[asterisk-users] Providers that offer contract
Hello,I'm interested in contract some providers VoIP that support IAX2 and that offers the possibility of contract for outbound(termination) calls only. I'm not interested in DIDS, only termination calls.Thanks in advance ___ --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] H323
On Sunday 27 August 2006 10:40, Mohammad Salaque wrote: any one try that with g723 codec? We use G.723.1, and it works well. My only problem is the bridging time (after pickup) takes at least 5 seconds. But this happenned even before Asterisk was in the picture, so I'm guessing it's the remote H.323 gateways (unless someone else has experienced this). Cheers, Mark. pgpA2t5GdGVZR.pgp Description: PGP signature ___ --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] Re: GSM gateway and FXO ATA
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 2) If the phone is answered on the first ring the call goes off to la la land. Explaining to users (or myself) that you need to wait for the second audible ring on the handset's before answering isn't acceptable. Hi Marty! Can you tell me more about this? You mean when call from SIP goes to FXO port, if phone attached on FXO port answers after the first ring (before second) ATA will always stop to work? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Selecting outbound trunk
On Tue, Aug 29, 2006 at 02:18:32PM +1000, Devraj Mukherjee wrote:
The simplest way I can think of solving this is using prefixes, so
someone appends a 0 or 1 and the dialplan puts the call through the
selected trunk, where 0 being voip and 1 being PSTN.
Whats wrong with something like this :
exten = _91X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _92X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _93X.,1,Dial(IAX2/iaxprov/${EXTEN:2})
Users would dial 91 to dial outbound on SIP Provider 1, 92 for
outbound on SIP Provider 2, and 93 for outbound on IAX. Personally
I use 9X for automatic routing (along with some sane forced routing,
ie local, emerg calls etc), and am planning on using 8X for manual
forced routing.
I have figured out how to use a Substring like function to extract the
number out of the dialed extension. My question is how do I make a
decision in the dialplan to dynamically select a trunk for the call?
Is there a SetIf function or an If function by itself?
Checkout the command GotoIf()
Heres an example that I use to in my exten Macro, that does
slightly different things depending on the number range the
extension dialed is from:
[macro-exten]
exten = s,1,GotoIf($[${ARG1:0:1} = 1]?11:21) ; Did we call a real ext ?
exten = s,11,SetVar(TODIAL=${ARG2}/${ARG1}); Yes so we have the ext
exten = s,12,Goto(91) ; Jump to Dial() routint
exten = s,21,GotoIf($[${ARG1:0:2} = 20]?31:41) ; Did we call a virt or soft ?
exten = s,31,SetVar(VMBOX=${ARG1}) ; Virt, So vm is the same
exten = s,32,SetVar(TODIAL=${VIRT[${ARG1}]}) ; Grab the list of real exts
exten = s,33,Goto(91) ; Jump to the dial routine
exten = s,41,SetVar(VMBOX=20${ARG1:1:1}) ; Soft, So vm is the virt
exten = s,42,SetVar(TODIAL=${ARG2}/${ARG1}); But it is a real ext
exten = s,43,Goto(91)
exten = s,91,Dial(${TODIAL},25,Tt)
exten = s,92,GotoIf($[${ARG1:0:1} = 2]?93:94) ; Do we need to handle vm ?
exten = s,93,GoSub(s-${DIALSTATUS},1)
exten = s,94,Hangup()
exten = s-NOANSWER,1,Voicemail(u${VMBOX}) ; Virtual extensions have
exten = s-BUSY,1,Voicemail(b${VMBOX}) ; VM, so transfer caller
exten = s-CHANUNAVAIL,1,Voicemail(u${VMBOX}) ; Offline, so transfer call
I have a dialplan where 1xx are real extensions, with no voicemail,
20x are virtual extensions, identified with an induvidual, with voicemail, and
2xy are extensions assoiated with the same induvidual as the virtual
number (ie 21x are all linked to 201 etc..)
HTH
Iain
___
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[asterisk-users] Re: Nokia E60/61/70 and SIP
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 1. you need qualify set as the wifi radio on the phone sucks big oranges What is qualify set? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] SIP Error message
Does anyone know what causes the following error message means. Aug 29 10:11:08 WARNING[30913]: chan_sip.c:2561 sip_write: Asked to transmit frame type 256, while native formats is 8 (read/write = 256/256) I've not yet tracked down what is causing this, but I get a lot of them at the same time. It may be related to a nokia E60 trying to pick up the call. (It's hard to tell atm.) ___ --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] misdn-init.conf card parameter for a monoBRI
Patrick, thank you for your help, now I installed pciutils which is what I couldn't find so misdn-init works Unfortunately aftere a misdn-init start I get invalide module format errors. I search on internet hoping to solve it. Thanks again! Giorgio Incantalupo Patrick wrote: On Mon, 2006-08-28 at 09:08 +0200, Giorgio Incantalupo wrote: Hi Patrick, thanks for your answer. Unfortunately I cannot use misdn-init command because my distro has not the lspci command misdn-init is based on. That's why I want to bypass it. I'm doing all this mess because Debian Sarge installer does not work with new asus motherboards, so I'm trying to use the testing Etch version. I tryed monoBRI as parameter but I do not know if it is the right choice for a monoBRI card. Why not use a distro that supports your motherboard? Why not just install pciutils manually on your box so you have lspci and the misdn-init script can do its job? Anyway, I don't know what card you have and even if I did I would not know what module it should use and even if I did I would not know what layermask en other misdn parameters to specify because that's what misdn-init scan figures out for me... Regards, Patrick ___ --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 -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.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] Re: Nokia E60/61/70 and SIP
qualify=yes Put in in the sip.conf file in the configuration section for the specific phones. On Tue, 2006-08-29 at 09:50, Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 1. you need qualify set as the wifi radio on the phone sucks big oranges What is qualify set? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] misdn-init.conf card parameter for a monoBRI
what about a subscription on the Misdn-asterisk@lists.beronet.com mailing list! http://lists.beronet.com/cgi-bin/mailman/listinfo/misdn-asterisk Regards Kai ___ --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] Re: DNS
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Have you tried setting timeout, attempts and rotate in resolv.conf? Can you please tell me more about this? How to do it and what would I achieve with that? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] sip giving problems, please help.
Can anyone explain this:- I am getting this on virtually every call. -- Executing Dial(SIP/15552830438-990b, SIP/[EMAIL PROTECTED]|) in new stack -- Called [EMAIL PROTECTED] -- SIP/192.168.1.1-859d is making progress passing it to SIP/15552830438-990b -- SIP/192.168.1.1-859d is ringing -- SIP/192.168.1.1-859d is making progress passing it to SIP/15552830438-990b -- SIP/192.168.1.1-859d answered SIP/15552830438-990b -- Attempting native bridge of SIP/15552830438-990b and SIP/69.54.75.50-859d == Spawn extension (macro-dialroute, s, 14) exited non-zero on 'SIP/15552830438-990b' in macro 'dialroute' == Spawn extension (macro-dialroute, s, 14) exited non-zero on 'SIP/15552830438-990b' Aug 29 15:10:12 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:12 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Aug 29 15:10:12 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:13 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:13 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Aug 29 15:10:13 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:13 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:13 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Aug 29 15:10:13 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:14 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:14 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Aug 29 15:10:14 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:14 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:14 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Aug 29 15:10:14 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:14 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:14 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Aug 29 15:10:14 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:15 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:15 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Aug 29 15:10:15 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:15 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:15 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Aug 29 15:10:15 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:17 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:17 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Aug 29 15:10:17 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:17 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:17 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Aug 29 15:10:17 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:21 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:21 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Aug 29 15:10:21 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:21 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:21 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Aug 29 15:10:21 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:25 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:25 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Aug 29 15:10:25 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:25 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:25 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Aug 29 15:10:25 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:29 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:29 ERROR[30036]: chan_sip.c:11324
[asterisk-users] does misdn-mqueue work if compiled with gcc 4?
Hi, does misdn-mqueue work if compiled with gcc 4? I get some errors when trying to load misdn drivers: FATAL: Error inserting mISDN_core (/lib/modules/2.6.15-1-486/extra/mISDN_core.ko): Invalid module format WARNING: Error inserting mISDN_core (/lib/modules/2.6.15-1-486/extra/mISDN_core.ko): Invalid module format TIA Giorgio Incantalupo -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.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
[asterisk-users] transform bridged call into a conference
Hi! I wonder if it is possible to transform a bridged call into a conference. E.g. phone 1 calls phone 2 (normal bridged call with Dial()). Further phone 3 wants to join? Is this possible? Can you please refer me the proper applications? thanks klaus ___ --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 1.2.4 I hear other party's voice only when I speack need help
I have this problem with Asterisk 1.2.4 I hear other party's voice only when I speack or i make some noise. Otherwise i hear nothing. Futhermore every time i receive a call , this message is displayed : -- Started music on hold, class 'my_class', on SIP/ some random public ip address -08222740 any help? ___ --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 - Comfort Noise
Hi, Does anybody know if asterisk 1.4 will support comfort noise? Or if there is a patch for it now? If it will be in 1.4 any idea of release date? Thanks, Dean Bath. ___ --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] Re: H264
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Actually you need to use the SVN version of Asterisk to support H264 video. It should be part of the planned 1.4 release. When can I expect 1.4 release? Will it be this year? First quarter of 2008? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Mix Monitor call quality
Hi trying to record calls using mixmonitor, but I'm having problems with call quality the call seems OK but then it drops frames with silence ( for less than 0.5 seconds) then call continues All I'm doing is bridging two zap channels and recording no transcoding or changes to the channels Asterisk version 1.2.10 also under certain conditions Asterisk just stops any advice would be appreciated Thanks Robb ___ --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] does misdn-mqueue work if compiled with gcc 4?
gcc -v : gcc version 4.1.0 no problems using latest stuff from beronet/downloads/misdn_queue.stuff suse standard kernel regards KAI ___ --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] does misdn-mqueue work if compiled with gcc 4?
Hi Kai, thanks, maybe I used the wrong kernel-headers, do not know. I can compile misdn-queue but misdn-start (after scan + config) tells me my module has an invalid format. Thanks Giorgio Incantalupo Kai Ober wrote: gcc -v : gcc version 4.1.0 no problems using latest stuff from beronet/downloads/misdn_queue.stuff suse standard kernel regards KAI ___ --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 -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.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
[asterisk-users] working chan_bluetooth enviroment
Hello, can anyone tell me which of the thousand chan_bluetooth development is the latest one and where can I get it? Which version of Asterisk is supposed to be the best choice with bt-support? Thanks alot 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] Handytone 286 T.38 SDP parameters
Hello, I installed trunk version of asterisk. I'm testing T.38 fax. My configuration is FaxMachine1--HandyTone286--asterisk--spa2100--FaxMachine2. When I send fax from FaxMachine1, I cannot see any T.38 SDP parameters. Any idea is appreciated. Thanks. Jason. [general] bindport=5060 bindaddr=0.0.0.0 allow=all t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] does misdn-mqueue work if compiled with gcc 4?
Giorgio Incantalupo schrieb: Hi Kai, thanks, maybe I used the wrong kernel-headers, do not know. Did you change them? Now everything works fine? regards ___ --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] transform bridged call into a conference
you're searching for 3pty... which TECH do ya use? zap/misdn/bristuff/sip/ or do ya look for a generel solution? zap does this by itself! there is a possibility do throw calle 2 into an conference, get calle 3 throw it into conference, and them self join the conference. Klaus Darilion schrieb: Hi! I wonder if it is possible to transform a bridged call into a conference. E.g. phone 1 calls phone 2 (normal bridged call with Dial()). Further phone 3 wants to join? Is this possible? Can you please refer me the proper applications? thanks klaus ___ --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] Cisco 7970
Is 8.0.2.SR1 still the latest firmware? I still haven't managed to do anything useful with that weary expensive phone. It still only receives and places calls, nothing else. Is there any exciting feature that can work with asterisk and SIP firmware? Has anybody managed to do anything of the following: - my screensaver - picture of calling person - External directory - dialplan.xml - How to setup hinting (Multiple Call Appearance) - How to login true ssh? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 t38passthrough
Finally it's working! I was doing everything well, the problem was that neither the latest branch of Asterisk-t38 worked (http://svn.digium.com/svn/asterisk/team/group/t38passthrough/), neither the patched version of Asterisk 1.2.7. Only the branch of Asterisk-t38 made from source version 1.2.4 works for me. If anyone deployed with success those versions that I didn't make to work, please tell me! Regards, Ricardo. Ricardo Carvalho wrote: Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstream ATA to connect Asterisk to a Fax machine. Every time I send a fax, it gets sent using codec G711, and never T.38. I added the following parameters in the [general] section as well as in device configurations: t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes I think that's the only thing that is needed to do to enable T.38 pass through... Why does Asterisk keeps sending in G711? Any help? Regards, Ricardo. ___ --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] transform bridged call into a conference
Kai Ober wrote: you're searching for 3pty... which TECH do ya use? zap/misdn/bristuff/sip/ or do ya look for a generel solution? yes - general zap does this by itself! how? there is a possibility do throw calle 2 into an conference, get calle 3 throw it into conference, and them self join the conference. ok - how does it work? with app_chanredirect? regards klaus Klaus Darilion schrieb: Hi! I wonder if it is possible to transform a bridged call into a conference. E.g. phone 1 calls phone 2 (normal bridged call with Dial()). Further phone 3 wants to join? Is this possible? Can you please refer me the proper applications? thanks klaus ___ --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: [asterisk-users] Handytone 286 T.38 SDP parameters
Dear Jason, Only version 1.2.4 of the Asterisk-t38 branch worked for me. Matybe your problem could be that. Try to install version 1.2.4, it should work. Regards, Ricardo. Jason Kim wrote: Hello, I installed trunk version of asterisk. I'm testing T.38 fax. My configuration is FaxMachine1--HandyTone286--asterisk--spa2100--FaxMachine2. When I send fax from FaxMachine1, I cannot see any T.38 SDP parameters. Any idea is appreciated. Thanks. Jason. [general] bindport=5060 bindaddr=0.0.0.0 allow=all t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Handytone 286 T.38 SDP parameters
Also try to use Kapanga softphones (www.kapanga.net) that are able to send T.38 faxes. I've tested one Grandstream handytone 386 ATA device, and it doesn't work well with T.38. I only have been able to do T38 fax pass-through between two Kapanga softphones with Asterisk in the middle. You should try that too. If well succeeded in that tests, only then you should try with ATAs. That's what I'm going to do now! In every context of every ata in sip.conf you should also add those lines: (not only in general section) t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no Regards, Ricardo. Ricardo Carvalho wrote: Dear Jason, Only version 1.2.4 of the Asterisk-t38 branch worked for me. Matybe your problem could be that. Try to install version 1.2.4, it should work. Regards, Ricardo. Jason Kim wrote: Hello, I installed trunk version of asterisk. I'm testing T.38 fax. My configuration is FaxMachine1--HandyTone286--asterisk--spa2100--FaxMachine2. When I send fax from FaxMachine1, I cannot see any T.38 SDP parameters. Any idea is appreciated. Thanks. Jason. [general] bindport=5060 bindaddr=0.0.0.0 allow=all t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [asterisk-users] transform bridged call into a conference
zap does this by itself! how? threewaycalling=yes in zapata.conf there is a possibility do throw calle 2 into an conference, get calle 3 throw it into conference, and them self join the conference. ok - how does it work? with app_chanredirect? this was used to run in astersik 1.09... actually it does not, and i have no really running version of this http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro idea at the moment. Maybe you get this on the run... lemme plz know! Regards KAI ___ --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] does misdn-mqueue work if compiled with gcc 4?
Hi Kai, the problem is to find the right kernelI used apt-get *install* kernel-headers-*`**uname* -r*`* -y but it seems not to be the right one...Even zaptel is not working: FATAL: Error inserting zaptel (/lib/modules/2.6.15-1-486/misc/zaptel.ko): Invalid module format I've always used Debian Sarge, now I'm oblidged to use Etch testing because Sarge installer doesn't work with new hardware and I do not want to change distro. Thank you. Giorgio Incantalupo * * Kai Ober wrote: Giorgio Incantalupo schrieb: Hi Kai, thanks, maybe I used the wrong kernel-headers, do not know. Did you change them? Now everything works fine? regards ___ --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] MOR and MCC - billing solutions for Asterisk released
Hello, Kolmisoft: http://www.kolmisoft.com released new versions of MCC and MOR - Billing solutions for Asterisk PBX MOR - is new product, it's MCC v2. Rewritten to support MySQL and is based on Ruby on Rails. ChangeLog for MCC: DB: New fields: providers.enable_cid_prefix varchar providers.disable_cid_prefix varchar providers.min_time integer default 1 providers.increment integer default 1 cids.enable_to_provider boolean default true cids.nat boolean default true cids.voicemail boolean defaut true cids.voicemail_psw charvar default '' rates.connection_fee double precision default 0 calls.rate double precision default 0 calls.user_connection_fee double precision default 0 calls.rate_connection_fee double precision default 0 users.first_name varchar users.last_name varchar users.min_time integer default 1 users.increment integer default 1 Fixed 386 code from Slovakia to Slovenia APP: Fixed bug with transfers - thanks German Aracil - suspended, needs more testing Changes to support CID manipulation - sponsored by Imre Csaba Varasdy Changes to support connection_fee based on rates(destinations) Rate, user_connection_fee, rate_connection_fee now are added to calls data Min_time and increment for billed time now taken from db, not conf file GUI: Fixed bug with email exists message Added Spanish translation - thanks German Aracil Added Hungarian translation - thanks Imre Csaba Varasdy Added German translation - thanks Inga A. Added Albanian translation - thanks Arben Myrtaj Now possible to assign connection_fee for rate(destination) User name split into First Name and Last Name Voicemail support in autoconfiguration, reachable by *98 for VoIP users User/admin can change cid/nat/voicemail/voicemail password for user's every CID (which supports autoconf.) under his details and when registering Possible to change call's status from processed to not (Changes color in GUI) and hide 'processed' calls in invoices. Possible to hide calls shorter than 'x' seconds. User can see his payments When registering, possible to set address like: http://mcc.company.com/register.php?ref=27, then referrer's field will be filled automatically Register authentication with noisy picture to prevent bot-registering Registration process reworked, check more here Reseller's CID's moved to new section - Devices Now various billing options (1/1, 6/6, 30/6) could be set per user basis - sponsored by Patrick Cardozo ASR (Average Success Rate) / ALOC (Average Length of Call) counting - sponsored by Patrick Cardozo New window to check CIDs and Extensions New values to define.php $USE_PROCESSED_CALLS $REG_ADDITIONAL - Additional info in registrtion page (like come visit our VoIP store) Regards, Midnaugas Kezys ___ --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] IP phone with 2 ethernet jacks
Matt Birmingham wrote: Can you supply any information/links as to how you got this working? Sure. I followed the instructions here: http://www.voip-info.org/wiki/view/Asterisk+presence However, there was one important fact missing. By default the presence feature is turned off in the Polycom's (at least that was the case with my Polycom 501's). It needs to be turned on before any of the other settings take effect. To turn it on you can edit sip.cfg. Look near the end for the feature section. Feature 1 is presence, changed feature.1.enabled from 0 to 1. This is also covered in section 4.6.1.23 of the Polycom admin manual. John ___ --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 Development and Release Cycle
Do Asterisk team care about this anymore? Whole text can be found here: http://www.asterisk.org/developers/releasecycle -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Can I increase DTMF sensitivity?
Steve Edwards wrote: I have a client that is complaining that they are having trouble with Asterisk not reliably recognizing DTMF from some phones. The calls are being delivered by ATT Qwest over PRI to te410p's with firmware that was upgraded about 10 months ago. Can I tweak the amplitude and duration of what Asterisk considers a keypress? (I'm in the US if it matters.) What will happen if I tweak too much? Can I measure the amplitude and duration of a keypress to help identify the phones? Tweaking is almost never the answer. The DTMF detector will detect a wide range of mangled DTMF tones correctly. If it can't detect what you are receiving on a clean digital line, something serious is wrong. The likely candidates are: - You have frame slips, because your T1 is not clocked properly, or because your PCI bus is not keeping up. The DTMF detector was modified to be somewhat tolerant of hiccups in the data, but it can still be fooled, especially if a big chunk of audio goes missing. - You have the gain turned up so far, the signal is clipping. It sounds like that shouldn't be an issue, since if the DTMF clips the voice must sound awful. Well, I get regular wave files sent to me where the gain *is* set that high, and people say the voice has no problem. - People are using really really crappy phones. There are quite a few phones that just never dial reliably. Its hard to see why people put up with them, but they do. Steve ___ --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 codec strangeness
Hi All, I have two peers (call then peerA and peerB) on my server, both can accept g711, g729 and g723. However, when peerA initiates a request, asterisk decides to transcode g729 into ulaw when peerB could very well use g729... This behavior isn't very scalable (transcoding is CPU expensive) and also it's better to minimize the amount of transcoding wherever possible. Is there a way I can fix this? NB: if i set disallow = all and allow=g729 on peerB it all works fine, but then if peerA decides to send ulaw I'm transcoding again... Cheers, Jean-Michel. ___ --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] playback() breaks audio in zap-iax-iax-zap channel
all,i'm having a problem where a callee is unable to hear the caller,but the caller can hear the callee perfectly well.the call is processed as follows:1. caller dials into zap-1 via a pri line ( using chan_zap ) 2. zap-1 connects to iax-1 over an iax channel Dial(IAX2/[EMAIL PROTECTED])3. iax-1 plays the user a prompt using Playback() Playback(beep)4. iax-1 dials zap-2 over an iax channel Dial(IAX2/[EMAIL PROTECTED]/home)5. zap-2 dials an outside number using chan_zap Dial(ZAP/g1/2125551212) here's the wierd part: if i remove the Playback() from the iax-1 dialplan, both the caller and the callee can hear one another fine.but for some reason, though, after Playback() is called in the dialplan,the connection breaks down.i've run ethereal on all the servers and packets _are_ going back and forth as expected.( i should mention it's not just Playback(), i replaced Playback()with Wait() and had the same problem. )i also changed the diaplan on zap-2 to record the call instead dialing out. Record(/tmp/incoming_call.wav) the behavior with Record() was consistent with the Dial() problem:when Playback() was called on iax-1, the recorded file was empty ( except for the RIFF header ), and when i removed Playback(), the file recorded properly.i'm guessing the problem is somewhere in chan_iax2.maybe a state flag? or something? i'm kinda lost now...any tips or pointers would be greatly appreciated!jeff oconnell [EMAIL PROTECTED] ___ --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] does misdn-mqueue work if compiled with gcc 4?
Giorgio Incantalupo schrieb: Hi Kai, the problem is to find the right kernelI used apt-get *install* kernel-headers-*`**uname* -r*`* -y so the only i can tell is: - my kernel is 2.6.16.13-4-default (meaning suse 10.1 default) - installed latest zap and libpri packages from asterisk-org - installed asterisk 1.2.10 (from ftp) - got the latest from http://beronet.com/downloads/install-misdn-mqueue.tar.gz ___ --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] Re: transform bridged call into a conference
Hi! I wonder if it is possible to transform a bridged call into a conference. E.g. phone 1 calls phone 2 (normal bridged call with Dial()). Further phone 3 wants to join? Is this possible? Can you please refer me the proper applications? What I am doing: First, redirect to bridged calls to a conference room. I'm doing so using Redirect via Manager. You can redirect both channels at once with just one Redirect command. Second, make the call to the third party and send it to aN extension associated to the same conference room. -- Atly. Alvaro Palma. ___ --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] Re: Nokia E60/61/70 and SIP
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... qualify=yes Put in in the sip.conf file in the configuration section for the specific phones. I don't think he thought on that. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Grabbing authenticated mailbox value from VoicemailMain()
Still stumped on this one. Am I going to have to scrap the idea of using the outgoing call function of VoicemailMain as a way to authenticate and process an outgoing call (i.e. should I be building something specific using VMAuthenticate) ? N. On Mon, 28 Aug 2006 09:56:17 -0400, sip wrote Is there a way for me to grab the value of the authenticated mailbox from the VoicemailMain() app? If a user calls in to the main extension, enters in a mailbox and password and authenticates, I want to know what mailbox number was authenticated for use in another app. For instance, when the AdvancedVoicemail stuff is being used, and a user is dialing out from within voicemail (option 4 to place an outgoing call), I currently have that dialout going to a particular context and within that context, I'd like to be able to set the new callerID to be the callerID of the authenticated user (not the original caller). Any ideas? ___ --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] Analyze core file prodeced after safe_asterisk crashh
Hi, I have a Call Center running with safe_asterisk script.When Asaterisk crash produce a core file but I don´t know how analyze it!Any ideas?? ___ --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] does misdn-mqueue work if compiled with gcc 4?
Hi Kai, it works!!! Not with gcc 4.1 but with 4.0! Giorgio Incantalupo. Kai Ober wrote: Giorgio Incantalupo schrieb: Hi Kai, the problem is to find the right kernelI used apt-get *install* kernel-headers-*`**uname* -r*`* -y so the only i can tell is: - my kernel is 2.6.16.13-4-default (meaning suse 10.1 default) - installed latest zap and libpri packages from asterisk-org - installed asterisk 1.2.10 (from ftp) - got the latest from http://beronet.com/downloads/install-misdn-mqueue.tar.gz ___ --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 1.2.4 I hear other party's voice only when I speack need help
Hi Tommaso, have you tried to search for noise suppression? I remember some phone has a function to automatically suppress it so the caller does not hear anything and thinks the other party has hung up. Giorgio Incantalupo Tommaso Calosi wrote: I have this problem with Asterisk 1.2.4 I hear other party's voice only when I speack or i make some noise. Otherwise i hear nothing. Futhermore every time i receive a call , this message is displayed : -- Started music on hold, class 'my_class', on SIP/ some random public ip address -08222740 any help? ___ --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 Manager Interface Question
100ms On 8/28/06, Roi Stork [EMAIL PROTECTED] wrote: The version that I'm using is 1.2.7.1. What is the default value of writetimeout in manager.conf? ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Detect if cell phone or users
Hi List, I have sprint pcs cellular service and I discoverd thatI am able to send a text message to a landline. If I do I get an SMS from the saying I sent a text message to a landline. Then the landline that I sent a text message to gets a call with my message (text to speach). I was wondering if there was any way for me to detect if a number is a mobile phone or a landline. Is this something that only cellular providers can do or can I have asterisk do it (I asume I would need to create a patch). Thanks a lot. Dovid ___ --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] Connecting two asterisk servers
Hi friends,Thank you to all for your response and cooperation to me. I have a doubt.I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India).1) Is it possbile to connect these two * servers?2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)?Looking forward to your response. Thank you.With ward regards,Chandra. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --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] Which BRI Card ?
I've been asked to comment on how the Digium B410P stacks up, but have not had any experience of this card. Can anyone comment on the strengths / weaknesses or compare it to any other BRI isdn card that is asterisk compatible ? On another note, would it be better to buy a TDM04B instead of 4 ata devices like a sipura 3000 ? Many thanks for any advice :) Julian ___ --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] IP interface box for Meridian type digital phone
Hi, Anyone can point me to a product that would allow to connect Meridian type digital phone to an Asterisk PBX. I am looking for something like an ATA that you would connect the digital phone to and the ATA would attached to the IP network going to the Asterisk server. Thanks for any hints. Andre Courchesne ___ --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] Extension Ring on Multiple Phones
This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120) ; works fine new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) I edit extensions.conf to the new line above, type 'reload' into the CLI, see the new line with 'show dialplan' and actually see the new line above, but when I dial the DID 879-6777 it rings on extension 120 only. Have I missed a step? Larry Jonathan k. Creasy wrote: EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow ___ --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] ISDN BRI, and Trixbox
On Tue, 2006-08-29 at 00:59 +0100, Nigel Godfrey wrote: Hi, I have 2 Billion BRI ISDN cards, and intend to set up a Trixbox server. I currently have a Asterisk 1.0 server which has been running for a couple of years using Billion BRI cards and Junghanns BRIstuff. Trixbox, if it works, seems to offer various useful UI elements, and one easy standardised install. Or would I be better off not using Trixbox? Astbill perhaps? Does anyone have experience of setting up BRI in a Trixbox environment? I'll report on a good method for the record when I find one. I don't have experience with Trixbox but know that several people have tried to make mISDN work and failed. The alternative for now is to use vISDN which (sort of) works. Search the Trixbox forum as it has a well documented report of these attempts (author Rehan iirc). Regards, Patrick ___ --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] IP interface box for Meridian type digital phone
The Citel SIP Handset Gateway http://www.citel.com allows you to connect Meridian phones (and others) to SIP services, such as Asterisk (and others). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andre Courchesne - Consultant Sent: 29 August 2006 16:29 To: asterisk-users@lists.digium.com Subject: [asterisk-users] IP interface box for Meridian type digital phone Hi, Anyone can point me to a product that would allow to connect Meridian type digital phone to an Asterisk PBX. I am looking for something like an ATA that you would connect the digital phone to and the ATA would attached to the IP network going to the Asterisk server. Thanks for any hints. Andre Courchesne ___ --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] Advice needed - asterisk Mitel 200SX
Greetings: I have a Mitel 200SX PBX right now that is experiencing some voicemail difficulties. The voicemail system is a SmoothOperator system - yep, thats right, genuine 1995 Dos-based ISA motherboard technology running our corp voicemail system. In a word, boring and quickly becoming unstable. Does anyone have any experience with implementing Asterisk as the voicemail provider for a Mitel system like the 200SX? The voicemail lines are dual discreet lines (2 lines, 1 cord, analog), the boards in the SmoothOperator are Dialogic boards (ISA). 6 lines are the lines feeding the voicemail and 2 lines are signaling. Oh yeah, the SmoothOperator system is also doing all ACD functions as well. I need to replace the SmoothOperator system - now. Hints, tips, ideas would be helpful. No we are NOT replacing the Mitel. No we are not shifting the PBX functionality to Asterisk. This is voice mail and ACD only. It would be nice if I could recycle the Dialogic cards but that's not a major requirement - I can always go to a pair of Digium TDM cards if needed. I appreciate any and all advice! Thanks! Ron Gage Westland, MI ___ --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] IP interface box for Meridian type digital phone
Take a look at the citel handset gateway. The SIP one. It's an ATA for 24 nortel/meridian phones http://www.citel.com/products/handset_gateways/ david -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Andre Courchesne - Consultant Envoyé : 29 août 2006 11:29 À : asterisk-users@lists.digium.com Objet : [asterisk-users] IP interface box for Meridian type digital phone Hi, Anyone can point me to a product that would allow to connect Meridian type digital phone to an Asterisk PBX. I am looking for something like an ATA that you would connect the digital phone to and the ATA would attached to the IP network going to the Asterisk server. Thanks for any hints. Andre Courchesne ___ --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] Connecting two asterisk servers
Please search the wiki first. Most of your questions you post can easily be found by doing a search. Put some effort into finding the answers to your questions first and on your own, and then if you still have questions, I'm sure everyone would be more than willing to help. On 8/29/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi friends,Thank you to all for your response and cooperation to me. I have a doubt.I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India). 1) Is it possbile to connect these two * servers?2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)?Looking forward to your response. Thank you. With ward regards,Chandra. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- 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] DTMF between cisco and sipura going through asterisk
Hello all, we're having an issue with DTMFs being sent to Sipura's. Calls are originating from a Cisco AS5300 being sent to asterisk which in turn sends it to the Sipura. Connected to the Sipura is a legacy PBX (or actually shows the same problem with a cheap answering machine). The DTMFs sent from the AS5300 aren't recognised by the legacy PBX. - DTMFs are recognised correctly on the asterisk (when we check voicemail) - The cisco is setup with dtmf-relay rtp-nte - in sip.conf the cisco and sipura are set to rfc2833 If I set the cisco in dtmf-relay rtp-cisco it works on the sipura, but not on the asterisk. Unfortunately I can only set one dtmf-relay mode on the cisco. Is there anything I can change on asterisk or sipura to get the sipura to work with the rtp-nte (or to get asterisk to work with the cisco-rtp)? Any hints can help, Thanks Ben ___ --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
[Fwd: Re: [asterisk-users] Asterisk t38passthrough]
You can download the patch for t.38 passthrough from the URL: http://bugs.digium.com/file_download.php?file_id=9335type=bug Regards, Ricardo. Patrick wrote: On Tue, 2006-08-29 at 12:50 +0100, Ricardo Carvalho wrote: Finally it's working! I was doing everything well, the problem was that neither the latest branch of Asterisk-t38 worked (http://svn.digium.com/svn/asterisk/team/group/t38passthrough/), neither the patched version of Asterisk 1.2.7. Only the branch of Asterisk-t38 made from source version 1.2.4 works for me. If anyone deployed with success those versions that I didn't make to work, please tell me! Ricardo, If possible can you please email me the t38passthrough patch that works for you. Thanks and regards, Patrick ___ --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] Standard for transfer via IAX provider?
On 25 Aug 2006, at 17:42, Henry J. Cobb wrote: Is there any standard way to signal to an IAX provider that I want them to conference in another Asterisk box located elsewhere and then hand off the call to the remote center after a short period of three-way talk? My problem is that I don't want to take a double hit for latency back and forth from the United States. Probably. If the last of your steps is a transfer, and there is no other reason for your local asterisk to stay in the media stream, then by default asterisk will try and step out of the stream. I had an amusing instance of this where both legs of the call were in fact outbound calls to the PSTN via the same provider. My asterisk stepped out of the call and left the provider's asterisk bridging the call. It drove their billing software nuts! Tim Panton www.mexuar.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
[asterisk-users] Copying a recording to a voice mail box
Hello,I am new to asterisk and have a very newbie question. I am try to implement a simple IVR solution that prompts users to say an item, record that then prompt for another item record that.. etc... Here is what I have so far. [custom-lbp]exten = s,1,Playback(LBPsayname)exten = s,2,Record(mymessage:gsm)exten = s,3,Playback(LBPcityzip)exten = s,4,Record(mymessage:gsm,a)exten = s,5,Playback(LBPsayphone) exten = s,6,Record(mymessage:gsm,a)exten = s,7,Playback(LBPgoodbye)exten = s,8,HangupWhich seems to work well except for two problems:1) How do I move mymessage.gsm to a users voice mailbox so if a message is recorded by this IVR users can dial-in or use ARI to retrieve the messages. 2) How do I handle concurrent callers so if two or more people are calling the IVR at once the mymessage doesn't get overwritten or incorrect information appended to it.Thanks for any help-Nate ___ --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] Copying a recording to a voice mail box
Hello,I am new to asterisk and have a very newbie question. I am try to implement a simple IVR solution that prompts users to say an item, record that then prompt for another item record that.. etc... Here is what I have so far. [custom-lbp]exten = s,1,Playback(LBPsayname)exten = s,2,Record(mymessage:gsm)exten = s,3,Playback(LBPcityzip)exten = s,4,Record(mymessage:gsm,a)exten = s,5,Playback(LBPsayphone) exten = s,6,Record(mymessage:gsm,a)exten = s,7,Playback(LBPgoodbye)exten = s,8,HangupWhich seems to work well except for two problems:1) How do I move mymessage.gsm to a users voice mailbox so if a message is recorded by this IVR users can dial-in or use ARI to retrieve the messages. 2) How do I handle concurrent callers so if two or more people are calling the IVR at once the mymessage doesn't get overwritten or incorrect information appended to it.Thanks for any help -Nate ___ --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] Re: [asterisk-biz] Asterisk Tools
Interesting, I was asking about the FCC dictating new (read stupid) methods of doing this which will cause higher communications systems. The end user will get a bigger bill in the future because folks are faking their CallerID/ANI today. A private investigator making a few calls or a call center shifting numbers to bypass blocks or no call lists. Personally I don't use callerID, I just answer the phone, few have the number but I am waiting for a dialer to trip on to it. How will the regulators mandate to the service providers requirements for proper call presentation On 8/29/06, Matthew Rubenstein [EMAIL PROTECTED] wrote: How can we preserve (create?) caller authentication while allowing the equivalent of email's Reply-To redirection, within current call metadata protocols? Lots of people have a single incoming phone# which also rings their mobile phone, and even emails them where they can return calls from a softphone - all of which should share the same caller ID, and reply address. Right now spoofing is the only way, but that should be distinguished from inauthentic spoofing. On Tue, 2006-08-29 at 13:13 -0400, Andrew Latham wrote: 1. Buy a T1 2. Setup 3.. I am afraid of why you want to do this, I am also afraid of what the FCC will do to curb this in the future by altering switching on copper and fiber connections. As a BIZ list discussion, what can the FCC do to curb Caller ID and other spoofing, many of us in the business know how insecure and unreliable the system currently is. Will circuit ID lookups attach an ANI in the future and how long could this tie up new and upgrade installations... On 8/29/06, perl ninja [EMAIL PROTECTED] wrote: Hello, i was in need of a script for ANI Spoofing as ive read that CallerID spoofing is relativly easy but now adays places such as purolator and so on Check with ANI rather then with the CallerID to find the callers location, thus i was wanting a script that would spoof both, if possible.. Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz -- (C) Matthew Rubenstein -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] GXP-2000 auf Betafirmware updaten?
Hi, currently I use version 1.1.0.16 for my GXP-2000 which works really fantastic. The only drawback I see is the addressbook. Is the firmware 1.1.1.9 stable enough to use the phone in normal environment? The webpage http://www.voip-info.org/wiki/view/GXP-2000 says that there it is possible to download the addressbook as a XML-file. The problem is if the version not works it is not possible to downgrade to 1.1.0 Thx for any feedback, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --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] Mix Monitor call quality
Hi trying to record calls using mixmonitor, but I'm having problems with call quality the call seems OK but then it drops frames with silence ( for less than 0.5 seconds) then call continues All I'm doing is bridging two zap channels and recording no transcoding or changes to the channels Asterisk version 1.2.10 also under certain conditions Asterisk just stops any advice would be appreciated Thanks Robb ___ --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] SIP T1 timer and qualify=yes
I ran in the same issues as John Todd did some while ago: http://lists.digium.com/pipermail/asterisk-users/2005-November/129541.html I use qualify=yes to ping our internal SIP proxies for failover and therefore I have very low delays, e.g. Name/usernameHostDyn Nat ACL Port Status mid2-3 xx.xx.xx.xx 5060 OK (1 ms) which causes Asterisk to use a very small T1 to retransmit SIP requests: TimeProtocol Info 4.107899SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.113318SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.113339SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.121283SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.129283SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.145284SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] 4.177281SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] (this looks like a SIP DOS attack to me) Setting T1 according the SIP qualify delay only makes sense if the delay measurements are done with the final target of a SIP request. If I ping a SIP proxy instead, the ping delay does not say anything about the actual end-to-end SIP signaling path delay. My recommendation would be to statically set T1 to 500ms according to RFC 3261. If that is not an option I'd set a minimum T1 that is at least 100ms. -Christian smime.p7s Description: S/MIME Cryptographic Signature ___ --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] compile problems with app_rxfax.c and asterisk 1.2.11
Hi All, Trying to add faxing to asterisk but get a compile error. Any ideas? Is it broken for Asterisk 1.2.11 or was it me again :-) I followed the instructions from here: http://www.asteriskguru.com/tutorials/spandsp.html Thanks in advance Phil gcc -shared -Xlinker -x -o app_page.so app_page.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c: In function `phase_e_handler': app_rxfax.c:105: error: structure has no member named `column_resolution' app_rxfax.c:105: error: structure has no member named `row_resolution' app_rxfax.c:116: error: structure has no member named `row_resolution' app_rxfax.c:122: error: structure has no member named `row_resolution' app_rxfax.c: In function `phase_d_handler': app_rxfax.c:147: error: structure has no member named `columns' app_rxfax.c:147: error: structure has no member named `rows' app_rxfax.c:148: error: structure has no member named `column_resolution' app_rxfax.c:148: error: structure has no member named `row_resolution' app_rxfax.c: In function `rxfax_exec': app_rxfax.c:281: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:281: error: too many arguments to function `fax_init' app_rxfax.c:304: warning: passing arg 1 of `fax_rx' from incompatible pointer type app_rxfax.c:307: warning: passing arg 1 of `fax_tx' from incompatible pointer type app_rxfax.c:344: warning: passing arg 1 of `fax_release' from incompatible pointer type app_rxfax.c: At top level: app_rxfax.c:81: warning: 't30_flush' defined but not used make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.11/apps' make: *** [subdirs] Error 1 ___ --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 Development and Release Cycle
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tomislav Parčina wrote: Do Asterisk team care about this anymore? I don't know. Do you use Asterisk? That makes you part of the team. Have you tested the trunk version? Provided assistance testing out patches waiting for completion? Really, once all the new features have been completed, it will be released. If you would prefer it to be released now (I.E. before everything has been tested and possibly fixed), just download SVN trunk. - -- 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) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE9IheS6d5vy0jeVcRAmdEAJ4yrtoa4wcjv442g2QG/TTqa+GYaQCePhUx 5YJIJc1bwCBqGsGVfLEDSbY= =xSQZ -END PGP SIGNATURE- ___ --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] Analyze core file prodeced after safe_asterisk crashh
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 equis software wrote: Hi, I have a Call Center running with safe_asterisk script. When Asaterisk crash produce a core file but I don´t know how analyze it! Any ideas?? http://www.asterisk.org/doxygen/AstDebug.html - -- 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) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE9Ij4S6d5vy0jeVcRAnJ5AJ9WxQ1bOAKRSZ/vmna5Fz2tnIeogQCePd5D oMK6Re/6xukIzoGHaYhSUws= =vuCp -END PGP SIGNATURE- ___ --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] Detect if cell phone or users
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dovid Bender wrote: Hi List, I have sprint pcs cellular service and I discoverd that I am able to send a text message to a landline. If I do I get an SMS from the saying I sent a text message to a landline. Then the landline that I sent a text message to gets a call with my message (text to speach). I was wondering if there was any way for me to detect if a number is a mobile phone or a landline. Is this something that only cellular providers can do or can I have asterisk do it (I asume I would need to create a patch). Thanks a lot. Don't cellphone numbers start with a different code where you live? Or were you wanting to do something worldwide? - -- 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) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE9Ik1S6d5vy0jeVcRAmXcAJ9ytuu7tU2B1IpHjTW6onFolnt3VQCeINeJ FCRGKTnmi/k0oZmea6lXo4Q= =GGrd -END PGP SIGNATURE- ___ --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] Voicemail/Email Integration
[EMAIL PROTECTED] wrote: Is there a way to implement voicemail/email integration such that you could retrieve the voicemail with either the phone or email, but only have to delete the message once? You can try our voicemail client called Tycho, available for MacOS X, Linux and Windooze. You need an (apache) webserver with php 4.3 or better on the same box the voicemail is stored on. Before you can use the client you need to install the vmxml server scripts. The Stuff is beta but works pretty well. Were right now adding imap as transport layer so you wont need the server side php scripts in the future. The Stuff is available at: http://sip-syndication.com regards, Arnd ___ --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] Extension Ring on Multiple Phones
Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: This is a reply to a fairly old thread.My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120); works finenew: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124)I edit extensions.conf to the new line above, type 'reload' into the CLI, see the new line with 'show dialplan' and actually see the new lineabove, but when I dial the DID 879-6777 it rings on extension 120 only.Have I missed a step?LarryJonathan k. Creasy wrote: EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all.I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow___--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] Detect if cell phone or users
How do you handle situations where a cellphone number has been ported to a land line/VoIP provide or vice versa? The phone number isn't a reliable indicator of provider or medium.-brandon On 8/29/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE-Hash: SHA1Dovid Bender wrote: Hi List, I have sprint pcs cellular service and I discoverd that I am able to send a text message to a landline. If I do I get an SMS from the saying I sent a text message to a landline. Then the landline that I sent a text message to gets a call with my message (text to speach). I was wondering if there was any way for me to detect if a number is a mobile phone or a landline. Is this something that only cellular providers can do or can I have asterisk do it (I asume I would need to create a patch). Thanks a lot. Don't cellphone numbers start with a different code where you live?Orwere you wanting to do something worldwide?- --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)-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.2 (MingW32)Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.orgiD8DBQFE9Ik1S6d5vy0jeVcRAmXcAJ9ytuu7tU2B1IpHjTW6onFolnt3VQCeINeJFCRGKTnmi/k0oZmea6lXo4Q==GGrd-END PGP SIGNATURE-___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED] AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --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] Detect if cell phone or users
I don't think there's any authority in North America that tells you whether a number is a cellular number. However, it's conceivable you could write a script to access information available on, for example, www.telcodata.us, and check the prefix-type for a given phone number. The prefix type for all known (to me) wireless numbers is, in fact, WIRELESS -- you can't pay too much attention to the company given on that site, as WLNP makes this information quite superfluous. Keep in mind that LNP in general will make this information much less reliable than it used to be before LNP -- it's now possible to take your landline to a cell-phone and vice versa, so bets are off when it comes to that. Dovid Bender wrote: Hi List, I have sprint pcs cellular service and I discoverd that I am able to send a text message to a landline. If I do I get an SMS from the saying I sent a text message to a landline. Then the landline that I sent a text message to gets a call with my message (text to speach). I was wondering if there was any way for me to detect if a number is a mobile phone or a landline. Is this something that only cellular providers can do or can I have asterisk do it (I asume I would need to create a patch). Thanks a lot. Dovid ___ --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] OT: Bandwidth calculations and PCI/PCIX/PCIE
I found this interesting but old white paper at Dell.com tech solutions and another one from INTEL. It compares bandwidth usage of a PCI, PCI-X, PCI-E in33/66/100/133mhz bus and different technologies that can saturate the bus. It helped me understand the bandwidth required for TDM (sangoma/digium) cards and how far can I push the PCI bus in an old and newmotherboard. I hope it help others to understand how much a network card can pump and make calculations about consumptions in TDM cards. make sure the link is a one-line in your browser Original online document http://www.dell.com/content/topics/global.aspx/vectors/en/2004_pciexpress?c=uscs=08Wl=ens=bsdv here is the link to the same Dell article but in PDF form. http://www.dell.com/downloads/global/vectors/2004_pciexpress.pdf Another interesting document from INTEL www.intel.com/technology/pciexpress/devnet/docs/WhatisPCIExpress.pdf The facts learned from these documents are: a- 3.3volts/32bit PCI cards can be used in PCI-X slots. (i just discovered that, sorry forliving under a rock) b- The slowest PCI card in Mhz will dictate that PCI-X bus speed. So avoid degradation by not installing a PCI card and a PCI-X card in the same bus (check you motherboard design), your motherboard design usually have two buses. c- If you use a PCI-X based implementation motherboard, you will not saturate the bandwidth of the board, using Quad or Octal port cards (e1/t1/j1). -- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Advice needed - asterisk Mitel 200SX
Fix the computer It worked for 10 years, it can be made to continue working The worst thing you can do is try and implement such a radical change without lots of testing, and more testing. Unless, of course, you are the owner of the company or want to go on a job search! JMO John Novack Ron Gage wrote: Greetings: I have a Mitel 200SX PBX right now that is experiencing some voicemail difficulties. The voicemail system is a SmoothOperator system - yep, thats right, genuine 1995 Dos-based ISA motherboard technology running our corp voicemail system. In a word, boring and quickly becoming unstable. Does anyone have any experience with implementing Asterisk as the voicemail provider for a Mitel system like the 200SX? The voicemail lines are dual discreet lines (2 lines, 1 cord, analog), the boards in the SmoothOperator are Dialogic boards (ISA). 6 lines are the lines feeding the voicemail and 2 lines are signaling. Oh yeah, the SmoothOperator system is also doing all ACD functions as well. I need to replace the SmoothOperator system - now. Hints, tips, ideas would be helpful. No we are NOT replacing the Mitel. No we are not shifting the PBX functionality to Asterisk. This is voice mail and ACD only. It would be nice if I could recycle the Dialogic cards but that's not a major requirement - I can always go to a pair of Digium TDM cards if needed. I appreciate any and all advice! Thanks! Ron Gage Westland, MI ___ --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 t38passthrough
Hi Ricardo, On a 1.2.4 with the T.38 patch, I tried both t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes and t38pt_udptl = yes t38pt_rtp = no t38pt_tcp = no but still got ...chan_sip.c:3716 process_sdp: Unknown SDP media type in offer: image 5144 UDPTL t38 Warnings I tried it on Kapanga Softphone as suggested, and I'll tried it on Grandstream ATA's later. Are there anything I'm missing? Thank you. Andy On 8/24/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstream ATA to connect Asterisk to a Fax machine. Every time I send a fax, it gets sent using codec G711, and never T.38. I added the following parameters in the [general] section as well as in device configurations: t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes I think that's the only thing that is needed to do to enable T.38 pass through... Why does Asterisk keeps sending in G711? Any help? Regards, Ricardo. ___ --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] Connecting two asterisk servers
In short, yes... The wiki (http://www.voip-info.org) has documentation on how to configure your servers, how to configure the dialplan, etcI don't mean to single you out mate, but has anyone else noticed an increase in the number of questions being asked that could have been answered simply by visiting the wiki, reading the sample docs in the package, or even doing a Google search? I seem to recall the general rule of this list is that you should have already at least tried to find the answer. Here's a few links to get you started: The Asterisk Wiki, Asterisk Guru, Getting Started, GNU Inter, AGI Guide, O'reilly Onlamp Article - by John Todd, One Unified. It took me more time to cut and past those links than it did to find them, they were on the Asterisk.org support page. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy BoySent: Tuesday, August 29, 2006 11:16 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Connecting two asterisk servers Hi friends,Thank you to all for your response and cooperation to me. I have a doubt.I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India).1) Is it possbile to connect these two * servers?2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)?Looking forward to your response. Thank you.With ward regards,Chandra. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --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] Extension Ring on Multiple Phones
That's very very odd...that should work fine :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120) ; works fine new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) I edit extensions.conf to the new line above, type 'reload' into the CLI, see the new line with 'show dialplan' and actually see the new line above, but when I dial the DID 879-6777 it rings on extension 120 only. Have I missed a step? Larry Jonathan k. Creasy wrote: EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow ___ --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] Analyze core file prodeced after safe_asterisk crashh
It's not clear if the OP wanted 1) information on how to analyse the core file or 2) provide information to the bug tracker for others to analyse. Matt's answer addresses #2. How about #1? Anybody care to share their techniques for analysing a core dump? On Tue, 29 Aug 2006, Matt Riddell (IT) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 equis software wrote: Hi, I have a Call Center running with safe_asterisk script. When Asaterisk crash produce a core file but I don´t know how analyze it! Any ideas?? http://www.asterisk.org/doxygen/AstDebug.html - -- 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) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE9Ij4S6d5vy0jeVcRAnJ5AJ9WxQ1bOAKRSZ/vmna5Fz2tnIeogQCePd5D oMK6Re/6xukIzoGHaYhSUws= =vuCp -END PGP SIGNATURE- ___ --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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ --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] Connecting two asterisk servers
Crazy Boy a écrit : Hi friends, Thank you to all for your response and cooperation to me. I have a doubt. I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India). 1) Is it possbile to connect these two * servers? Yes. Just have something like: [serverA] type=peer host=serverA.IP.Address In ServerB's sip.conf and [serverB] type=peer host=serverB.IP.Address In ServerA's sip.conf 2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)? Of course. Say user joe is registered with serverB, then within serverA's dialplan, you can use: exten = 123456,1,Dial(SIP/[EMAIL PROTECTED]) ; [EMAIL PROTECTED] has extension '123456' Within serverB's dialplan, you'd simply use: exten = 123456,1,Dial(SIP/joe) ; [EMAIL PROTECTED] has extension '123456' Cheers, Jean-Michel. ___ --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 1.2.11 and ${SIPDOMAIN} variable
Hi,
I've just upgraded from Asterisk 1.2.10 to 1.2.11 and I've noticed that the
${SIPDOMAIN} variable now contains a different (and to my mind, incorrect)
value than what it used to. Instead of (say) example.com, it now contains
the string example.com;user=phone instead which causes calls to fail if you
then try and use the Dial app to call [EMAIL PROTECTED] or try to do a
match on a particular domain. I just wanted to find out if anyone else has
noticed this so I can get some evidence to report this as a bug...
Thanks
Gary H
--
Gary Hawkins MBCS [EMAIL PROTECTED]
PGP: 0x6D4E5C77 (expires 31 Dec 2006)
Web: http://www.garyhawkins.me.uk
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Re: [asterisk-users] Detect if cell phone or users
On 13:56, Tue 29 Aug 06, Jay Milk wrote: I don't think there's any authority in North America that tells you whether a number is a cellular number. However, it's conceivable you could write a script to access information available on, for example, www.telcodata.us, and check the prefix-type for a given phone number. The prefix type for all known (to me) wireless numbers is, in fact, WIRELESS -- you can't pay too much attention to the company given on that site, as WLNP makes this information quite superfluous. Keep in mind that LNP in general will make this information much less reliable than it used to be before LNP -- it's now possible to take your landline to a cell-phone and vice versa, so bets are off when it comes to that. hhmm, I wonder how long it takes here for that to happen. You cant turn a cell number into a landline nor viceversa. Heck, most providers dont even like it when you install a gsm gateway to call your coworkers for free with the gateway. So here it's pretty simple to know wether a call is to landline or cell. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CPU configuration for 250 calls SIP to SIP to IAX and fonebridge and two asterisk servers
Hi, I would like to read your comments for the following setup: Building A: 3 voice E1incoming toa quad redfone fonebridge (TDMoE) The fonebridge goes to a port in a 24 port gigabit switch in the gigabit switch VLAN1 is for the fonebridge and the first gigabit NIC on a dual NIC server in the gigabit switch VLAN2 is for the second gigabit NIC card on the server andeleven 10/100 switches with 250 SIP phone users running g711 codec (24 phones per 10/100 switch,each switch is 24port) Building A and Building B are connected over a 10Mbits fiber link. Numeric Extensions at building A are 1xxx Building B: same config E1/switch/users as building A Building A and Building B are connected over a 10Mbits fiber link. Numeric Extensions at building B are 2xxx The asterisk servers at each side will talk IAX2 between each other for building-to-building call transfers. Suggested machine: Im considering a Dell PowerEdge 9G 1950, Dual Xeon 3.20Ghz, 1066 FSB, 4GB ram. two 73GB SAS 15k RPMs hard disk and dual gbit network card. Asterisk Features: Music on hold call transfer call waiting (but only on executive phones, around 20) voicemail a small queue (about 10 persons) and a simple IVR (play prompts for department selection, transfer according to selection). No call recording requested at this time. Operating System: Centos 4.3 Codecs: G711 for the SIP to asterisk and IAX for server to server transfers. If IAX is not recommended, please advice. Notes: a- Is is expected to have the 250 SIP users talking either to each other and/or to the other building and/or to the fonebridge E1s. b- I know that for SIP-to-ZAP a calculation of 30Mhz per voice channel is a rule of thumb, but i also read somewhere that the same calculation does not apply when doing Pure IP, no SIP/ZAP and pure g711 implementations I'm in that category. c-Just for the record, what if I change to g729? d- It is expected to have 80% of the calls over the E1 being incoming from the PSTN and the other 20% ar the SIP users calls to the PSTN Is is also expected to haveone 24 port Rhino FXS channel banks connected to the 4th port of the fonebridge. Is used, it will add another 24 users to the setup. Thanks in advance. Your comments are welcomed. Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de PanamaCel Panama. +(507) 6694-4780 ___ --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] Analyze core file prodeced after safe_asterisk crashh
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Edwards wrote: It's not clear if the OP wanted 1) information on how to analyse the core file or 2) provide information to the bug tracker for others to analyse. Matt's answer addresses #2. How about #1? Anybody care to share their techniques for analysing a core dump? Doing the bt full as described in the document I posted is how you analyse the core file. - -- 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) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE9J+OS6d5vy0jeVcRAlSYAJ4rt5j9UPkiMqsjumHAdgWCrZhcOgCfWy1Q tlXc8iRplvZp3IE/TvWroZ8= =p+IX -END PGP SIGNATURE- ___ --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 codec strangeness
Jean-Michel Hiver a écrit :
Hi All,
I have two peers (call then peerA and peerB) on my server, both can
accept g711, g729 and g723. However, when peerA initiates a request,
asterisk decides to transcode g729 into ulaw when peerB could very
well use g729...
This behavior isn't very scalable (transcoding is CPU expensive) and
also it's better to minimize the amount of transcoding wherever
possible. Is there a way I can fix this?
NB: if i set disallow = all and allow=g729 on peerB it all works fine,
but then if peerA decides to send ulaw I'm transcoding again...
Okay, I have digged the archives a bit, and apparently I'm not the only
one having this problem. I am thinking of maybe sorting out this problem
by having:
[peerA-g711]
type=peer
host=123.123.123.123
disallow=all
allow=ulaw
allow=alaw
[peerA-g729]
type=peer
host=123.123.123.123
disallow=all
allow=g729
[peerA-g723]
type=peer
host=123.123.123.123
disallow=all
allow=g723
And then using ${SIP_CODEC} to route the call correctly maybe?
I don't think having multiple peers with the same IP address would be a
big deal for outgoing calls, but asterisk will probably we confused for
incoming calls from 123.123.123.123... what do you think?
Cheers,
Jean-Michel.
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Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?
On Tue, Aug 29, 2006 at 02:49:54PM +0200, Giorgio Incantalupo wrote: Hi Kai, the problem is to find the right kernelI used apt-get *install* kernel-headers-*`**uname* -r*`* -y but it seems not to be the right one...Even zaptel is not working: FATAL: Error inserting zaptel (/lib/modules/2.6.15-1-486/misc/zaptel.ko): Invalid module format find /lib/modules/`uname -r` -name zaptel.ko Is there more than one? Also, compare: modinfo zaptel and: modinfo rtc # or any other module from the main kernel package Specifically, the vermagic line. I've always used Debian Sarge, now I'm oblidged to use Etch testing because Sarge installer doesn't work with new hardware and I do not want to change distro. Off-Topic: apt-get install zaptel zaptel-source m-i a-i zaptel -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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
[asterisk-users] Detecting sound before answer
Title: Message Hello, Lets say Im dialing out and before channels are bridged I hear beep or something similar. That way I know Im calling to other Telco/Provider. Is it possible to detect that beep before channel is answered and to redial through other trunk? Regards/Pagarbiai, Mindaugas Kezys ___ --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] Administrator Forum Email
Hi List; Can someone advise me what is the email of the administrator forum so I can send for him to fix my account? The forum that I am talking about it existed in the following link: http://forums.digium.com/ Regards Bilal Ghayad __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Does anyone use T.38?
We have successfully used Sipura 2100 ATAs for this with an external fax machine connected to its FXS port. The Sipura is connected to a Cisco fax gateway right now, we haven't been able to test it with Asterisk yet. On Fri August 25 2006 06:58, Ricardo Carvalho wrote: Does anyone use T.38 for fax? If you use it, what hardware / software do you use? Thanks, Ricardo. ___ --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 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] Does anyone use T.38?
XMedius is a great T.38 fax product, integrate with LDAP/AD/Exchange. Integrates with the PRI card in our Cisco Routers using H.323. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --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 Development and Release Cycle
- Tomislav Parčina [EMAIL PROTECTED] wrote: Do Asterisk team care about this anymore? Whole text can be found here: http://www.asterisk.org/developers/releasecycle Of course we care. Turns out that schedule was unrealistic, and when we start the next cycle we will regroup and decide if we either stretch out the cycle or reduce the amount of new functionality that gets added during the cycle. -- Kevin P. Fleming Senior Software Engineer Digium, 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
Re: [asterisk-users] Asterisk codec strangeness
are your codec allow= statements in the same order in each peer block?
meaning does peerA have g729 at a different priority than peerB?
Moj
Jean-Michel Hiver wrote:
Jean-Michel Hiver a écrit :
Hi All,
I have two peers (call then peerA and peerB) on my server, both can
accept g711, g729 and g723. However, when peerA initiates a request,
asterisk decides to transcode g729 into ulaw when peerB could very
well use g729...
This behavior isn't very scalable (transcoding is CPU expensive) and
also it's better to minimize the amount of transcoding wherever
possible. Is there a way I can fix this?
NB: if i set disallow = all and allow=g729 on peerB it all works fine,
but then if peerA decides to send ulaw I'm transcoding again...
Okay, I have digged the archives a bit, and apparently I'm not the only
one having this problem. I am thinking of maybe sorting out this problem
by having:
[peerA-g711]
type=peer
host=123.123.123.123
disallow=all
allow=ulaw
allow=alaw
[peerA-g729]
type=peer
host=123.123.123.123
disallow=all
allow=g729
[peerA-g723]
type=peer
host=123.123.123.123
disallow=all
allow=g723
And then using ${SIP_CODEC} to route the call correctly maybe?
I don't think having multiple peers with the same IP address would be a
big deal for outgoing calls, but asterisk will probably we confused for
incoming calls from 123.123.123.123... what do you think?
Cheers,
Jean-Michel.
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!DSPAM:500,44f4a435224481596210392!
--
Mojo [EMAIL PROTECTED]
Office Manager, Horan Company, LLC
(907) 747- x112
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Re: [asterisk-users] Copying a recording to a voice mail box
There are flags to the VoiceMail application that instruct it to behave differently than normal. It probably won't let you append the messages into one message, however. That seems like it would be a problem in this project, but if it's not, you might try: VoiceMail([EMAIL PROTECTED]) Substitute the correct mailbox number for the 100. The s means don't play unavailable or busy messages, and don't read instructions. It just beeps and starts recording, and a # will terminate it (as will a timeout configured in voicemail.conf). As discussed recently on the list, asterisk does its own mailbox locking, so concurrent callers wouldn't be a problem. But in this design, if two callers WERE to use the system concurrently, you might end up with messages in the following order: name1 name2 zip1 zip2 phone1 phone2 They would be even further mixed up if the two callers took different lengths of time to answer the questions. I might mitigate that confusion by determining the maximum number of concurrent callers and each one would use a different voicemailbox ? I don't think this will help you achieve your goal but it should give you some more building blocks to play with :) Moj Nate Criss wrote: Hello, I am new to asterisk and have a very newbie question. I am try to implement a simple IVR solution that prompts users to say an item, record that then prompt for another item record that.. etc... Here is what I have so far. [custom-lbp] exten = s,1,Playback(LBPsayname) exten = s,2,Record(mymessage:gsm) exten = s,3,Playback(LBPcityzip) exten = s,4,Record(mymessage:gsm,a) exten = s,5,Playback(LBPsayphone) exten = s,6,Record(mymessage:gsm,a) exten = s,7,Playback(LBPgoodbye) exten = s,8,Hangup Which seems to work well except for two problems: 1) How do I move mymessage.gsm to a users voice mailbox so if a message is recorded by this IVR users can dial-in or use ARI to retrieve the messages. 2) How do I handle concurrent callers so if two or more people are calling the IVR at once the mymessage doesn't get overwritten or incorrect information appended to it. Thanks for any help -Nate !DSPAM:500,44f47aec315861174510073! ___ --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 !DSPAM:500,44f47aec315861174510073! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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 codec strangeness
Mojo with Horan Company, LLC a écrit : are your codec allow= statements in the same order in each peer block? meaning does peerA have g729 at a different priority than peerB? Aah, thanks that fixed it because most of the traffic is g729. Now, if peerA does send me ulaw instead of g729 (because it choose to, say), and the order of peerB is g729, ulaw, alaw, am I still going to have the same issue? My guess is yes... Cheers, Jean-Michel. ___ --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] Digium makes the list!
There is a link on Groklaw for the following article: Open source companies to watch Digium makes the second entry on the list. Link below: http://www.networkworld.com/news/2006/082806-open-source.html?ts Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Extension Ring on Multiple Phones
Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120); works fine new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) I edit extensions.conf to the new line above, type 'reload' into the CLI, see the new line with 'show dialplan' and actually see the new line above, but when I dial the DID 879-6777 it rings on extension 120 only. Have I missed a step? Larry Jonathan k. Creasy wrote: EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow ___ --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 -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --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] New Parrot application, repeats what you say and more!
Lobster Technologies has just anounced the release of the most annoying open source IVR application ever devised by lobsters called PhoneParrot. PhoneParrot is an app that uses silence detection to repeat everything a person says in to the phone. http://www.lobstertech.com/code/phoneparrot/ For example, you could have PhoneParrot call your mother in the middle of the night. She will pick up the phone and say moshi moshi, the phone parrot will then say moshi moshi. Your mother, confused will then probably say, who is you playa? to which the phone parrot will respond, who is you playa?. I think you get the idea. But phone parrot can do more than just repeat what a person says! Here are some of the cool features: * Apply voice change effect to repeated voice (Only if libsoundtouch4c is installed and phone parrot is compiled from source) * Play random sound clip if caller rambles on for too long * Greet with random sound clip in response to first thing caller says * ToDo: Repeat things previously said in conversation Years of research has indicated that with proper deployment, most humans are unable to tell that they are talking to a machine if the voicechanger add-on is used. PhoneParrot is intended for the age 16-25 mischievous Linux hacker market; however, businesses deploying Asterisk may also find PhoneParrot useful as a torture extension for telemarketers. Please enjoy the new software, and remember: Keep it Open Source Pigs Justin Alexander Roberts Tunney Sanitation Engineer Lobster Technologies, Corp. Phone: (666) 700-1337 ___ --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 codec strangeness
I suspect so, but I'm not sure :) Jean-Michel Hiver wrote: Mojo with Horan Company, LLC a écrit : are your codec allow= statements in the same order in each peer block? meaning does peerA have g729 at a different priority than peerB? Aah, thanks that fixed it because most of the traffic is g729. Now, if peerA does send me ulaw instead of g729 (because it choose to, say), and the order of peerB is g729, ulaw, alaw, am I still going to have the same issue? My guess is yes... Cheers, Jean-Michel. ___ --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 !DSPAM:500,44f4c28a71772014189408! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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] Extension Ring on Multiple Phones
Then entire OLD extension must be removed so the new one will match -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120); works fine new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) I edit extensions.conf to the new line above, type 'reload' into the CLI, see the new line with 'show dialplan' and actually see the new line above, but when I dial the DID 879-6777 it rings on extension 120 only. Have I missed a step? Larry Jonathan k. Creasy wrote: EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow ___ --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 -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --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] Extension Ring on Multiple Phones
The whole thing. Both (old and new)have the same exten and the same priority, you can't do that and expect it to work properly. The new extenwill call all 3 phones at the same time, whoever answers first gets the call. If you want it to callSIP/120 first and if they don't answer then ring to all 3, you'd want to do this: exten =_879677[67],1,Dial(SIP/120|20) ;this will ring for 20 seconds then go to priority 2.exten =_879677[67],2,Dial(SIP/120SIP/122SIP/124)bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: Color me puzzled.What part of: exten = _879677[67],1,Dial(SIP/120)should be deleted?Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120); works fine new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) I edit extensions.conf to the new line above, type 'reload' into the CLI, see the new line with 'show dialplan' and actually see the new line above, but when I dial the DID 879-6777 it rings on extension 120 only. Have I missed a step? Larry Jonathan k. Creasy wrote: EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all.I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow ___ --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--Larry Alkoff N2LA - Austin TXUsing Thunderbird on Linux___--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] Asterisk Performance without RTP?
On Tue, 29 Aug 2006, Nick Hoffman wrote: On Tue August 29 2006 04:39, Greg Boehnlein [EMAIL PROTECTED] wrote: On Mon, 28 Aug 2006, Andrew Kohlsmith wrote: On Monday 28 August 2006 13:02, Greg Boehnlein wrote: I've pushed over 1,000 concurrent calls this way using the SIPP program for SIP performance testing. There was some tuning that needed to be done, but it worked. Never went that far in production, though. May you share some of your tuning with us? What gotchas did you discover? Just making sure your dial-plan as efficient as possible, that you have enough sockets and open file limits in the kernel, not connecting to the CLI console, never, ever using cdr_mysql or cdr_odbc for your CDR records (locking / contention issues) etc... Lots of basic common sense stuff that you often forget about.. :) Hi Greg. What problems/performance issues does cdr_mysql introduce? If the database is unavailable, or performance is slow, it can cause a blocking condition that will stop the entire system from processing anything. It may have been fixed since then, but I thought that cdr_mysql was deprecated.. -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] Re: [asterisk-biz] Asterisk Tools
On 8/29/06, Andrew Latham [EMAIL PROTECTED] wrote: Interesting, I was asking about the FCC dictating new (read stupid) methods of doing this which will cause higher communications systems. The end user will get a bigger bill in the future because folks are faking their CallerID/ANI today. A private investigator making a few calls or a call center shifting numbers to bypass blocks or no call lists. Personally I don't use callerID, I just answer the phone, few have the number but I am waiting for a dialer to trip on to it. Interesting you mention call centers, since they are prohibited by FCC rules to change CID. How will the regulators mandate to the service providers requirements for proper call presentation On 8/29/06, Matthew Rubenstein [EMAIL PROTECTED] wrote: How can we preserve (create?) caller authentication while allowing the equivalent of email's Reply-To redirection, within current call metadata protocols? Lots of people have a single incoming phone# which also rings their mobile phone, and even emails them where they can return calls from a softphone - all of which should share the same caller ID, and reply address. Right now spoofing is the only way, but that should be distinguished from inauthentic spoofing. On Tue, 2006-08-29 at 13:13 -0400, Andrew Latham wrote: 1. Buy a T1 2. Setup 3.. I am afraid of why you want to do this, I am also afraid of what the FCC will do to curb this in the future by altering switching on copper and fiber connections. As a BIZ list discussion, what can the FCC do to curb Caller ID and other spoofing, many of us in the business know how insecure and unreliable the system currently is. Will circuit ID lookups attach an ANI in the future and how long could this tie up new and upgrade installations... On 8/29/06, perl ninja [EMAIL PROTECTED] wrote: Hello, i was in need of a script for ANI Spoofing as ive read that CallerID spoofing is relativly easy but now adays places such as purolator and so on Check with ANI rather then with the CallerID to find the callers location, thus i was wanting a script that would spoof both, if possible.. Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz -- (C) Matthew Rubenstein -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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
