RE: [Asterisk-Users] Pros and Cons of T1/E1 cards
Hi Steef, Do you want to send me an email to [EMAIL PROTECTED] and I can assist you further. It should work as far as I am aware. Thanks David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of asterisk Sent: 28 November 2005 07:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Pros and Cons of T1/E1 cards Hi, does any of you have experience with these cards on SMP 64bit systems? I'm trying to get one to work but after fixing some some og the code they still do not work. When installing single CPU and 32 bit they work ok. Steef David Waugh wrote: Hi John, I'm going to have to disagree with some previous posts. The Eicon Diva Server PRI/E1/T1 cards support an E1 interface and reduce the load of the call handling, echo cancellation etc as this is all processed on board on the card, and not on the central CPU of the computer. You can use the CAPI interface of the card combined with chan_capi_cm with the card. I have not found any problems when using different kernels or different versions of asterisk. I have one setup in our test lab here at Eicon with Asterisk so it does work! You can have up to 8 Diva Server cards in once machine - including a mixture of the analog and BRI cards. The Diva Server cards in two variants - the V-Series if you only want to use them with Voice based applications and the normal All-in-one cards if you want to do fax and RAS too. If you need any more information let me know, and I will assist further David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Daragon Sent: 25 November 2005 00:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Pros and Cons of T1/E1 cards Hi; We're looking to standardise on a single family of E1 PRI cards. I guess our options are : Digium / Zaptel / libpri Sangoma/ Zaptel / Wanpipe AVM/ CAPI eIcon / CAPI Junghanns / Bristuff Can anyone share any comparative experience of these, please ? Do they differ much in terms of interrupt requirement, CPU load c ? Any info gratefully received. jd ___ --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] Truncated CDR records
Original Message From: Innocent Evil [EMAIL PROTECTED] you can use 'w' option with 'Dial' on 1.2.x I don't think w do anything like 'wait', If I am wrong, correct me someone please According to app_dial.c w- Allow the called party to enable recording of the call by sending\nthe DTMF sequence defined for one-touch recording in features.conf.\n W- Allow the calling party to enable recording of the call by sending\nthe DTMF sequence defined for one-touch recording in features.conf.\n; There is a difference between a w tagged on to the number, and a w as an option. The options come last in the dial command, after a | Leif ___ --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] cdr enhancement with 'rate' column
This is the purpose of AstBill at http://astbill.com It gives you cost based CDR's out of the box. AstBill is an Open Source Web Based Billing, Routing and Management Software for Asterisk and VOIP. AstBill Provides pre and post paid billing services and have a calling card module. AstBill completely automates Asterisk and VOIP billing from start to finish. Key benefits are the Central Web-based installation, Credit Control on outgoing and incoming calls and the call routing module. Alternative you can use it as a low cost office or home PABX. -- Are Casillahttp://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultantshttp://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.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] a2billing / php agi debugging
HI, I have been playing around with the A2billing agi for a few days( I really think its a great application) and trying to understand its working. As far as I have understood, the only debugging mechanism is going through the logs it generates. It would be really cool if I can run the agi in debug mode and step through the code. I have been searching around and found nothing related to debugging agi's (specifcally php agi's). If this is possible can someone please tell me how it can be achieved. If its not possisble please do tell me why and what are the alternates. Regards, Danish ___ --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] A rather big setup.
Vedran Dakic ha scritto: How does Asterisk handle this kind of setup with one-two/cluster central server(s) and a bunch of other servers connected with IAX(2)? If you have local calls, do they go directly from phone to phone, do they go from phone to per-floor-Asterisk server, or they have to be interconnected via the main Asterisk server(s)/cluster? With SIP the default is to directly connect the phones once the call is setup (I think also in IAX), investigate canreinvite / nat. Of course you can't do call detail record for calls which aren't forced to pass from the server, see if it's a problem ... imho As for maintenance we have a dozen of pcs with asterisk installed, each of them is server for 8/10 sip phones and client to a central asterisk server which then connects to E1. Asterisk pcs are scattered around, they pass trough at least one natted network, usually two. Never a problem. Connecting all the SIP phones to SER load balancing to more than one asterisk server will make you learn a lot about sip internals, proxy, domains, authentication and other interesting stuff, but if you need to have a working sistems and can start from zero go iax and spare yourself a lot of frustration. /imho ___ --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] Dialplan pattern match discrepancy
On 11/25/05, Daniel Wright [EMAIL PROTECTED] wrote: Steve Davies wrote: Hi, This is probably just me mis-reading the documentation, but I have been led to believe that the '.' in extensions.conf means zero or more digits, such that exten = _X.,1,NoOp() Would trigger for either a single digit, or for a longer number (as long as it starts with a digit) In practice (I am using 1.0.7 and 1.0.9) the '.' seems to match *one* or more digits, so in the above example, a single digit is not matched as expected. Is this correct? A bug? Fixed in 1.2 ;-) ? The period is match 1 or more characters(can be a number or letter). So in your example, you are saying first match a number 0-9, then match any one or more characters. Dan Thanks Dan. It does make the extensions.conf warning that '_.' should be replaced with '_X.' a bit counter-intuitive then :) I completely understand why '_.' is risky, but '_X.' is not the same thing at-all! I assume that the only way to match one digit, followed by zero-or-more alphanumeric is to have 2 sets of rules: exten = _X,1,NoOp(One Digit) exten = _X.,1,NoOp(One Digit and more stuff) even if the same operation is to be carried out. Ho hum! 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
Re: [Asterisk-Users] Zaptel errors on Debian
On Sun, Nov 27, 2005 at 05:22:39PM -0300, Rodrigo Campos wrote: On 11/27/05, Geotrix [EMAIL PROTECTED] wrote: Hello, I am trying to install zaptel wcfxo with X101.P board on Debian sarge without success. (previously compiled and worked OK on Redhat kernel) and in debian ? for compiling it in debian i suggest you to install kernel-headers (tha same version of your running kernel), module-assistant and dpatch (install all with apt or aptitude) and zaptel and zaptel-source, of course :) m-a (module-assistant) and dpatch are dependencies of zaptel-source. (dpatch shouldn't really be, actually. Just because an incorrectly-laid makefile) When i compile it i copy the /boot/config-running.kernel.version to /usr/src/kernel-headers-running.kernel.version/.config and then make cd /usr/src; m-a build zaptel this will generate a .deb in . then just dpkg -i name.of.the.deb.file.generated and reboot. Then you should can modprobe zaptel, etc... shorter version: m-a a-i zaptel I personally don't like the text-based interface and rather work with the scriptable version: m-a -t -i a-i zaptel -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Zaptel errors on Debian
On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote: Well, thanks, it might be great your package yet I would like to know how to adapt. I wouldn't like to rewrite Debian neither Asterisk but is somebody able to advice how you define modules in zconfig.h or whatever ? Any tip ? Geo Why would you need to define modules? The package builds wcfxo. What exactly do you try to do? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] a2billing / php agi debugging
This is so simple, you are going to kick yourself. Type agi debug in the console. Thanks, Steve -Original Message- From: Danish Samad [mailto:[EMAIL PROTECTED] Sent: Monday, November 28, 2005 3:52 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] a2billing / php agi debugging HI, I have been playing around with the A2billing agi for a few days( I really think its a great application) and trying to understand its working. As far as I have understood, the only debugging mechanism is going through the logs it generates. It would be really cool if I can run the agi in debug mode and step through the code. I have been searching around and found nothing related to debugging agi's (specifcally php agi's). If this is possible can someone please tell me how it can be achieved. If its not possisble please do tell me why and what are the alternates. Regards, Danish ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Intel G729 Codec Install error on [EMAIL PROTECTED]
Dear Tzafrir Cohen, I paste the detail message for you,please see the below message. I wonder it may cause by different linux system. [EMAIL PROTECTED] speech-coding]# patch -p1 /tmp/ipp-050903.diff patching file G723.1/Makefile patching file G723.1/samples/codec_g723.c patching file G723.1/samples/g723_slin_ex.h patching file G723.1/samples/print_bytes.c patching file G723.1/samples/slin_g723_ex.h patching file G729-float/Makefile patching file G729-float/samples/codec_g729.c patching file G729-float/samples/decoder.c patching file G729-float/samples/encoder.c patching file G729-float/samples/g729_slin_ex.h patching file G729-float/samples/gen_test.c patching file G729-float/samples/my_dec.c patching file G729-float/samples/my_enc.c patching file G729-float/samples/slin_g729_ex.h patching file G729-float/samples/util_e.c patching file G729-float/samples/util_e.h patching file README.Asterisk patching file debian/asterisk-ipp-codecs.dirs patching file debian/asterisk-ipp-codecs.install patching file debian/changelog patching file debian/control patching file debian/rules patching file runme.sh [EMAIL PROTECTED] speech-coding]# chmod a+x runme.sh [EMAIL PROTECTED] speech-coding]# ls AMRWB G722.1 G728 G729-float ipplic.htm runme.sh debian G723.1 G729 GSMAMR README.Asterisk support.txt [EMAIL PROTECTED] speech-coding]# ./runme.sh Makefile:126: target `samples/util_e.o' given more than once in the same rule. Makefile:126: target `samples/util_e.o' given more than once in the same rule. Makefile:126: target `samples/util_d.o' given more than once in the same rule. Makefile:126: target `samples/util_d.o' given more than once in the same rule. rm -f ./api/decg729fp.o ./api/encg729fp.o ./api/owng729fp.o ./api/usc729fp.o ./vm/src/vm_thread_linux32.o ./samples/util_e.o ./samples/util_d.o ./sample s/codec_g729.o ./samples/encoder.o ./samples/util_e.o ./samples/util_e.o ./s amples/my_enc.o ./samples/decoder.o ./samples/util_d.o ./samples/util_d.o ./ samples/my_dec.o Makefile:126: target `samples/util_e.o' given more than once in the same rule. Makefile:126: target `samples/util_e.o' given more than once in the same rule. Makefile:126: target `samples/util_d.o' given more than once in the same rule. Makefile:126: target `samples/util_d.o' given more than once in the same rule. gcc -I./include -I./vm/include -I/opt/intel/ipp41/ia32_itanium/include -include /opt/intel/ipp41/ia32_itanium/tools/staticlib/ipp_w7.h -D__unix__ -Dlinux -Dlinu x32 -DNDEBUG -DLINUX32 -DNO_SCRATCH_MEMORY_USED -c -O6 -march=pentium4 -mcpu=pen tium4 -ffast-math -fomit-frame-pointer -osamples/util_e.o samples/util_e.c gcc -I./include -I./vm/include -I/opt/intel/ipp41/ia32_itanium/include -include /opt/intel/ipp41/ia32_itanium/tools/staticlib/ipp_w7.h -D__unix__ -Dlinux -Dlinu x32 -DNDEBUG -DLINUX32 -DNO_SCRATCH_MEMORY_USED -c -O6 -march=pentium4 -mcpu=pen tium4 -ffast-math -fomit-frame-pointer -osamples/util_d.o samples/util_d.c gcc -I./include -I./vm/include -I/opt/intel/ipp41/ia32_itanium/include -include /opt/intel/ipp41/ia32_itanium/tools/staticlib/ipp_w7.h -D__unix__ -Dlinux -Dlinu x32 -DNDEBUG -DLINUX32 -DNO_SCRATCH_MEMORY_USED -c -O6 -march=pentium4 -mcpu=pen tium4 -ffast-math -fomit-frame-pointer -osamples/codec_g729.o samples/codec_g72 9.c In file included from ./vm/include/vm_types.h:17, from ./vm/include/vm_thread.h:14, from samples/encoder.h:28, from samples/codec_g729.c:39: ./vm/include/sys/vm_types_linux32.h:36: error: syntax error before use_ast_cond _t_instead_of_pthread_cond_t ./vm/include/sys/vm_types_linux32.h:36: warning: no semicolon at end of struct o r union ./vm/include/sys/vm_types_linux32.h:37: warning: data definition has no type or storage class ./vm/include/sys/vm_types_linux32.h:40: error: syntax error before '}' token ./vm/include/sys/vm_types_linux32.h:40: warning: data definition has no type or storage class ./vm/include/sys/vm_types_linux32.h:51: error: syntax error before use_ast_mute x_t_instead_of_pthread_mutex_t
[Asterisk-Users] Problem with ADIT 600 and FXO configuration
I've looked through the archives of the mailing list for the last year and although informative I've not been successful at get this to work. We had a working Asterisk PBX system with 3 Digium X101P FXO lines and two TDM400P FXS cards. I've setup an ADIT 600 with an 8 port FXO card (and an 8 port FXS card not currently installed). We are going to be adding a T1 for incoming calls this week. I removed two of the X101P cards and installed a TE406P. I'm using Asterisk 1.0.9 (and matching zaptel, libpri) from tar files. /etc/zaptel.conf has this configuration: span=1,1,0,esf,b8zs,yellow span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs #Modular unit, first card is FXO fxsks=1-3 unused=4-8 #Modular unit, 1 FXS cards unused=9-16 unused=17-24 unused=25-48,49-72,73-96 fxsks=97 fxoks=98-101 fxoks=102-105 /etc/asterisk/zapata.conf has this: group = 0 signalling=fxs_ks context = incoming busydetect = yes overlapdial = no channel = 1-3 signalling=fxs_ks channel = 97 ;X100P group = 1 signalling = fxo_ks context = internal ;TDM400P callerid = Available 200 channel = 98-100 callerid = x channel = 101 ;TDM400P callerid = x channel = 102 callerid = x channel = 103 Parts of my adit configuration: -Setting slot a. set a:1 up set a:1 fdl none set a:1 lbo 4 set a:1 framing esf set a:1 id Inbound set a:1 linecode b8zs set a:1 loopdetect csu set a:1:1-24 side drop set a:1:1-24 type voice set a:1:1-24 signal ls set a:2 up set a:2 fdl none set a:2 lbo 1 set a:2 framing esf set a:2 id Outbound PBX set a:2 linecode b8zs set a:2 loopdetect csu set a:2:1-24 side drop set a:2:1-24 type voice set a:2:1-24 signal ls -Setting slot 1. set 1:1-8 signal lscpd set 1:1-8 txgain -3 set 1:1-8 rxgain -6 -Setting primary and secondary clock sources. set clock1 a:1 set clock2 internal -Setting the system idle pattern for DS0s. set idle 0xff -Making connections. connect a:2:1-3 1:1-3 Inbound calls just ring and ring (the leds on the ADIT change state) however asterisk doesn't respond. Attempts to make outgoing calls get: -- Executing Dial(SIP/202-ba07, Zap/g0/5551212) in new stack Nov 28 02:54:45 NOTICE[8627]: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Congestion(SIP/202-ba07, ) in new stack == Spawn extension (from-sip, 95942060, 3) exited non-zero on 'SIP/202-ba07' -- Executing Hangup(SIP/202-ba07, ) in new stack I've tried just about all combinations of gs/ls/ks for the signalling to no avail. Here is the output of status: status a:2:1-3 DS0 Rx AB Tx AB Signal T1 TP --- - - -- - -- a:2:1 01 01 LS TrafficN a:2:2 01 01 LS TrafficN a:2:3 01 01 LS TrafficN status 1:1-3 FXORx AB Tx AB Signal=T1 Sig T1 TP ---- - -- - -- 1:1 01 01 LSCPD = LS Traffic N 1:2 01 01 LSCPD = LS Traffic N 1:3 01 01 LSCPD = LS Traffic N show connect a:2:1-3 FromDescDescTo --- -- - - A:02:01LS VOICE DS0 -- FXOVOICE LSCPD 1:01 A:02:02LS VOICE DS0 -- FXOVOICE LSCPD 1:02 A:02:03LS VOICE DS0 -- FXOVOICE LSCPD 1:03 Can anyone spot what I've got wrong? Any suggestions or hints welcome. Thanks, William. ___ --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] a2billing / php agi debugging
HI Steve, Thanks for the reply. I have tried agi debug before. Actually by debugging I mean inserting breakpoints and stepping through the code. I have not been able to find any solution for this sort of debugging setup. Regards, DanishOn 11/28/05, Steve Totaro [EMAIL PROTECTED] wrote: This is so simple, you are going to kick yourself.Type agi debug inthe console.Thanks,Steve -Original Message- From: Danish Samad [mailto: [EMAIL PROTECTED]] Sent: Monday, November 28, 2005 3:52 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] a2billing / php agi debugging HI, I have been playing around with the A2billing agi for a few days( Ireally think its a great application) and trying to understand its working.As far as I have understood, the only debugging mechanism is going through the logs it generates.It would be really cool if I can run the agi in debug mode and step through the code. I have been searching around and found nothingrelated to debugging agi's (specifcally php agi's). If this is possible can someone please tell me how it can be achieved. If its not possisbleplease do tell me why and what are the alternates. Regards, Danish___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Legacy PBX integration problem
Hi, We are trying to integrate Asterisk in front of our existing legacy PBX: outside line -- FXO -- Asterisk PC -- FXS -- PBX -- Extensions Asterisk answers outside calls and the IVR asks the user to dial extension #. The problem is, that when Asterisk forwards calls from the outside line to the old PBX (using DIAL()) the PBX answers, and as far as * is concerned, the call is answered. Asterisk then dials the extension number, but even if the extension is busy or no one answers, the call is answered as far as * is concerned. We want Asterisk to: -DIAL the PBX -wait for the PBX to answer -dial extension number -If there is no answer at this extension start voicemail. Any ideas? How can we detect when an extension is busy/no answer AFTER PBX already answered the call? Thanks,Kupchinetsky Dmitry ___ --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] Legacy PBX integration problem
Hi, We are trying to integrate Asterisk in front of our existing legacy PBX: outside line -- FXO -- Asterisk PC -- FXS -- PBX -- Extensions Asterisk answers outside calls and the IVR asks the user to dial extension #. The problem is, that when Asterisk forwards calls from the outside line to the old PBX (using DIAL()) the PBX answers, and as far as * is concerned, the call is answered. Asterisk then dials the extension number, but even if the extension is busy or no one answers, the call is answered as far as * is concerned. We want Asterisk to: -DIAL the PBX -wait for the PBX to answer -dial extension number -If there is no answer at this extension start voicemail. Any ideas? How can we detect when an extension is busy/no answer AFTER PBX already answered the call? Thanks,Kupchinetsky Dmitry ___ --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] Presence + Eyebeam + Asterisk 1.2
Hi, anyone managed to get a Presence Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe this should be paritally supported now in 1.2 ? regards Mark___ --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] Legacy PBX integration problem
Hi, We are trying to integrate Asterisk in front of our existing legacy PBX: outside line -- FXO -- Asterisk PC -- FXS -- PBX -- Extensions Asterisk answers outside calls and the IVR asks the user to dial extension #. The problem is, that when Asterisk forwards calls from the outside line to the old PBX (using DIAL()) the PBX answers, and as far as * is concerned, the call is answered. Asterisk then dials the extension number, but even if the extension is busy or no one answers, the call is answered as far as * is concerned. We want Asterisk to: -DIAL the PBX -wait for the PBX to answer -dial extension number -If there is no answer at this extension start voicemail. Any ideas? How can we detect when an extension is busy/no answer AFTER PBX already answered the call? Thanks, Kupchinetsky Dmitry ___ --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] A rather big setup.
Vedran, Email me off topic and I can provide you some case studies of different providers for your review. [EMAIL PROTECTED] Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 Website: http://www.upperclassman.net This e-mail and any of its attachments may contain sensitive information, which is privileged, confidential, or subject to copyright belonging to RW Management Inc, Universal Holdings LLC or Upperclassman LLC. This e-mail is intended solely for the use of the individual or entity to which it is addressed. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution, copying, or action taken in relation to the contents of and attachments to this e-mail is strictly prohibited and may be unlawful. If you have received this e-mail in error, please notify the sender immediately and permanently delete the original and any copy of or printout of this e-mail. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jan Saell Sent: Monday, November 28, 2005 12:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] A rather big setup. if this is a brand new thing you can force the phones on people and then you can to provisioning remotly of for instance Grandstream so they can change the config themself. By forcing a common set of codex you can avoid cpu overhead of translation so you only have to think of teh datashuffle. Bu doing god work at the dialpla you make shure that all the calls thats internal never hit the main pbx'es in the celler and oly use them for outgoing! Best regards jan --On Monday, November 28, 2005 04:22:09 AM 퍝 Vedran Dakic [EMAIL PROTECTED] wrote: Hello, Those people currently aren't using any kind of phones, but the investment company that has this building in the works wants to deliver everything for them so they just have to - move in and do business. What worries me is the fact that when you have 100-200 offices - they're used to having 2-3 lines only for them - one for fax, two for voice, etc. So, in a way, having in mind around 200-300 outbound calls at peak time is pretty much normal. Also, when you think of the number of phones - it would only be normal to assume for people to have up to 1000 internal phone conversations peak (the less transcoding - the better, of course). I have a freedom of making whatever I want, so I can have a separate LAN for VoIP purposes only - a bunch of dedicated patch panels, VLANs on Cisco switches, or whatever. I'm just considering this setup way before it has to go online because of the price of traditional PBX for this kind of setup which can only make you hurl. And you know how much potential upgrades cost for a setup like this - a traditional PBX can be a nightmare :( Cheers, Vedran. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans Witvliet Sent: Monday, November 28, 2005 12:08 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] A rather big setup. I think there is more to consider. One or two fat machines in the basement forr connecting to the PSTN is very fine. But are all the people allready using voip handsets, or old fashioned analoge handsets? If so, you need quite a large number of channelbanks. You speak of 300/1500 concurrent phone calls? If so how many handsets are you considering? Is the lan capable of handling this load? Is the lan 100% dedicated for voip, or are there a bunch of servers/workstations also using this lan? Interesting project Hans ___ --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 -- +-- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int 19 58 25 15 Fax 19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B ___ --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] Presence + Eyebeam + Asterisk 1.2
Don't waste your time asterisk does not support presence --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit : Hi, anyone managed to get a Presence Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe this should be paritally supported now in 1.2 ? regards Mark ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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
[Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?
Hi it's possible to upgrade the firmware of a cisco 7910 with asterisk ? he have a other solution for upgrade it without callmanager ? thansk for your 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] Re: Presence + Eyebeam + Asterisk 1.2
Are you sure? I've got it working with Eyebeam, showing me just who is available and who isn't. http://www.voip-info.org/wiki-Asterisk+phone+snom A couple of pages down you'll see this: SNOM SUBSCRIBE/NOTIFY support for monitoring extension states The methods and configuration here are also valid for Eyebeam. BB harry gaillac [EMAIL PROTECTED] uttered the following thing: Don't waste your time asterisk does not support presence --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit : Hi, anyone managed to get a Presence Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe this should be paritally supported now in 1.2 ? regards Mark ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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] Upgrade Cisco 7910 with Asterisk ?
Noc Phibee ha scritto: it's possible to upgrade the firmware of a cisco 7910 with asterisk ? You need the legal firmware upgrade file download the chan_sccp code from http://chan-sccp.berlios.de configure it and use the imageversion param to upgradde the phone firmware. Of course you need a tftpserver and if you run a tftpserver you just need a SEPmac to upgrade the phone So the correct answer is: you don't need a CCM nor asterisk to upgrade a cisco phone firmware. You just need the firmware file, a tftpserver and a configuration file (SEPmac) take a look here http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx Sergio ___ --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: Presence + Eyebeam + Asterisk 1.2
I'm sure look at rfc3265 (SUBSCRIBE/NOTIFY) which is not support by asterisk. How can you monitor the states of the buddies ? Harry --- Ben Buxton [EMAIL PROTECTED] a écrit : Are you sure? I've got it working with Eyebeam, showing me just who is available and who isn't. http://www.voip-info.org/wiki-Asterisk+phone+snom A couple of pages down you'll see this: SNOM SUBSCRIBE/NOTIFY support for monitoring extension states The methods and configuration here are also valid for Eyebeam. BB harry gaillac [EMAIL PROTECTED] uttered the following thing: Don't waste your time asterisk does not support presence --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit : Hi, anyone managed to get a Presence Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe this should be paritally supported now in 1.2 ? regards Mark ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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] Re: Presence + Eyebeam + Asterisk 1.2
aha, it was the subscribecontext= that was missing. Basic presence works fine now. Offline, online, on the phone :-) What about IM, did you have that working too ? thanks Mark Ben Buxton [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 28/11/2005 11:59 PM Please respond to Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To asterisk-users@lists.digium.com cc Subject [Asterisk-Users] Re: Presence + Eyebeam + Asterisk 1.2 Are you sure? I've got it working with Eyebeam, showing me just who is available and who isn't. http://www.voip-info.org/wiki-Asterisk+phone+snom A couple of pages down you'll see this: SNOM SUBSCRIBE/NOTIFY support for monitoring extension states The methods and configuration here are also valid for Eyebeam. BB harry gaillac [EMAIL PROTECTED] uttered the following thing: Don't waste your time asterisk does not support presence --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit : Hi, anyone managed to get a Presence Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe this should be paritally supported now in 1.2 ? regards Mark ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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
[Asterisk-Users] Re: Re: Presence + Eyebeam + Asterisk 1.2
Can't say I've actually tried IM, but Ill give it a go sometime. I think the wiki needs updating on all this...the eyebeam page is very incomplete on subscribe, im, etc. BB Mark van Kerkwyk [EMAIL PROTECTED] uttered the following thing: aha, it was the subscribecontext= that was missing. Basic presence works fine now. Offline, online, on the phone :-) What about IM, did you have that working too ? thanks Mark Ben Buxton [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 28/11/2005 11:59 PM Please respond to Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To asterisk-users@lists.digium.com cc Subject [Asterisk-Users] Re: Presence + Eyebeam + Asterisk 1.2 Are you sure? I've got it working with Eyebeam, showing me just who is available and who isn't. http://www.voip-info.org/wiki-Asterisk+phone+snom A couple of pages down you'll see this: SNOM SUBSCRIBE/NOTIFY support for monitoring extension states The methods and configuration here are also valid for Eyebeam. BB harry gaillac [EMAIL PROTECTED] uttered the following thing: Don't waste your time asterisk does not support presence --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit : Hi, anyone managed to get a Presence Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe this should be paritally supported now in 1.2 ? regards Mark ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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 ___ --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] troubles with voicemail
hi list! I've configured some voicemailboxes and at the beginning everything was working fine. In the past few days, evetime i want to hear the messages, recorded on my box the following lines come up the asterisk logfile: Nov 28 14:12:31 WARNING[25446]: file.c:508 ast_openstream_full: File digits/1F does not exist in any format Nov 28 14:12:31 WARNING[25446]: file.c:820 ast_streamfile: Unable to open digits/1F (format ulaw): No such file or directory == Spawn extension (sip, 82103, 1) exited non-zero on 'SIP/soft_leeb-273f' Nov 28 14:12:31 ERROR[25446]: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : No such file or directory Can anybody imagine why this error occurs? Best regard, 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
Re: [Asterisk-Users] stop asterisk when Idle
Thank you for your asnswer I found that between 7 and 8 in the mornig I have a low load of my box. I modify my script in this way: asterisk02:/ # cat /closeasteriskandreboot.sh #!/bin/bash echo chiusura schedulata when convenient di asterisk /usr/sbin/asterisk -rx stop when convenient mypid=$(pidof asterisk) echo mypid=${mypid} while [ ${mypid}empty != empty ] ; do echo Il pid di asterisk esiste ancora ( = ${mypid} ) , aspetto altri 10 secondi # adesso dormo per dieci secondi per aspettare che asterisk chiuda tutto bene sleep 10 mypid=$(pidof asterisk) echo mypid=${mypid} done echo Il pid di asterisk non esiste piu' , riavvio il pc echo reboot del server, ciao ciao /sbin/reboot exit *** I think that it should be possible to reach a convenient (idle) time. I could also implement a sort of timeout (if in half an hour asterisk is still up, then send stop now command and then reboot)... On the other side, I didn'i install any monitoring utility on my box (I don't konw them) The bigger problem is that it is a productivity box, and to do a lot of experiments is not so easy So I think that I will leave my box as it is now, and I will check in the future if this strange oh323 problem will be found elsewhere and resolved. thanks all, Andrea Leif Neland [EMAIL PROTECTED] Sent by: To asterisk-users-bo Asterisk Users Mailing List - [EMAIL PROTECTED] Non-Commercial Discussion m.com asterisk-users@lists.digium.com cc 26/11/2005 17.35 Subject Re: [Asterisk-Users] stop asterisk when Idle Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I still continue to reboot my asterisk box everyday. I posted a message on November 22, but it was on another thread and no one answered me, so I try again here, where a lot of people told be I was a bad administrator (Like a Windows administrator and I don'0t want to resolve my problem) Actually I would like to resolve my problem, but I am not able to do this, so I ask help to anybody who can help me, and repost my last of 22/11/2005 In short, my problem is that, after one or two days of running, chan oh323 suddendly disappear from asterisk box, without giving any warning / error In example, you type oh323 show stats at 11 o'clock , and get an answer from asterisk, about usage of oh323 At 12, without doing anything to the box or to the asterisk, you type the same command, and you get a No such command 'oh323' (type 'help' for help) If you type help, no oh323 commands are available. If you quit asterisk, (STOP NOW) and restart asterisk , no oh323 channel command is available if you reboot the machine everything is again fine ! It is so a crazy situation that to reboot appears (to me) the best thing (I am sorry about this) If you really need to have oh323, then you should test say every 5 minutes or so, and then shutdown asterisk and reboot. if asterisk -r -x oh323|grep help then echo oh323 missing|mail administrator asterisk -r -x stop now reboot fi Ii is better to disconnect the existing users if they can not use the box without oh323 Do you have some kind of monitoring running? Like Big Brother or nagios? It might be interesting to see when oh323 dies. Perhaps you could also use mrtg to graph usage levels to see if there is some kind of correlation between usage and oh323 fatality Leif ___ --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 --
Re: [Asterisk-Users] Sangoma problems!?
hat is the content of your wanpipe?.conf files ?On 11/26/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Stig Even Larsen wrote: I'm having problems connecting my Sangoma cards to our PRI (E1) interface. It seems that the card get connected (green led), but Asterisk reports: Status: Provisioned, Down, Active. When I install another Sangoma card on the same system (pri_net), and connect the two cards with a PRI cross-over cable both cards get connected (green led) and Asterisk reports both spans: Status: Provisioned, UP, ActiveYour telco has not brought the D-channel on your PRI circuit to 'up' state yet. Plug the card back in, make the red alarm go away, and thencall them and tell them you are ready to use the circuit.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal BielickiHalo Kwadrat Sp. z o.o.http://www.asterisk.pl/http://www.openpbx.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
Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandsp app_rxfax
I looked into why I can't get the original DID number called when a fax is detected (so I can later route to the correct email address). There is a variable called FAXEXTEN that is created when a fax is detected, but it is not being populated with the original extension / did number called. It always has 's' as the original extension. Does anyone know how to fix this? I just want to use the zap fax detection so each person can have their personal DID number accept faxes and calls, then route the fax to their email address based on the DID number called. - James James Armstrong wrote: In this example faxdetect is overwriting the DID. So the trick then would be to somewhere early-on in your dialplan grab the DID into some variable, and then restore it after the fax detection occurs... [default] exten = _X.,1, SetVar(ORIGEXTEN=${EXTEN}) exten = s,2,Wait(3) . exten = fax,1,Dial(IAX2/ttyIAX0/${ORIGEXTEN}) Lee. I am having no luck here. It seems the fax detection is overriding everything. I added the above and it never gets called. I guess it might be time to look at the Asterisk code and see if I can create another variable before the redirect happens. -- Starting simple switch on 'Zap/1-1' -- Redirecting Zap/1-1 to fax extension -- Executing Dial(Zap/1-1, IAX2/999/|20) in new stack ___ --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] Problem forwarding zap to sip
I have a problem when i have asterisk connected to voxtream parlay i60 PRA port. When call is received to asterisk and forwarded to SIP IP gsm gateway call is always disconnected with cause 102. # 102 Recovery on timer expiry This cause indicates that a procedure has been initiated by the expiry of a timer in association with ETS 300 102-1 error handling procedures. Call is not disconnected if user answers the call in this 18 seconds. I think the problem is that all messages from ip gateway don't get forwardet to PRI channel and call is disconnected. What can be done to solve this problem greetings mk ___ --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] Sangoma problems!?
Hi Stig, When I install another Sangoma card on the same system (pri_net), and connect the two cards with a PRI cross-over cable both cards get connected (green led) and Asterisk reports both spans: Status: Provisioned, UP, Active It does look your installation is fine, i.e you are being able to send/receive packets. One way to confirm this is to perform a trace on span 1: (with the E1 cross-over cable between the two ports) 1. In the Asterisk CLI, type: pri intense debug span 1. You should see that packets are being transmitted in both directions, i.e. Some packets will have On the left hand side and others will have 2. In another command window (linux bash), type: Wanpipemon -I w1g1 -c trd This is a sangoma tool that will trace the d-channel of span 1. Check if you are having both incoming and outgoing packets. If for both tests, you are receiving and transmitting packets, this means your connection is fine, but your telco line is not provisioned yet. Contact your telco and tell them that you are not receiving packets. If you are receiving and transmitting packets from the wanpipemon trace, but only transmitting or only receiving in the asterisk CLI, then this that the Sangoma card is not interfacing properly with zaptel. This usually happens because you did not re-compile zaptel after installing wanpipe. Please re-compile zaptel. The correct installation sequence is: 1. Zaptel, Libpri, Asterisk 2. wanpipe 3. recompile zaptel. 4. perform configurations (wancfg, zaptel.conf etc.. ) Do not hesitate to contact me if you have any questions. Regards, David Yat Sin Sangoma Technologies (905) 474-1990 x119 (800) 388-2475 x119 Fax: (905) 474 9223 MSN: [EMAIL PROTECTED] Email: [EMAIL PROTECTED] Website: www.sangoma.com -Original Message- From: Stig Even Larsen [mailto:[EMAIL PROTECTED] Sent: November 25, 2005 3:21 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sangoma problems!? I'm having problems connecting my Sangoma cards to our PRI (E1) interface. It seems that the card get connected (green led), but Asterisk reports: Status: Provisioned, Down, Active. When I install another Sangoma card on the same system (pri_net), and connect the two cards with a PRI cross-over cable both cards get connected (green led) and Asterisk reports both spans: Status: Provisioned, UP, Active What could be wrong? I'm I missing something? zaptel.conf ;Telco PRI span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 ;PBX span=2,0,0,ccs,hdb3,crc4,yellow bchan=32-46,48-62 dchan=47 zapata.conf (minimal) [channels] context = from-pstn switchtype = euroisdn group=0 signalling = pri_cpe channel = 1-15 channel = 17-31 context = alcatel switchtype = euroisdn group=1 signalling = pri_net channel = 32-46 channel = 48-62 Best regards, Stig Even Larsen David Yat Sin Sangoma Technologies (905) 474-1990 x119 (800) 388-2475 x119 Fax: (905) 474 9223 MSN: [EMAIL PROTECTED] Email: [EMAIL PROTECTED] Website: www.sangoma.com -Original Message- From: Stig Even Larsen [mailto:[EMAIL PROTECTED] Sent: November 25, 2005 3:21 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sangoma problems!? I'm having problems connecting my Sangoma cards to our PRI (E1) interface. It seems that the card get connected (green led), but Asterisk reports: Status: Provisioned, Down, Active. When I install another Sangoma card on the same system (pri_net), and connect the two cards with a PRI cross-over cable both cards get connected (green led) and Asterisk reports both spans: Status: Provisioned, UP, Active What could be wrong? I'm I missing something? zaptel.conf ;Telco PRI span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 ;PBX span=2,0,0,ccs,hdb3,crc4,yellow bchan=32-46,48-62 dchan=47 zapata.conf (minimal) [channels] context = from-pstn switchtype = euroisdn group=0 signalling = pri_cpe channel = 1-15 channel = 17-31 context = alcatel switchtype = euroisdn group=1 signalling = pri_net channel = 32-46 channel = 48-62 Best regards, Stig Even Larsen ___ --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 mailing list: AstCallCenters
Hello list, this is just an announce of a new mailing list dedicated to deploying, running and managing real- world Asterisk-based call centers. The mailing list is in English and allows knowledge sharing for this very important - and yet somehow less considered - Asterisk deployment area. The homepage is located at http://groups.yahoo.com/group/astcallcenters/ Yours, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] FAX difference IAXModem / Hylafax and spandsp app_rxfax
Found out why there is no original DID set. It looks like while waiting for the incoming digits timeout (DID), we are getting the fax tone detect and it is sending a digit 'f' which immediately starts the fax extension before the incoming DID has been saved. Is there a way to set in the zap config how many digits we are receiving so there is not timeout waiting for the last digit? We only get 7 digits coming in. - James James Armstrong wrote: I looked into why I can't get the original DID number called when a fax is detected (so I can later route to the correct email address). There is a variable called FAXEXTEN that is created when a fax is detected, but it is not being populated with the original extension / did number called. It always has 's' as the original extension. Does anyone know how to fix this? I just want to use the zap fax detection so each person can have their personal DID number accept faxes and calls, then route the fax to their email address based on the DID number called. - James James Armstrong wrote: In this example faxdetect is overwriting the DID. So the trick then would be to somewhere early-on in your dialplan grab the DID into some variable, and then restore it after the fax detection occurs... [default] exten = _X.,1, SetVar(ORIGEXTEN=${EXTEN}) exten = s,2,Wait(3) . exten = fax,1,Dial(IAX2/ttyIAX0/${ORIGEXTEN}) Lee. I am having no luck here. It seems the fax detection is overriding everything. I added the above and it never gets called. I guess it might be time to look at the Asterisk code and see if I can create another variable before the redirect happens. -- Starting simple switch on 'Zap/1-1' -- Redirecting Zap/1-1 to fax extension -- Executing Dial(Zap/1-1, IAX2/999/|20) in new stack ___ --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] Realtime Extensions Problem
Hi Guys, Having a little problem with Realtime Extensions. I've created the table, (using the same database as i use for realtime peers/users), however when a call comes through, the CLI shows the following warning:- Nov 28 15:13:08 WARNING[7522]: res_config_mysql.c:623 mysql_reconnect: MySQL RealTime: Unable to select database: extensions. Still Connected. and the call drops. Has anyone seen this before, or suggest a solution? Thanks Dan www.TextOver.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] AGI script always returning 0
Hello, I've noticed my AGI, in Perl, was always returning 0 even if exit from it with something else than 0. On http://www.voip-info.org/wiki/view/Asterisk+cmd+AGI, it's said : [AGI] Returns -1 on hangup or if application requested hangup, or 0 on non-hangup exit. But I tried also to hang up during the execution of the AGI and it's the same. I also tried to use setcallback like that: - # send callback reference $AGI-setcallback(\callback); # our callback function sub callback(){ my ($returncode) = @_; warn The call has ended ($returncode)\n; exit($returncode); } - The $returncode is set to -1 when I hung up my phone, but in Asterisk logs there is still AGI Script pps.agi completed, returning 0. Is that the normal behavior of Asterisk? Or is it possible to get -1 or any other code when finishing the execution of an AGI? Thanks, Benoit -- Benoit Merouze Network Software Developer [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: Re: Re: [Asterisk-Users] Zaptel errors on Debian
It should build wcfxo. Not trying anything special. I just follow the procedure ! When I reboot I have: ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded !!! HiSax: Linux Driver for passive ISDN cards HiSax: Version 3.5 (module) HiSax: Layer1 Revision 2.46.2.5 HiSax: Layer2 Revision 2.30.2.4 HiSax: TeiMgr Revision 2.20.2.3 HiSax: Layer3 Revision 2.22.2.3 I don't need ISDN, I do not configure any ISDN and I have no ISDN BRIstuffed or whatever Is ISDN susbsystem needed for using fxo devices using fxs signalling with Asterisk ? than Zapata Telephony Interface Registered on major 196 wcfxo: disagrees about version of symbol zt_receive wcfxo: Unknown symbol zt_receive wcfxo: disagrees about version of symbol zt_ec_chunk wcfxo: Unknown symbol zt_ec_chunk wcfxo: disagrees about version of symbol zt_transmit .. Testing modprobe zaptel = OK zaptel driver but not wcfxo and ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Yet my config is OK. = Installind Asterisk make install compiling OK but errors on zap .. chan_zap.c:8935: error: dereferencing pointer to incomplete type chan_zap.c:8936: error: dereferencing pointer to incomplete type chan_zap.c:8950: error: dereferencing pointer to incomplete type .. On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote: Well, thanks, it might be great your package yet I would like to know how to adapt. I wouldn't like to rewrite Debian neither Asterisk but is somebody able to advice how you define modules in zconfig.h or whatever ? Any tip ? Geo Why would you need to define modules? The package builds wcfxo. What exactly do you try to do? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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: Re: Re: [Asterisk-Users] Zaptel errors on Debian
It should build wcfxo. Not trying anything special. I just follow the procedure ! When I reboot I have: ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded !!! HiSax: Linux Driver for passive ISDN cards HiSax: Version 3.5 (module) HiSax: Layer1 Revision 2.46.2.5 HiSax: Layer2 Revision 2.30.2.4 HiSax: TeiMgr Revision 2.20.2.3 HiSax: Layer3 Revision 2.22.2.3 I don't need ISDN, I do not configure any ISDN and I have no ISDN BRIstuffed or whatever Is ISDN susbsystem needed for using fxo devices using fxs signalling with Asterisk ? than Zapata Telephony Interface Registered on major 196 wcfxo: disagrees about version of symbol zt_receive wcfxo: Unknown symbol zt_receive wcfxo: disagrees about version of symbol zt_ec_chunk wcfxo: Unknown symbol zt_ec_chunk wcfxo: disagrees about version of symbol zt_transmit .. Testing modprobe zaptel = OK zaptel driver but not wcfxo and ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Yet my config is OK. = Installind Asterisk make install compiling OK but errors on zap .. chan_zap.c:8935: error: dereferencing pointer to incomplete type chan_zap.c:8936: error: dereferencing pointer to incomplete type chan_zap.c:8950: error: dereferencing pointer to incomplete type .. On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote: Well, thanks, it might be great your package yet I would like to know how to adapt. I wouldn't like to rewrite Debian neither Asterisk but is somebody able to advice how you define modules in zconfig.h or whatever ? Any tip ? Geo Why would you need to define modules? The package builds wcfxo. What exactly do you try to do? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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
[Asterisk-Users] Download Ringtones for 7960's?
Does anybody know where I can download ringtones for Cisco 7960's? Need to be .pcm files. ___ --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] Can 'spandsp' ack as an intermediary between a fax machine and a TDM400P?
Hi, I understand that a fax machine cannot connect through a Digium TDM400p card (FXS connected to fax and FXO connected to a pots line) but can spandsp send and receive faxes as an intermediary between the pots line and the fax machine. Thanks ___ --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] sound problem, please help!
Hi, I have noticed that most mobile phones (GSM and CDMA at least) seem to have a tendency to interrupt the incoming audio stream when the microphone levels get louder than a certain threshold (such as when you are speaking into it). I do not know exactly why this happens, nor whether it is something that happens in the handset (the handset mutes the incoming audio, which doesn't make much sense) or in the network (maybe to try to save bandwidth?). Also, the GSM codec is intended to intelligibly encode human speech, not general audio signals such as music. GSM uses approximately 6-8 times less bandwidth than G.711u/a, and the degradation in the quality of music and other non-speech signals is one of the tradeoffs that we make in order to not have the cost of mobile calls increase 6-8 times over. Maybe if Europe was as backward with its mobile systems as the United States currently is, you could try to get your users to call your system from analog mobile phones... They don't use digital compression (by definition, I suppose) and would probably work much better for your specific application ;). -Rusty On 11/25/05, Esteban Maestre [EMAIL PROTECTED] wrote: Hi all!I have a strange problem when using asterisk. I have configured asteriskto receive calls (FX0). In my configuration, I want asterisk to play musicwhileI record the caller's speech. If the caller does the call from a fixed line telephone, there is no problem, but in case the caller does thecall from a mobile GSM phone, the quality of the music he hears becomes sobad, and even more when he speaks.I have tried several codecs. Any idea or advice?thanks in advance,-esteban-___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users 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] Trunk SIP howto ?
Hi anyone know if a Trunk SIP howto are created ? I have 8 VoIP account with for all 1 login/pass per number. i want add into my asterisk but not know where ;=) Other questions: my supplierhave a dns:sip.phonesystems.net this name have 2 IP address it's not a problems for Asterisk that he have registred on the first ip address and receive information of the second ? Thanks for your 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
Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?
Thanks sergio for your answer. But cisco france say me that i cant' bye SmartNet contract on this product. Only one solution are possible: Bye a special contract at $180.00 ... Pff i can bye a new equipment with this price hihihi i can't guest the latest firmware, for me i thinks that the solution are buy new voip phone and put the 7910 in Dead If anyone know a solution for get the latest firmware, mail me Bye Sergio Chersovani a écrit : Noc Phibee ha scritto: it's possible to upgrade the firmware of a cisco 7910 with asterisk ? You need the legal firmware upgrade file download the chan_sccp code from http://chan-sccp.berlios.de configure it and use the imageversion param to upgradde the phone firmware. Of course you need a tftpserver and if you run a tftpserver you just need a SEPmac to upgrade the phone So the correct answer is: you don't need a CCM nor asterisk to upgrade a cisco phone firmware. You just need the firmware file, a tftpserver and a configuration file (SEPmac) take a look here http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx Sergio ___ --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] ip phones
Hi all, Does anybody have any info on a decent quality sip hard phone that is headset compatible? Thank you 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] How does DTMF get sent over a PRI in Asterisk
I am trying to trouble shoot some problems with DTMF over PRI. I have a digium wct1xxp card and these lines in extensions.conf: exten = 5556000,1,Record(testtone:gsm) exten = 5556000,2,Wait(2) exten = 5556000,3,Playback(testtone) I called in over the PSTN --to-- Asterisk. I did a pri debug, from asterisk 1.2.0 console and a few debug lines came up when the call connects (but not all the q931 message that should have been there). But nothing came up when hitting the dtmf keys. Also when I listen to the play back the recorded tones are little more then chirps not long enough for a human ear to distinguish tone. My two question are how does DTMF get send over a PRI (inband? q931 message?) and how would I go about seeing the duration that was sent to me so that I know weather the problem is Asterisk or my telcos! ___ --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] Upgrade Cisco 7910 with Asterisk ?
Noc Phibee wrote: Thanks sergio for your answer. But cisco france say me that i cant' bye SmartNet contract on this product. You can, but only in the US I believe. I've never found any deal less than £150 (UK). Only one solution are possible: Bye a special contract at $180.00 ... Pff i can bye a new equipment with this price hihihi It's not so bad... you do get access to firmware to all cisco devices with that, so if you have more than one device it becomes worth it. You also get access to cisco TAC. Their support sucks in my experience, but it's far better than having no support at all when your hardware fails (cisco won't even give a manufacturers guarantee unless you have a smartnet contract). Tony ___ --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] Upgrade Cisco 7910 with Asterisk ?
Noc Phibee ha scritto: But cisco france say me that i cant' bye SmartNet contract on this product. why not? You can buy those smartnet contract via internet. You just need to mail US cisco and ask them for the contract activation. Only one solution are possible: Bye a special contract at $180.00 ... buy CON-SW-VPKG1 59 euros in europe Pff i can bye a new equipment with this price hihihi yep that is cisco i can't guest the latest firmware, for me i thinks that the solution are buy new voip phone and put the 7910 in Dead Yes you are right Sergio ___ --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] sound problem, please help!
Hello, Rusty! Thanks for your reply... You have been the only one ;) Hahaha, it could become dangerous... ;) Well, I have been investigating it a little bit, and I guess the main reason is something related to what you have pointed in the first paragraph. I've found some interesting information here: http://www.privateline.com/PCS/GSM08.html There is an interesting article I've already tried to post, but toy can find it easily in the www above. regards, -esteban- I don't think you'd want to hear most of what I say when I am on hold, especially if I am on hold with a telephone company! It tends to be somewhat on the profane side. *grin* -Rusty On 11/25/05, Esteban Maestre [EMAIL PROTECTED] wrote: kind of... ;) I want to know what the people say when they are waiting... :P do you have any idea on what the problem could be? -esteban- Original Message From: Esteban Maestre [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, November 25, 2005 11:22 AM Subject: [Asterisk-Users] sound problem, please help! Hi all! I have a strange problem when using asterisk. I have configured asterisk to receive calls (FX0). In my configuration, I want asterisk to play music while I record the caller's speech. Dialup-karaoke? :-) Leif ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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] ip phones
The only one I can think of to decent price level is the Grandstream GXP 2000. Also have headset jack¨ Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Fraser Sent: den 28 november 2005 17:27 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ip phones Hi all, Does anybody have any info on a decent quality sip hard phone that is headset compatible? Thank you 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 ___ --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] Upgrade Cisco 7910 with Asterisk ?
Thanks all for your answer ... all smartnet contrat have access to all firmware in voip ? thanks Ryan Amos a écrit : Cisco phones are not ideal for single-phone setups. If you were to have a lot of them, a $180 support contract is no big deal... However, for Europeans, there should be an $8 online-only support contract that gives you access to file downloads only. Being an American, This should be enough, however if you are only wanting a small number of phones you might want to look elsewhere. The main advantage of Cisco's phones comes when installing a large number of them, as the central management is ideal in an office PBX environment. Try this part number though: CON-SNT-PKG1-VS Supposedly costs 66 euros from wstore.fr (I found this in an old e-mail asking about smartnet contracts on the chan_sccp mailing lists.) Best of luck! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noc Phibee Sent: Monday, November 28, 2005 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ? Thanks sergio for your answer. But cisco france say me that i cant' bye SmartNet contract on this product. Only one solution are possible: Bye a special contract at $180.00 ... Pff i can bye a new equipment with this price hihihi i can't guest the latest firmware, for me i thinks that the solution are buy new voip phone and put the 7910 in Dead If anyone know a solution for get the latest firmware, mail me Bye ___ --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] IAX jitterbuffer and trunking settings between 1.0.9 and 1.2
Hello all, Since upgrading a couple of our servers to 1.2 I've noticed problems when talking to users on 1.0.9 servers. The servers are connected via IAX2 with trunking and jitter buffer enabled (jitter buffer on default settings). Reading through posts in the list archives, there are a number referring to problems with 1.0.9 and the jitter buffer when used in conjunction with trunking. Are there any recommended settings to use for the following scenarios? 1.0.9 - 1.0.9 1.2.0 - 1.0.9 1.2.0 - 1.2.0 Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] Download Ringtones for 7960's?
On Monday 28 Nov 2005 15:39, Frank McCarthy wrote: Does anybody know where I can download ringtones for Cisco 7960's? Need to be .pcm files. Download the ringtone generator from Grandstream and make your own. B ___ --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 phones
On Mon, 28 Nov 2005 08:26:57 -0800, John Fraser wrote: Hi all, Does anybody have any info on a decent quality sip hard phone that is headset compatible? Thank you John Aastra 480i ( Ilove it!), Polycom IPx00/x01 series, Snom's all provide for headsets. Most via RJ style connections, some (SNOM) both RJ and 2.5mm or 3.5mm jacks. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --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] how to stop ringing while talking
hi,while a user talking on, if he gets a new call from queue he hears some noises about this call. I think this happens to inform user about new coming call. But it boring user... So can i stop this noise?Thanks.-ek Yahoo! Music Unlimited - Access over 1 million songs. Try it free.___ --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 it mean I was blocked by STUN?
Hi. I think the best way to do it, is just a IAX2 between the 2 *'s. Regards. Juan. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Hiu Yen Onn Enviado el: Domingo, 27 de Noviembre de 2005 10:50 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Does it mean I was blocked by STUN? Hi all, I have 2 respectively networks, LAN A and LAN B, connected via my wireless links and routers. I have setup an asterisk machine at LAN A. It works fine when i was in LAN A. But, when i was in LAN B, xlite client can get connected to the server. But, it has no sound when i try to make an echo test. Does it mean, i was blocked by STUN? I have a wild search on google, i found that asterisk doesnt really support STUN. What is the workaround to make two network clients enjoy the intercalling via asterisk? IAX please advise... thanks ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.8/184 - Release Date: 27/11/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.8/184 - Release Date: 27/11/2005 ___ --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] Upgrade Cisco 7910 with Asterisk ?
You can, but only in the US I believe. I've never found any deal less than £150 (UK). I was quoted £36 a couple of weeks ago by one of the Cisco resellers a google search provided me with, if that's any help. I can't remember the company name I'm afraid... Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] Call progress from sip gsm gateway to pri interface - doesn't get through
Hi, we have following setup : PBX - Parlay -ISDN PRI- Asterisk -SIP- GSM Gateway Call comes from PBX through Parlay to Asterisk and it routes it over SIP to GSM gateway. GSM gateway gives back call progress (it takes some time to ring or get through), but this info won't get back to Parlay on ISDN PRI interface (Digium PRI card), so Parlay after some timeout disconnects call We guess that this setup should work, but we're not sure. Anyone with working setup like this? Anyone with experience of call progress getting from SIP to PRI or BRI interfaces ? Any advice or pointer to more info ? Thanks in advance, regards, Rob. ___ --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 phones
Depends on the type of headset. The Grandstream GXP-2000 and Liksys SPA-941 have headphone jacks on them, most phones are compatible with Plantronics univeral headsets. Kerry Garrison Publisher - GeekGazette.com - VOIPSpek.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Fraser Sent: Monday, November 28, 2005 8:27 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ip phones Hi all, Does anybody have any info on a decent quality sip hard phone that is headset compatible? Thank you 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 ___ --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] Upgrade Cisco 7910 with Asterisk ?
On Monday 28 Nov 2005 16:29, Tony Hoyle wrote: Noc Phibee wrote: Thanks sergio for your answer. But cisco france say me that i cant' bye SmartNet contract on this product. You can, but only in the US I believe. I've never found any deal less than £150 (UK). I guess that should depend as to whether it is hardware or software only. Only one solution are possible: Bye a special contract at $180.00 ... Pff i can bye a new equipment with this price hihihi It's not so bad... you do get access to firmware to all cisco devices with that, so if you have more than one device it becomes worth it. And it is also illegal. B ___ --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] Problem with ADIT 600 and FXO configuration
What does the TE406 leds indicate? On 11/28/05, William K. Volkman [EMAIL PROTECTED] wrote: I've looked through the archives of the mailing list for the last year and although informative I've not been successful at get this to work. We had a working Asterisk PBX system with 3 Digium X101P FXO lines and two TDM400P FXS cards. I've setup an ADIT 600 with an 8 port FXO card (and an 8 port FXS card not currently installed). We are going to be adding a T1 for incoming calls this week. I removed two of the X101P cards and installed a TE406P. I'm using Asterisk 1.0.9 (and matching zaptel, libpri) from tar files. /etc/zaptel.conf has this configuration: span=1,1,0,esf,b8zs,yellow span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs #Modular unit, first card is FXO fxsks=1-3 unused=4-8 #Modular unit, 1 FXS cards unused=9-16 unused=17-24 unused=25-48,49-72,73-96 fxsks=97 fxoks=98-101 fxoks=102-105 /etc/asterisk/zapata.conf has this: group = 0 signalling=fxs_ks context = incoming busydetect = yes overlapdial = no channel = 1-3 signalling=fxs_ks channel = 97 ;X100P group = 1 signalling = fxo_ks context = internal ;TDM400P callerid = Available 200 channel = 98-100 callerid = x channel = 101 ;TDM400P callerid = x channel = 102 callerid = x channel = 103 Parts of my adit configuration: -Setting slot a. set a:1 up set a:1 fdl none set a:1 lbo 4 set a:1 framing esf set a:1 id Inbound set a:1 linecode b8zs set a:1 loopdetect csu set a:1:1-24 side drop set a:1:1-24 type voice set a:1:1-24 signal ls set a:2 up set a:2 fdl none set a:2 lbo 1 set a:2 framing esf set a:2 id Outbound PBX set a:2 linecode b8zs set a:2 loopdetect csu set a:2:1-24 side drop set a:2:1-24 type voice set a:2:1-24 signal ls -Setting slot 1. set 1:1-8 signal lscpd set 1:1-8 txgain -3 set 1:1-8 rxgain -6 -Setting primary and secondary clock sources. set clock1 a:1 set clock2 internal -Setting the system idle pattern for DS0s. set idle 0xff -Making connections. connect a:2:1-3 1:1-3 Inbound calls just ring and ring (the leds on the ADIT change state) however asterisk doesn't respond. Attempts to make outgoing calls get: -- Executing Dial(SIP/202-ba07, Zap/g0/5551212) in new stack Nov 28 02:54:45 NOTICE[8627]: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Congestion(SIP/202-ba07, ) in new stack == Spawn extension (from-sip, 95942060, 3) exited non-zero on 'SIP/202-ba07' -- Executing Hangup(SIP/202-ba07, ) in new stack I've tried just about all combinations of gs/ls/ks for the signalling to no avail. Here is the output of status: status a:2:1-3 DS0 Rx AB Tx AB Signal T1 TP --- - - -- - -- a:2:1 01 01 LS TrafficN a:2:2 01 01 LS TrafficN a:2:3 01 01 LS TrafficN status 1:1-3 FXORx AB Tx AB Signal=T1 Sig T1 TP ---- - -- - -- 1:1 01 01 LSCPD = LS Traffic N 1:2 01 01 LSCPD = LS Traffic N 1:3 01 01 LSCPD = LS Traffic N show connect a:2:1-3 FromDescDescTo --- -- - - A:02:01LS VOICE DS0 -- FXOVOICE LSCPD 1:01 A:02:02LS VOICE DS0 -- FXOVOICE LSCPD 1:02 A:02:03LS VOICE DS0 -- FXOVOICE LSCPD 1:03 Can anyone spot what I've got wrong? Any suggestions or hints welcome. Thanks, William. ___ --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] Emailed voicemail messages not being deleted
According to the Asterisk wiki, adding the delete=yes option to a voicemail definition should automatically delete messages after they are emailed. This is the format that I'm using: 101 = ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes When I leave a message at mailbox 101, it gets emailed correctly but is not removed from the voicemail box. Am I missing something here, or is this feature broken? - .Dustin Wenz ___ --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] PROGRESS with cause code 31 received
I have been trying to work this problem out with my IAX provider. I dial a toll-free number, ex: 1-888-876-6262, and I get a due to technical difficulties message. I set my debug level to 9, and all I see when I dial out is this: -- Executing Dial(SIP/27-51de, IAX2/voctel/1766262||T) in new stack-- Called voctel/1766262-- Call accepted by 204.14.18.189 (format ulaw)-- Format for call is ulaw-- IAX2/voctel-3 is proceeding passing it to SIP/27-51de-- IAX2/voctel-3 is making progress passing it to SIP/27-51de-- Hungup 'IAX2/voctel-3' == Spawn extension (longdistance, 1766262, 1) exited non-zero on 'SIP/27-51de' What my IAX provider sees on the other end is this: -- Executing Dial(IAX2/tor-hub-13, Zap/G1/1766262||g) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/1766262 -- Zap/21-1 is proceeding passing it to IAX2/tor-hub-13 -- PROGRESS with cause code 31 received -- Zap/21-1 is making progress passing it to IAX2/tor-hub-13 -- Hungup 'Zap/21-1' I did a search through the mailing list and in the wiki. I found that cause code is used to report a normal event only when no other cause in thenormal class applies. and #defineAST_CAUSE_NORMAL_UNSPECIFIED 31. I am running Asterisk 1.2.0 and I am not sure what my provider is using, some version of HEAD is all I know. I am at a loss... I don't know the last time I tried to dial a toll-free from here, but it was working. Can anyone help steer me in the right direction? Thanks! Dana ___ --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] SNOM Phones MWI, Hold Retrieve buttons not working with Asterisk v1.2
We just upgraded to Asterisk 1.2 a few days ago. And now the Retrieve voicemail and hold buttons on our SNOM 360 phones are no longer working. When you put a caller from one of our zaptel lines on hold it hangs up on them immediately. Interestingly, if you put an internal extension on hold it does not drop the call. Additionally, the Retrieve voicemail butotn on the phones no longer work. The MWI (Message Waiting Indicator) lights up, but when you press the button you get Not Found sip:asterisk@ and busy signal. Any ideas on what might be wrong on how to fix it? Thanks! ___ --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: Problem connecting Two * servers with SIP (used to be: SIP Forward)
I apologize for the resend. I haven't received much feedback from this. I also noticed that what I'm getting is the caller id as the caller name and the sip peer name as the caller id number. Does anyone have any ideas/suggestions? Thanks, Waldo On Nov 26, 2005, at 2:52 AM, Waldo Rubinstein wrote: Hi guys, I'm trying to forward a call from one * server to another using SIP. Everything works when I use fromuser in the sip entry of the * forwarding the call. The problem is that when the receiving * sends the call to the UA, it puts the caller to be the value of fromuser instead of the caller-id (as documented). If I remove the fromuser, then the calls are denied because of wrong password on authentication for INVITE to '{calleridnum} ...'. Meaning, it's trying to authenticate the call based on caller-id instead of the peer name. How can I set it up to authenticate without necessarily using fromuser? Thanks, Waldo ___ --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 project converts to Subversion version control system
The Asterisk development team is pleased to announce that we have migrated our project repositories and development processes over to the Subversion version control system! Effective immediately, the primary source code distribution point for Asterisk, Zaptel and other related projects (other than release tarballs of course) will be http://svn.digium.com. The actual SVN repositories are available at http://svn.digium.com/svn, and there is a ViewCVS web viewer available at http://svn.digium.com/view. There is a separate repository for each major project, and each repository is organized in the typical Subversion fashion... for example, the Asterisk repository is organized as follows: http://svn.digium.com/svn/asterisk/trunk (was CVS HEAD) http://svn.digium.com/svn/asterisk/branches/1.2 (was CVS v1-2) http://svn.digium.com/svn/asterisk/tags/1.2.0 (was CVS v1-2-0) Other branches and tags are named similarly. Builds of Asterisk made from the new repositories will report a 'show version' tag made of the SVN branch name and the repository revision number that was checked out (unlike the CVS 'show version' tags which incorporated the date/time of checkout). The 'asterisk-cvs' mailing list has been renamed to 'svn-commits' and will continue to receive commit messages for the all the major projects on our SVN server (existing subscriptions are still in effect). In addition, there are new project specific commit mailing lists as well: asterisk-commits asterisk-addons-commits zaptel-commits libpri-commits libiax2-commits All of these lists are available on lists.digium.com. Additionally, the commit messages will contain 'X-SVN-Author' and 'X-SVN-Branch' mail headers to allow you to sort/filter the commit messages in any way you wish. One of the major benefits of this transition is that we will be opening up 'developer branches' for Asterisk Development Team members to be able to work on projects and make them available for public review, testing and participation; look for another announcement later this week when that process is ready. For the near future, we will continue to provide access to source code via CVS using the same servers/paths that you have previously been using; once every day, the relevant Subversion branches will be copied over into CVS and brought up to date. We expect to keep updating CVS HEAD this way for three to six months; the other branches will be maintained for six to nine months. However the CVS repositories will be updated in a single commit each day and will not contain any detailed revision history for the changes that are made. We encourage all users to transition to using Subversion for tracking development as soon as possible. (Special thanks to chipig, sussman, darix, jerenkrantz, eh, mbk and the others on #svn-dev who helped solve some sticky issues on Saturday evening of Thanksgiving weekend G) ___ --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] Problem with pulses dialing on asterisk 1.2
I'm using asteriskathome 2.0 beta 6 (asterisk 1.2) with SIP softphones, 7905G cisco SCCP and analog phone( DTMF dialing). All is working nice, however when I change DTMF for an analog pulse dialing,my analog phone is not working. I've found the following : http://www.voip-info.org/wiki-Asterisk+config+zapata.conf pulse=yes http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing pulsedial=yes What is correct : pulse=yes or pulsedial=yes ? However none are working ! I probably miss something. I need your help. Is there any french users with pulses dialing on analog phones ? Thanks. ___ --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] Pros and Cons of T1/E1 cards
I haven't heard of this product before so I did some searches on the Internet, this card is $5,400 for a single span T1 card? ouch! http://www.eiconworks.com/DivaServerV-PRI_T1%20.asp On 11/25/05, David Waugh [EMAIL PROTECTED] wrote: Hi John,I'm going to have to disagree with some previous posts.The Eicon Diva Server PRI/E1/T1 cards support an E1 interface and reduce the load of the call handling, echo cancellation etc as this is all processed on board on the card, and not on the central CPU of the computer. You can use the CAPI interface of the card combined with chan_capi_cm with the card.I have not found any problems when using different kernels or different versions of asterisk.I have one setup in our test lab here at Eicon with Asterisk so it does work! You can have up to 8 Diva Server cards in once machine - including a mixture of the analog and BRI cards.The Diva Server cards in two variants - the V-Series if you only want to use them with Voice based applications and the normal All-in-one cards if you want to do fax and RAS too. If you need any more information let me know, and I will assist furtherDavid-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of JohnDaragonSent: 25 November 2005 00:46To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Pros and Cons of T1/E1 cardsHi;We're looking to standardise on a single family of E1 PRI cards.I guess our options are :Digium / Zaptel / libpriSangoma/ Zaptel / Wanpipe AVM/ CAPIeIcon/ CAPIJunghanns/ BristuffCan anyone share any comparative experience of these, please ? Do theydiffer much in terms of interrupt requirement, CPU load c ? Any info gratefully received.jd--John Daragon[EMAIL PROTECTED]argv[0] limitedLambs Lawn Cottage,Staple Fitzpaine,Taunton,TA3 5SL,UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] AGI + CDR
Hi, I have an AGI script that is called after receving a call on a channel. And my script executel AGI cmd Dial to make another call. Is there any reason not to have CDR record for the call that was initiated in the AGI script? Or I am just missing something basics . Thanks, -- You don't have any choice, you already made it before you came here.___ --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] Problem with pulses dialing on asterisk 1.2
Cyrille DERORY wrote: I'm using asteriskathome 2.0 beta 6 (asterisk 1.2) with SIP softphones, 7905G cisco SCCP and analog phone( DTMF dialing). All is working nice, however when I change DTMF for an analog pulse dialing,my analog phone is not working. I've found the following : http://www.voip-info.org/wiki-Asterisk+config+zapata.conf pulse=yes http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing pulsedial=yes Pulsedial=yes works You should be aware that inserting a w into the dialstring does NOT work with pulsedial, however, so if your PSTN connection is a little slow, you may get misdials Remember you need to restart after making the change. To be really safe, reboot. John Novack What is correct : pulse=yes or pulsedial=yes ? However none are working ! I probably miss something. I need your help. Is there any french users with pulses dialing on analog phones ? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: Re: [Asterisk-Users] Zaptel errors on Debian
Hi On Mon, Nov 28, 2005 at 04:35:34PM -0800, Geo wrote: It should build wcfxo. Not trying anything special. I just follow the procedure ! When I reboot I have: ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded !!! HiSax: Linux Driver for passive ISDN cards HiSax: Version 3.5 (module) HiSax: Layer1 Revision 2.46.2.5 HiSax: Layer2 Revision 2.30.2.4 HiSax: TeiMgr Revision 2.20.2.3 HiSax: Layer3 Revision 2.22.2.3 I don't need ISDN, I do not configure any ISDN and I have no ISDN BRIstuffed or whatever Is ISDN susbsystem needed for using fxo devices using fxs signalling with Asterisk ? What's this ISDN driver doing here? A look at lspci will show you: Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface But by now you already knowthat this line represents your X100P card that hapens to have the same PCI ID as that TigerJet device. 'lspci -n' will show that the actual device has vendor ID e159 and product ID 1 . 'grep e159 /lib/modules/`uname -r`/modules.pcimap' will show that a number of zaptel modules look for devices with those vendor/product IDs but with some specific subvendor IDs and that the hisax driver tries to load them all. hotplug uses that information (extracted from the modules at depmod time) to load modules by bus IDs. Don't want it? blacklist it: echo hisax /etc/hotplug/blacklist.d/local Consider blacklisting other modules whose automatic modprobe seems unnecessary/pointless in just the same way (or $EDITOR /etc/hotplug/blacklist ) than Zapata Telephony Interface Registered on major 196 wcfxo: disagrees about version of symbol zt_receive wcfxo: Unknown symbol zt_receive wcfxo: disagrees about version of symbol zt_ec_chunk wcfxo: Unknown symbol zt_ec_chunk wcfxo: disagrees about version of symbol zt_transmit This beats me: version mipatch between zaptel and wcfxo ? One possible guess: you installed everything from one place. And then you compiled it again (without wcfxo this time) and reinstalled. Are you using m-a? .. Testing modprobe zaptel = OK zaptel driver but not wcfxo and ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Seems that wcfxo is not loaded. lsmod | grep zaptel Yet my config is OK. = Installind Asterisk make install compiling OK but errors on zap .. chan_zap.c:8935: error: dereferencing pointer to incomplete type chan_zap.c:8936: error: dereferencing pointer to incomplete type chan_zap.c:8950: error: dereferencing pointer to incomplete type .. On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote: Well, thanks, it might be great your package yet I would like to know how to adapt. I wouldn't like to rewrite Debian neither Asterisk but is somebody able to advice how you define modules in zconfig.h or whatever ? Any tip ? Geo Why would you need to define modules? The package builds wcfxo. What exactly do you try to do? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.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 -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Help connecting Avaya S8700 and Asterisk through H.323 trunk
Hi I'm trying to connect Avaya S8700 and Asterisk through H323 trunk (using channel oh323). I can make calls from S8700 H323 extension to Asterisk SIP phone using G711a codec but when I try to make a call from SIP phone to S8700 extension I listen one ringing tone and the call is dropped. Can anybody help me??? I see some errors in log but I can't resolve it: 1. -- H.323 call 'ip$localhost/29416-70919cfe' cleared, reason 8 (Transport failure) 2. 0:23.635 H225 Caller:8264258 H225Connect of H245 failed: Connection refused Full logs are attached below Thanks. --- oh323.conf: ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; ; Port to listen to. ; Default value is 1720. ; listenPort=1720 connectPort=1720 ; ; Configure the TCP port range to be used by H.323 ; tcpStart=1 tcpEnd=2 ; ; Configure the UDP port range to be used by H.323 ; Note: The port range used by RTP are configured from ; rtp.conf ; udpStart=1 udpEnd=2 ; ; Enable fast start (yes,no). ; fastStart=yes ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; Moreover, an integer (in decimal or hex format) may be entered. ; ipTos=none ; ; Set the maximum number of inbound/outbound/simultaneous ; H.323 connections. ; outboundMax=100 inboundMax=100 simultaneousMax=100 ; ; Call Rate Limiter params (ingress direction). When the total number ; of active calls is above 'crlThreshold' then the rate of the incoming ; H.323 calls is restricted in a way where no more than 'crlCallNumber' ; calls are allowed in 'crlCallTime' milliseconds, thus limiting the rate ; of incoming calls to: ; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec. ; ;crlCallNumber=20 ;crlCallTime=2 ;crlThreshold=30 ; ; Set the bandwidth limit for H.323 connections. ; The value is in Kbps. ; ;bandwidthLimit=1024 ; ; Set tracing options for the wrapper library and for the ; OpenH323 library. ; libTraceFile can be 'stdout' or a full path name to the tracefile. ; Only the trace info for OpenH323 is logged in libTraceFile. ; wrapLibTraceLevel=10 libTraceLevel=10 libTraceFile=/var/log/asterisk/h323.log ; ; Disable gatekeeper or specify a gatekeeper. The gatekeeper's ID is the zone name. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; gatekeeper's id@gatekeeper's name or address ; gatekeeper=DISABLE ; ; Set the gatekeeper password. If used, it enables H.235 access to gatekeeper. ; ;gatekeeperPassword=secret ; ; Set the gatekeeper registration timeout. Before the expiration of ; the timeout, a re-registration is attempted. ; gatekeeperTTL=600 ; ; Set the mode for sending user-input (DTMF) ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; INBAND - ; userInputMode=INBAND ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Default language ; language=en ; ; Default Music-On-Hold class ; musiconhold=default ; ; Set the default context of H.323 calls. ; context=voip-h323 ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ; ; Aliases/prefixes routed in all-aliases context. ; ; Aliases/prefixes routed in more-aliases context. ; ; Aliases/prefixes routed in all-prefixes context. ; ; Aliases/prefixes routed in more-stuff context. ; ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726- G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ;
[Asterisk-Users] Wrong usage of [] in the extension?
Hello, I am trying to set up my dialplan in such a manner that calls to numbers in the form 1-NPA-NXX- will only go through if the NPA dialed is a geographical NPA in the continental United States. I have collected a list of all NPAs that I want to allow, and have made the following dialplan entries (below). Calls that match the pattern do go through, but for each digit that I dial, I get the following error on the Asterisk console: WARNING[7045]: pbx.c:718 ast_extension_close: Wrong usage of [] in the extension or WARNING[7045]: pbx.c:699 ast_extension_match: Wrong usage of [] in the extension If I comment out the three entries below, the errors stop (and of course the calls do not go through). I have checked that each dialplan entry is on one line in the file; there are no accidental carriage returns or newlines present. Does anyone know what I am doing wrong? I realize that I have a rather large pattern defined, but it seems to be using correct syntax as far as I can tell. If there is a better way to do this, or if someone sees what I am doing incorrectly, could you please let me know? Thanks very much, Rusty Extensions.conf excerpt: exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2 24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2 81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3 23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4 09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4 84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5 51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6 09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6 61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7 20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7 86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8 45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9 09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9 51|952|954|956|970|971|972|973|978|979|980|985|989]NXX,1,Playback(local/ding ding) exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2 24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2 81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3 23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4 09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4 84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5 51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6 09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6 61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7 20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7 86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8 45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9 09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9 51|952|954|956|970|971|972|973|978|979|980|985|989]NXX,n,Dial(IAX2/[EMAIL PROTECTED] jet/${EXTEN},60) exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2 24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2 81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3 23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4 09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4 84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5 51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6 09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6 61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7 20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7 86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8 45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9 09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9 51|952|954|956|970|971|972|973|978|979|980|985|989]NXX,n,Goto(call-dispositi on|s|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] Emailed voicemail messages not being deleted
On 11/28/05, Dustin Wenz [EMAIL PROTECTED] wrote: According to the Asterisk wiki, adding the delete=yes option to a voicemail definition should automatically delete messages after they are emailed. This is the format that I'm using: 101 = ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes [snip] Try: 101 = ,First Last,[EMAIL PROTECTED],,attach=yes|delete=yes (notice the extra comma after the email address) I believe the setting that goes in between the empty commas is the pager email address Hope this helps. Cheers, Gonzalo ___ --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] cdr_manager.conf
Hello, While I was trying to get right CDR record from AGI script, I came across cdr_manager.conf I am trying to learn about cdr_manager.conf What is the purpose of cdr_manager.conf? How I can configure it? I did google, really didn't have very good luck. Would anybody please write couple of sentenses regarding this. Any link on documentaion, tutorial would be great help. Thanks, -- You don't have any choice, you already made it before you came here.___ --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] Help connecting Avaya S8700 and Asterisk through H.323 trunk
On 11/28/05, Pablo Chacón [EMAIL PROTECTED] wrote: Hi I'm trying to connect Avaya S8700 and Asterisk through H323 trunk (using channel oh323). I can make calls from S8700 H323 extension to Asterisk SIP phone using G711a codec but when I try to make a call from SIP phone to S8700 extension I listen one ringing tone and the call is dropped. Can anybody help me??? I've had greater success increasing the number of frames in an RTP packet when dealing with the med pro resources on the S8700. Also, make sure you're sending the call to the IP that is bound to the CLAN board that also has the signaling group you're trying to call into bound to it. With the connection refused here it seems like you might be trying to send the call to the IP of the med pro board instead of a CLAN board. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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] DTMF errors
I'm getting the following messages when a call is answered by a SIP device: Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP Transmission error to 192.168.1.254:19262: Operation not permitted For a Cisco 7940 line, I have the following sip.conf entry: [desk2] type=friend username=desk2 secret=xxx host=dynamic dtmfmode=rfc2833 context=international canreinvite=no callerid=xxx3034144980 [EMAIL PROTECTED] nat=yes qualify=yes accountcode=xxx disallow=all allow=ulaw allow=g729 The Asterisk system faces the Internet on a public IP. The phone is behind NAT. Asterisk version is 1.0.7. What can I do to fix this problem? Thanks, ___ --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] Upgrade Cisco 7910 with Asterisk ?
Bob Goddard wrote: It's not so bad... you do get access to firmware to all cisco devices with that, so if you have more than one device it becomes worth it. And it is also illegal. Not true - that's the *point* of the more expensive contracts. They cover you for each device that you own (there's probably a limit.. I only have 3 or 4 items though). I recently had to deal with cisco over a router I bought - they had no issue handling it under my current contract (in fact I still have one open, waiting for an RMA to complete). Tony ___ --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] Upgrade Cisco 7910 with Asterisk ?
Bob Goddard wrote: I guess that should depend as to whether it is hardware or software only. AFAIK all smartnet are software only... I've never heard of a hardware contract. (actually they're just an account on TAC which has access to certain parts of the website - there's no physical part to them at all). Tony ___ --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] Upgrade Cisco 7910 with Asterisk ?
On Monday 28 Nov 2005 20:41, Tony Hoyle wrote: Bob Goddard wrote: It's not so bad... you do get access to firmware to all cisco devices with that, so if you have more than one device it becomes worth it. And it is also illegal. Not true - that's the *point* of the more expensive contracts. They cover you for each device that you own (there's probably a limit.. I only have 3 or 4 items though). I recently had to deal with cisco over a router I bought - they had no issue handling it under my current contract (in fact I still have one open, waiting for an RMA to complete). You misunderstand. Buying a smartnet contract for a phone does not give you the right to download software for other hardware. One smartnet contract equals only one device covered. ___ --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] Upgrade Cisco 7910 with Asterisk ?
On Monday 28 Nov 2005 20:42, Tony Hoyle wrote: Bob Goddard wrote: I guess that should depend as to whether it is hardware or software only. AFAIK all smartnet are software only... I've never heard of a hardware contract. No, the vast majority of the smartnet contracts are hardware and software with some of the hardware having a 2 hour turnaround. (actually they're just an account on TAC which has access to certain parts of the website - there's no physical part to them at all). Yes, I know. ___ --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] Help connecting Avaya S8700 and Asterisk through H.323 trunk
Hi BJ Weschke, thanks but unfortunately Ip address is the correct one. Do you have S8700 with Asterisk working? using oh323 channel?? Maybe can help you my S8700 configuration... My S8700 configuration is: --- list ip-interfaces clan IP INTERFACES Num Skts Net ON Slot Code Sfx Node Name/ Subnet Mask Gateway Address Warn Rgn VLAN IP-Address -- --- --- --- --- --- . y 04A04 TN799 D CLND04A04 255.255.255.0 10.64.108.254 400 2 n 10.64.108.132 --- change signaling-group 23 Page 1 of 5 SIGNALING GROUP Group Number: 23 Group Type: h.323 Remote Office? n Max number of NCA TSC: 0 SBS? n Max number of CA TSC: 0 IP Video? nTrunk Group for NCA TSC: Trunk Group for Channel Selection: 23 Supplementary Service Protocol: a Network Call Transfer? n T303 Timer(sec): 10 Near-end Node Name: CLND04A04 Far-end Node Name: ASTERISK Near-end Listen Port: 1720Far-end Listen Port: 1720 Far-end Network Region: 2 LRQ Required? n Calls Share IP Signaling Connection? n RRQ Required? n Media Encryption? n Bypass If IP Threshold Exceeded? n DTMF over IP: out-of-bandDirect IP-IP Audio Connections? n IP Audio Hairpinning? n Interworking Message: PROGress DCP/Analog Bearer Capability: 3.1kHz - display trunk-group 23 Page 1 of 19 TRUNK GROUP Group Number: 23 Group Type: isdn CDR Reports: y Group Name: ASTERISK-H323 COR: 1TN: 1TAC: #23 Direction: two-wayOutgoing Display? n Carrier Medium: IP Dial Access? yBusy Threshold: 255 Night Service: Queue Length: 0 Service Type: tie Auth Code? nTestCall ITC: rest Far End Test Line No: TestCall BCC: 4 TRUNK PARAMETERS Codeset to Send Display: 0 Codeset to Send National IEs: 6 Max Message Size to Send: 260 Charge Advice: none Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc Trunk Hunt: cyclical QSIG Value-Added? n Digital Loss Group: 18 Incoming Calling Number - Delete: Insert: Format: Bit Rate: 1200 Synchronization: asyncDuplex: full Disconnect Supervision - In? y Out? n Answer Supervision Timeout: 0 display trunk-group 23 Page 2 of 19 TRUNK FEATURES ACA Assignment? nMeasured: none Wideband Support? n Internal Alert? nMaintenance Tests? y Data Restriction? n NCA-TSC Trunk Member: Send Name: y Send Calling Number: y Used for DCS? n Suppress # Outpulsing? nFormat: public Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider Replace Restricted Numbers? n Replace Unavailable Numbers? n Send Connected Number: n Network Call Redirection: noneHold/Unhold Notifications? n Send UUI IE? yModify Tandem Calling Number? n Send UCID? n Send Codeset 6/7 LAI IE? y SBS? n Network (Japan) Needs Connect Before Disconnect? n display ip-network-region 2 Page 1 of 19 IP NETWORK REGION Region: 2 Location: 1 Authoritative Domain: Name: ** Pool LR VoIP ** Intra-region IP-IP Direct Audio: yes MEDIA
[Asterisk-Users] Avaya 4620SW Invalid Subscription-State - Issue
Hello All, I'm using an Avaya 4620SW with Asterisk, the phone when hooked up to the network, works for sometime, (I have not actually monitored the time) maybe 20-30 minutes, after which the phone will still have a dial tone, but can't dial out or recieve calls. I scanned thru the logs and found this: Got SIP response 400 Invalid Subscription-State back from 192.168.202.200 ( 192.168.202.200 is the static IP assigned to the Avaya device). Thanks ___ --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: Wrong usage of [] in the extension?
Sorry to reply to myself, but I need to add some information: I have been informed (and now understand why) that the [] syntax does not do what I had in mind here. Is there any syntax that will do it? If not, I will just create a separate pattern for each NPA, which is not a big deal, but I am now curious as to whether there is such a syntax. Thanks, RustyOn 11/28/05, Rusty Dekema [EMAIL PROTECTED] wrote: Hello, I am trying to set up my dialplan in such a manner that calls to numbers in the form 1-NPA-NXX- will only go through if the NPA dialed is a geographical NPA in the continental United States. I have collected a list of all NPAs that I want to allow, and have made the following dialplan entries (below). Calls that match the pattern do go through, but for each digit that I dial, I get the following error on the Asterisk console: WARNING[7045]: pbx.c:718 ast_extension_close: Wrong usage of [] in the extension or WARNING[7045]: pbx.c:699 ast_extension_match: Wrong usage of [] in the extension If I comment out the three entries below, the errors stop (and of course the calls do not go through). I have checked that each dialplan entry is on one line in the file; there are no accidental carriage returns or newlines present. Does anyone know what I am doing wrong? I realize that I have a rather large pattern defined, but it seems to be using correct syntax as far as I can tell. If there is a better way to do this, or if someone sees what I am doing incorrectly, could you please let me know? Thanks very much, Rusty Extensions.conf excerpt: exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2 24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2 81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3 23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4 09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4 84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5 51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6 09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6 61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7 20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7 86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8 45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9 09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9 51|952|954|956|970|971|972|973|978|979|980|985|989]NXX,1,Playback(local/ding ding) exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2 24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2 81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3 23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4 09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4 84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5 51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6 09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6 61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7 20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7 86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8 45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9 09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9 51|952|954|956|970|971|972|973|978|979|980|985|989]NXX,n,Dial(IAX2/[EMAIL PROTECTED] jet/${EXTEN},60) exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2 24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2 81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3 23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4 09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4 84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5 51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6 09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6 61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7 20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7 86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8 45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9 09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9
Re: [Asterisk-Users] beginner questions
Hello Amir On Sun, 2005-11-27 at 20:31 -0800, Amir Aziz wrote: Dear List Members, I am trying to setup a small asterisk box. My configure is pretty basic for now. my zaptel.conf is as follows [ ... ] 6. What other books/links can be helpful in learning this interesting software. I liked Switching to VoIP from oreilly. Very helpful and numerous references to *. I thank you all for you help in advance. Regards, Amir Aziz __ 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 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?
I guess that should depend as to whether it is hardware or software only. AFAIK all smartnet are software only... I've never heard of a hardware contract. Smartnet comes in serveral different flavors (eg, 24x7, 8x5) and all of the flavors cover the hardware in addition to the software (at least in the US). Been using smartnet since around 1985. Cisco actually stockpiles replacement hardware in key locations around the US based on what customers have signed up for, and those key locations are _not_ necessarily cisco sites. Most are contractual agreements with selected carriers, and the carrier actually dispactches the replacement hardware based on TAC orders. That's why they can sell 24x7 smartnet contracts with 4 hour response times. The local cisco sales office is not allowed to stockpile anything and, for the most part, will refer you to the TAC folks for failures, assistance, etc. Resellers frequently operate a little differently. The smartnet contracts for the 79x0 phones includes replacement hardware should the phone fail for any reason. (actually they're just an account on TAC which has access to certain parts of the website - there's no physical part to them at all). If you read the legal stuff that must be acknowledged when downloading software (under any cisco contract), you can only legally download the software for the stuff that you have under contract. But, I've never heard them enforce the web acknowledgement with any company to date. ___ --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] misdn, busy detection
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 hi, very often, when the caller hangs up the phone, the isdn phone rings without stopping. It seems, that asterisk does noch check, that the caller has hang up. I have this problem between ISDN-ISDN and ISDN-SIP. Is there a solution for misdn? cu denny -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDi3V4Klzhkqt9P+ARAmnrAJ482h1lr0BSKpZgc5IbOQPiXV2jNgCZAWAR fm7V58hiK8dZhtj+Y77yNTo= =RC3h -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] SNOM Phones MWI, Hold Retrieve buttons not working with Asterisk v1.2
On 13:48, Mon 28 Nov 05, Sascha Deri wrote: Additionally, the Retrieve voicemail butotn on the phones no longer work. The MWI (Message Waiting Indicator) lights up, but when you press the button you get Not Found sip:asterisk@ and busy signal. I have been fighting with the same thing for hours. The old asterisk sent MWI messages as [EMAIL PROTECTED] The new code sends this as [EMAIL PROTECTED] This can be overwritten by some setting in sip.conf, but I didn't look into that. I simply setup an extension: exten = asterisk,1,VoicemailMain(${CALLERIDNUM}) That did the trick for me. Hope this helps you. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM and 1.0.9
Greetings to all, I am trying to get the line lights on a SNOM 320 to work using 'hint' in extensions.conf. Unfortunately I have not been able to get it to work properly. Does anyone know for sure if the hint function works properly in 1.0.9? If anyone has gotten this to work properly under 1.0.9 please post a sample. I am using: Exten = xxx,hint,sip/xxx Extension no. and sip channel are the same. On the SNOM I have Destination chosen, and the extension on asterisk. Maybe output the output from the command 'sip show subscriptions' would also be nice. Regards, Joe ___ --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] Newbie requesting help!
Hi. Things are the same. I would be glad if you could help out. 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
[Asterisk-Users] Interface Cards that support QSIG
Anyone know of a board, Digium, Sangoma or other, that supports QSIG? Only hardware that I have seen that supports QSIG are Vegastream gateways. Thanks in Advance! -- Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory ___ --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] Problem with pulses dialing on asterisk 1.2
On Nov 28, 2005, at 3:00 PM, John Novack wrote: Cyrille DERORY wrote: I'm using asteriskathome 2.0 beta 6 (asterisk 1.2) with SIP softphones, 7905G cisco SCCP and analog phone( DTMF dialing). All is working nice, however when I change DTMF for an analog pulse dialing,my analog phone is not working. I've found the following : http://www.voip-info.org/wiki-Asterisk+config+zapata.conf pulse=yes http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse +dialing pulsedial=yes Pulsedial=yes works You should be aware that inserting a w into the dialstring does NOT work with pulsedial, however, so if your PSTN connection is a little slow, you may get misdials Remember you need to restart after making the change. To be really safe, reboot. I think this might be a problem peculiar to Asterisk @ home, since I cannot get my install to accept pulsedial=yes, either. Even though I have tried specifying pulsedial=yes in zapata.conf before the channel= line and also in zapata-auto.conf, as well as in zapata_additional.conf (along with the extension's config), none of the above work. (I figured I'd try everywhere, even if it didn't make any sense...) Anyhow, if I connect to the Asterisk console and run zap show channel 1, it reports Pulse phone: no even though pulsedial=yes is specified. I don't know how [EMAIL PROTECTED]/AMP would have modified zaptel to break this, but it is indeed possible that this is an [EMAIL PROTECTED] specific problem. Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM and 1.0.9
From memory (at a previous installation) you will need a newer version of Asterisk than 1.09 for the lights to work. PaulH - Original Message - From: Joseph Rothstein [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, November 29, 2005 8:32 AM Subject: [Asterisk-Users] SNOM and 1.0.9 Greetings to all, I am trying to get the line lights on a SNOM 320 to work using 'hint' in extensions.conf. Unfortunately I have not been able to get it to work properly. Does anyone know for sure if the hint function works properly in 1.0.9? If anyone has gotten this to work properly under 1.0.9 please post a sample. I am using: Exten = xxx,hint,sip/xxx Extension no. and sip channel are the same. On the SNOM I have Destination chosen, and the extension on asterisk. Maybe output the output from the command 'sip show subscriptions' would also be nice. Regards, Joe ___ --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] Upgrade Cisco 7910 with Asterisk ?
Bob Goddard wrote: You misunderstand. Buying a smartnet contract for a phone does not give you the right to download software for other hardware. One smartnet contract equals only one device covered. News to me... If cisco queried my TAC request of a router on an existing contract bought when I only owned a phone I'd believe it, but they didn't bat an eyelid. I see that smartnet is now available from my usual hardware dealer for £40 which is pretty damned competetive (since I paid £150 for it 5 months ago from an 'official' cisco dealer, who then proceeded to mess me about for 2 months before getting off their arses and actually registering it)... not sure I'd actually get one again though - the dealer support has been very good, it's just cisco's end that has sucked. Tony ___ --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] Help connecting Avaya S8700 and Asterisk through H.323 trunk
On 11/28/05, Pablo Chacón [EMAIL PROTECTED] wrote: Hi BJ Weschke, thanks but unfortunately Ip address is the correct one. Do you have S8700 with Asterisk working? using oh323 channel?? Maybe can help you my S8700 configuration... My S8700 configuration is: --- Yes. There's one client I setup where they have this in production. I don't have access via DSA anymore though to tell you what the sig group settings are there. Sorry. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Emailed voicemail messages not being deleted
That appears to have done the trick...I guess I expected some sort of warning at the console if I had inadvertently malformed the parameter string. It works now though, so it's all good. Thanks for the help! - .Dustin Wenz On Nov 28, 2005, at 2:15 PM, Gonzalo Servat wrote: On 11/28/05, Dustin Wenz [EMAIL PROTECTED] wrote: According to the Asterisk wiki, adding the delete=yes option to a voicemail definition should automatically delete messages after they are emailed. This is the format that I'm using: 101 = ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes [snip] Try: 101 = ,First Last,[EMAIL PROTECTED],,attach=yes|delete=yes (notice the extra comma after the email address) I believe the setting that goes in between the empty commas is the pager email address Hope this helps. Cheers, Gonzalo ___ --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] SNOM and 1.0.9
Joseph Rothstein wrote: Greetings to all, I am trying to get the line lights on a SNOM 320 to work using 'hint' in extensions.conf. Unfortunately I have not been able to get it to work properly. Does anyone know for sure if the hint function works properly in 1.0.9? If anyone has gotten this to work properly under 1.0.9 please post a sample. I am using: Exten = xxx,hint,sip/xxx Extension no. and sip channel are the same. On the SNOM I have Destination chosen, and the extension on asterisk. Maybe output the output from the command 'sip show subscriptions' would also be nice. Regards, Joe I can't tell you if it works in 1.0.9 as I've only used hints in 1.2 iterations (beta, rc, final), but, my hints have the channel type in upper case: exten = 509,hint,SIP/509. Don't know if your extract is direct from your dial plan or just the way you typed it in the email. If it is lower case in your dial plan try changing it. In 1.2 there is a CLI command 'show hints', but I don't believe that exists pre 1.2. Cheers, Kevin -- Optimacy Communications, LLC http://www.optimacycomm.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] Upgrade Cisco 7910 with Asterisk ?
Rich Adamson wrote: If you read the legal stuff that must be acknowledged when downloading software (under any cisco contract), you can only legally download the software for the stuff that you have under contract. But, I've never heard them enforce the web acknowledgement with any company to date. I've got a TAC case open at the moment on an IOS bug, waiting for more info from me (when I get the router back.. been a while now). They've actively encouraged me to download different versions of IOS - and the TAC is under my original contract (which I note is now available from my favourite hardware dealer for £40... I paid over £150 for it from a 'real' cisco dealer.). Tony ___ --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] Comedian Voicemail? PROBLEMS?
Hi, I am a newbie, and I am setting up a simple system to share a PSTN line with another location. In the process of setting this up I am also testing the various codecs. I am only able to get comedian voicemail (ie dialing 1234) to record or playback messages if I use the GSM codec? Is this normal and expected? If I use ulaw or alaw I get either trash noise or an immediate busy signal on attempted message playback. I am running asterisk 1.2 on OSX 10.4.3. Thanks for any help or ideas, and sorry if this is a tired old question. Marty ___ --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] Problem with Internet connection
Hello. I`m using asterisk 1.0.9 and it`s working fine until I disconect the WAN interface. Then asterisk doesn`t work fine, doesn`t make any Dial() and I don`t know where is the problem. When I connect the WAN interface all start working fine. I`m also using NAT in the same server. I don`t know what asterisk is looking for on the internet. Regards. -- José Luis Gómez Qualis Information Technology Av. Rivadavia 2553, PB Of. 43 EP 0342-4565684 int 102 Bs. As. 011-51990896 www.qualis.com.ar Soporte 0810-8880022 Santa Fe - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM and 1.0.9
On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote: From memory (at a previous installation) you will need a newer version of Asterisk than 1.09 for the lights to work. on 1.0.9 the lights work. In this way: person is on the phone: light is on Person is not on the phone: light is off since 1.2 the lights will blink when the phone is running and above states work the same. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM and 1.0.9
On 11/28/05, Kevin Hanson [EMAIL PROTECTED] wrote: Joseph Rothstein wrote: Greetings to all, I am trying to get the line lights on a SNOM 320 to work using 'hint' in extensions.conf. Unfortunately I have not been able to get it to work properly. Does anyone know for sure if the hint function works properly in 1.0.9? If anyone has gotten this to work properly under 1.0.9 please post a sample. This is definitely a 1.2 only feature. It is not in 1.0.9. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] small office setup
Hi Jason, There are a couple of boxes on the market these days that have the following ports: FXO/ISDN line out to PSTN 2 - FXS - analogue phone (or fax) WAN port for DSL As well as wfifi. Fritz WLAN FON box for example. Quick design: Asterisk server at HQ. Each remote employee has one of these boxes. Calls in go to Asterisk, and get forwarded via VOIP to a phone registered via SIP from one of the FRITZ boxes. If there is no SIP registration, the call could be redirected back out to the PSTN. Calls from an employee could go out either Asterisk or the PSTN based on dialplan. You would need either two FXO ports on the Asterisk (or one ISDN port). All the rest, voicemail, recording, time of day changes, etc. can all be done in Asterisk. Good luck, Joe Joseph Rothstein, CCIE Senior Network Engineer Comcentrixs GmbH Landsberger Str. 155 / House 2 D-80687 Munich Office: +49 (0) 89 2444 3168 -212 Softphone: +49 (0) 89 2444 3168 - 8451 Mobile: +49 (0) 172 104 3273 Fax: +49 (0) 89 2444 3168 -999 E-mail: [EMAIL PROTECTED] Web: http://www.comcentrixs.com PLEASE NOTE: Please inform us immediately if this e-mail and/or any attachment was transmitted incompletely or was not intelligible. This e-mail and any attachment is for authorized use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. ___ --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