... Anybody has one ? Thanks, Oliver Oliver VermeulenWorld Venture Group Telecom Corporate Address:Str Avionului Nr 35/bl16J/3Bucharest, 014333 RomaniaOffice: +(40)21-569-4700Office2: +(40)31-860-0030Fax: +(40)31-860-0031USA DID: +1 (305)722-1457BE DID: +(32)9-395-5620UK DID:+(44)870-478-8896SIP : [EMAIL PROTECTED]msn: [EMAIL PROTECTED]http://www.wvg-tele.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] doubts about asterisk configuration from database
hi, I want to complete asterisk configuration from database(MYSQL),now I come across some doubts: 1. http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+databasediff2=3 says that Dynamic 'friends' (Asterisk v1.0.*) and the number of options supported by this 'MySQL_Friends' system is currently very limited,at the same time I find asterisk-1.2.* don't provide this functions,why? for other factors? 2. If I want to do it with asterisk 1.2.*, do I need to add only chan_sip.c and Makefile files in channels directory? 3.Is there any other way to complete asterisk configuration from database? thanks ! BEST REGARDS! Sharon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] doubts about asterisk configuration from database
hello,, Yes, asterisk can use realtime mode On 5/29/06, 吴应芳 [EMAIL PROTECTED] wrote: hi, I want to complete asterisk configuration from database(MYSQL),now I come across some doubts: 1. http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+databasediff2=3 says that Dynamic 'friends' (Asterisk v1.0.*) and the number of options supported by this 'MySQL_Friends' system is currently very limited,at the same time I find asterisk-1.2.* don't provide this functions,why? for other factors? 2. If I want to do it with asterisk 1.2.*, do I need to add only chan_sip.c and Makefile files in channels directory? 3.Is there any other way to complete asterisk configuration from database? thanks ! BEST REGARDS! Sharon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeffery `∧ ∧�� ミ^r^ミ灬)~ iaxtel Num: 1-700-576-1311 fwdnet Num: 728150 http://www.diaip.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] doubts about asterisk configuration from database
3.Is there any other way to complete asterisk configuration from database? Have a look at this : http://www.voip-info.org/wiki-Asterisk+RealTime hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Go2call Configuration
Hello, Does anyone had tried to configure asterisk server as sip client to connect to go2call service.? If it works can you share your sip.conf and extension.conf configurations. Thanks, Leo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hook into authentication
to increase the security for remote extensions I would like to limit a sip-peer to a specific MAC address. Is it possible to hook into the authentication mechanism in asterisk and allow/deny incoming registrations? This would be only mildly useful on the same subnet and completely useless over the internet. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk registers but won't complete calls.
The sip debugging info is here http://pastebin.com/744005, the sip.conf, extensions.conf and consoleoutput are here http://pastebin.com/744065.Thatnk you, StevenSteve Totaro [EMAIL PROTECTED] wrote: We need your dialplan and output from the console to help.Thanks,Steve TotaroSteven Haldeman wrote: Hello, I work for a company that is experimenting with the implementation of Asterisk. We have a VoIP provider that is giving us a demo account with 200 minutes on it. We can register with their service but cannot complete calls with Asterisk. We can use a Grandstream GXP-2000 with the supplied registration info and it does work. We can also register Asterisk with FWD and dial the FWD loopback and it works. We can dial other extensions within Asterisk. The provider said that it appears that our usersname is comming over as the extensions after the / in the register line. We are running Asterisk 1.2.6 and the provider is running Tekelec 9000 switches. Thanks in advance. Be a chatter box. Enjoy free PC-to-PC calls with Yahoo! Messenger with Voice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR sounds not on certain inbound route
Got 1 issue I can't seem to knock out of this particular box. The IVR works fine on the zap channels and the incoming SIP routes. But coming in via the IAX2 route leaves me with a silent phone. The prompts all work still letting me navigate the menu. But just can't hear anything. This is with [EMAIL PROTECTED] 2.8 (Asterisk 1.2.7.1, with FreePBX 2.1.0 also installed) Any thoughts on where to start looking to solve this? Console shows it all executing fine. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hook into authentication
Henry J. Cobb wrote: to increase the security for remote extensions I would like to limit a sip-peer to a specific MAC address. Is it possible to hook into the authentication mechanism in asterisk and allow/deny incoming registrations? This would be only mildly useful on the same subnet and completely useless over the internet. -HJC I think it would work just fine over the internet using a bridged VPN. Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Radius Module
Never used it, but I knew some sort of Asterisk radius thing existed so I searched the Wiki for it and replied. Sorry it doesn't work for you, good luck with your search. --VoIP StreetDID origination serviceswith support you can count on!http://www.VoIPstreet.com - Original Message - From: Oliver Vermeulen To: 'VoIP Street .com' Sent: Sunday, May 28, 2006 9:13 PM Subject: RE: [Asterisk-Users] Asterisk Radius Module I need full AAA That just dose authentication.. But i will try it lets see ;) Are you using it ? Oliver From: VoIP Street .com [mailto:[EMAIL PROTECTED] Sent: Monday, May 29, 2006 3:59 AMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk Radius Module How about this one? http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth --VoIP StreetDID origination serviceswith support you can count on!http://www.VoIPstreet.com - Original Message - From: Oliver Vermeulen To: asterisk-users@lists.digium.com Cc: 'Commercial and Business-Oriented Asterisk Discussion' Sent: Sunday, May 28, 2006 4:09 PM Subject: [Asterisk-Users] Asterisk Radius Module Hi List, I'm looking for a Asterisk radius module ... Anybody has one ? Thanks, Oliver Oliver VermeulenWorld Venture Group Telecom Corporate Address:Str Avionului Nr 35/bl16J/3Bucharest, 014333 RomaniaOffice: +(40)21-569-4700Office2: +(40)31-860-0030Fax: +(40)31-860-0031USA DID: +1 (305)722-1457BE DID: +(32)9-395-5620UK DID:+(44)870-478-8896SIP : [EMAIL PROTECTED]msn: [EMAIL PROTECTED]http://www.wvg-tele.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls connected, but no audio
The asterisk host is connected directly to the internet, the phones I am having issues with are behind NAT, but I'm only having issues with some of them. Most specifically the phones on my linksys PAP2 adapter. NAT at the remote location is provided via a standard out of the box config of a Linksys WRT54GS router. Here are the settings for the PAP2: [pap2] type=friend secret=something qualify=yes nat=yes host=dynamic canreinvite=no context=private callgroup=6 pickupgroup=6 callerid=name 1234567890 disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 This is a situation where I do have multiple SIP devices behind NAT, tell me more about using different port numbers for different devices, and what other things should I look out for? Thanks Miles Steve Totaro wrote: You need to describe your NAT setup more. One thing to try is to set qualify to yes or a short number. Essentially a keepalive for any routers in the middle. If you have multiple phones behind a remote NAT, make sure they are using different ports. Miles Scruggs wrote: Using sip connections some peers are not able to transmit or recieve audio. All peers are setup the same aside from the NAT settings. The call will go through, called device will ring, but when it answers there is no audio connection. From the callee, they will not here the rings, only silence when they dial the phone. The kicker is that sometimes it will work, and other times it will not. Miles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] doubts about asterisk configuration from database
another questions！ According asterisk realtime sip webpage，I had done following steps： (1) Make, make install asterisk-addons then copy res_mysql.conf.sample to res_mysql.conf and edit the res_mysql.conf with my databases parameter (2) Edit extconfig.conf ---add sip.conf = mysql,asterisk,sipfriends (3) Create a sipfriends table in asterisk database and register some sip phone into the table (4) then restart asterisk... but ... *CLI May 29 12:50:08 WARNING[22360]: config.c:920 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available ask: MYSQL given database driver would be ? thanks~~ hello,, Yes, asterisk can use realtime mode On 5/29/06, 吴应芳 [EMAIL PROTECTED] wrote: hi, I want to complete asterisk configuration from database(MYSQL),now I come across some doubts: 1. http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+databasediff2=3 says that Dynamic 'friends' (Asterisk v1.0.*) and the number of options supported by this 'MySQL_Friends' system is currently very limited,at the same time I find asterisk-1.2.* don't provide this functions,why? for other factors? 2. If I want to do it with asterisk 1.2.*, do I need to add only chan_sip.c and Makefile files in channels directory? 3.Is there any other way to complete asterisk configuration from database? thanks ! BEST REGARDS! Sharon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeffery `∧ ∧�� ミ^r^ミ灬)~ iaxtel Num: 1-700-576-1311 fwdnet Num: 728150 http://www.diaip.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hook into authentication
On Sun, 2006-05-28 at 23:41 -0400, Steve Totaro wrote: Henry J. Cobb wrote: to increase the security for remote extensions I would like to limit a sip-peer to a specific MAC address. Is it possible to hook into the authentication mechanism in asterisk and allow/deny incoming registrations? This would be only mildly useful on the same subnet and completely useless over the internet. -HJC I think it would work just fine over the internet using a bridged VPN. even on a local network this can be forged. If you cant control the device that sends this information it is user supplied data, even over a vpn (which uses a virtual interface not the physical one). It has the same value as any user supplied data - other than perhaps its additional data which makes guessing slightly harder. TLS might be a better way to go since it would require a certificate that you can control the issuance of, but that certificate can be stolen and the remote end point would need to support the same scheme that you use (fortunately there are standards that make this easier with some devices but most dont implement this). A vpn would provide security in that it would make it harder for someone to eavesdrop on the auth and attempt to derrive the password, however there is overhead associated with that. At least 1 IP packet per real packet (sometimes more) on the network side, and the crypto parts on the cpu side. For the server you would want to have a hardware based crypto card to deal with the VPN connections... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] doubts about asterisk configuration fromdatabase
Hello: Do you need install Mysql-devel. Best Regards - Original Message - From: 吴应芳 [EMAIL PROTECTED] To: Asterisk Users Mailing List - No asterisk-users@lists.digium.com Sent: Monday, May 29, 2006 12:04 AM Subject: Re: Re: [Asterisk-Users] doubts about asterisk configuration fromdatabase another questions！ According asterisk realtime sip webpage，I had done following steps： (1) Make, make install asterisk-addons then copy res_mysql.conf.sample to res_mysql.conf and edit the res_mysql.conf with my databases parameter (2) Edit extconfig.conf ---add sip.conf = mysql,asterisk,sipfriends (3) Create a sipfriends table in asterisk database and register some sip phone into the table (4) then restart asterisk... but ... *CLI May 29 12:50:08 WARNING[22360]: config.c:920 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available ask: MYSQL given database driver would be ? thanks~~ hello,, Yes, asterisk can use realtime mode On 5/29/06, 吴应芳 [EMAIL PROTECTED] wrote: hi, I want to complete asterisk configuration from database(MYSQL),now I come across some doubts: 1. http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+databasediff2=3 says that Dynamic 'friends' (Asterisk v1.0.*) and the number of options supported by this 'MySQL_Friends' system is currently very limited,at the same time I find asterisk-1.2.* don't provide this functions,why? for other factors? 2. If I want to do it with asterisk 1.2.*, do I need to add only chan_sip.c and Makefile files in channels directory? 3.Is there any other way to complete asterisk configuration from database? thanks ! BEST REGARDS! Sharon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeffery `∧ ∧�� ミ^r^ミ灬)~ iaxtel Num: 1-700-576-1311 fwdnet Num: 728150 http://www.diaip.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
