Daniel Hikel
2006-Oct-05 07:11 UTC
[asterisk-users] AW: asterisk-users Digest, Vol 27, Issue 23
unsubscribe -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von asterisk-users-request@lists.digium.com Gesendet: Donnerstag, 5. Oktober 2006 16:08 An: asterisk-users@lists.digium.com Betreff: asterisk-users Digest, Vol 27, Issue 23 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-request@lists.digium.com You can reach the person managing the list at asterisk-users-owner@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. GXP - 2000 BLF (Andrew Shelton) 2. Re: AEL2 #include madness in Asterisk 1.4 - Murf? (Steve Murphy) 3. Re: Bandwidth requirements (Benny Amorsen) 4. RE: Extremely choppy sound on some of our POTSnetwork calls; goes away with mute (sdgesa gaeharth) 5. Re: Re: Bandwidth requirements (J. Oquendo) 6. Re: snom 360: how to make record button working ? (Joe Pukepail) 7. Re: TNT Max Password reset (James) 8. two asterisk and one NBX system (jose diaz) 9. Re: Video Conference (Noah Miller) 10. RE: TNT Max Password reset (asterisk) 11. Re: TNT Max Password reset (Don) 12. Re: Re: extensions.conf strangeness (Michael Neuhauser) ---------------------------------------------------------------------- Message: 1 Date: Thu, 5 Oct 2006 14:00:41 +0100 From: "Andrew Shelton" <andrew.shelton@stemnetworks.co.uk> Subject: [asterisk-users] GXP - 2000 BLF To: <asterisk-users@lists.digium.com> Message-ID: <792989DB4816704C830932E501F2E79206BCCC@corp-svr-1.corporate.local> Content-Type: text/plain; charset="us-ascii" Hello, I have been trying to get my Grandstream busy line filter to work for ages.. All the lights flash as they are supposed to. If one Grandstream 7000 calls another Grandstream 7003 I can use Grandstream 7002 to pick the call up pressing the BLF button and all works fine. However if I call Grandstream 7000 with a mobile phone and try to pickup the call with Grandstream 7002 all I get is a 603 error on Grandstream 7002. I am using firmware 1.1.12 for the Grandstream and 1.2.12.1 version of asterisk This is the error I get from my log.. if some one could please help Oct 5 12:12:51 DEBUG[7723] chan_sip.c: build_route: Contact hop: <sip:7003@192.168.1.94:5060> Oct 5 12:12:51 VERBOSE[8828] logger.c: -- Executing NoOp("SIP/7003-b721be28", "**7002") in new stack Oct 5 12:12:51 VERBOSE[8828] logger.c: -- Executing Pickup("SIP/7003-b721be28", "7002") in new stack Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No originating channel found. Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No call pickup possible... Oct 5 12:12:51 VERBOSE[8828] logger.c: == Spawn extension (inbound-from-stem, **7002, 2) exited non-zero on 'SIP/7003-b721be28' Oct 5 12:12:51 DEBUG[7716] channel.c: Avoiding initial deadlock for 'SIP/7003-b721be28' SIP [7000] type=friend context=inbound-from-stem Subscribecontext=BLF secret=* host=dynamic canreinvite=no callgroup=2 pickupgroup=2 mailbox=7000@default username=7000 dtmfmode=rfc2833 callerid="STEM" <17524543545> qualify=yes EXTENSIONS [default] include => stem include => to-siemens include => BLF include => BLF_group_pickup [stem] ;exten STEM GROUP = 01752 692205 exten => 123454,1,Ringing exten => 123454,n,Wait(1) exten => 123454,n,Answer() exten => 123454,n,NoOp(${CALLERID(all)}) exten => 123454,n,SetCIDName(Outside Caller) exten => 123454,n,Set(CALLERID(number)=9${CALLERIDNUM}) exten => 123454,n,NoOp(${CALLERID(all)}) exten => 123454,n,Macro(stdexten2,7003,${STEMGROUP},20) ;exten 7000 = 01752 692204 exten => 123455,1,Ringing exten => 123455,n,Wait(1) exten => 123455,n,Answer() exten => 123455,n,NoOp(${CALLERID(all)}) exten => 123455,n,SetCIDName(Outside Caller) exten => 123455,n,Set(CALLERID(number)=9${CALLERIDNUM}) exten => 123455,n,NoOp(${CALLERID(all)}) exten => 123455,n,Macro(stdexten2,7000,${stem},20) ;exten 7001 = 01752 692283 exten => 123456,1,Ringing exten => 123456,n,Wait(1) exten => 123456,n,Answer() exten => 123456,n,NoOp(${CALLERID(all)}) exten => 123456,n,SetCIDName(Outside Caller) exten => 123456,n,Set(CALLERID(number)=9${CALLERIDNUM}) exten => 123456,n,NoOp(${CALLERID(all)}) exten => 123456,n,Macro(stdexten2,7001,${stem1},20) [internal] ;Internal Extensions exten => _7XXX,1,Ringing exten => _7XXX,n,Wait(1) exten => _7XXX,n,Answer() exten => _7XXX,n,Set(FOO1=${CHANNEL:4}) exten => _7XXX,n,Set(FOO2=${CUT(FOO1,-,1)}) exten => _7XXX,n,Set(CALLERID(number)=${FOO2}) exten => _7XXX,n,Macro(stdexten,${EXTEN},SIP/${EXTEN}) [inbound-from-pstn] ; inbound calls to this context from outside lines include => default [inbound-from-sip] include => default [inbound-from-local] ;from sip default context used.. requires hints include => voicemail include => provider include => outbound ;include => stem ;for hints [inbound-from-stem] include => BLF include => internal include => DefExt include => voicemail include => outbound include => BLF_group_pickup include => feature-cfu include => feature-cfna include => feature-cfb [inbound-from-logicall] include => internal include => DefExt include => voicemail include => outbound include => BLF_group_pickup include => feature-cfu include => feature-cfna include => feature-cfb ;Test section for BLF on Grandstreams for Stem [BLF_group_pickup] include =>inbound-from-stem exten => _**.,1,NoOp(${EXTEN}) exten => _**.,2,Pickup(${EXTEN:2}) exten => _**.,3,Hangup [BLF] include =>inbound-from-stem exten =>7000,hint,SIP/7000 exten =>7000,1,Dial(SIP/7000,20,r) exten =>7001,hint,SIP/7001 exten =>7001,1,Dial(SIP/7001,20,r) exten =>7002,hint,SIP/7002 exten =>7002,1,Dial(SIP/7002,20,r) exten =>7003,hint,SIP/7003 exten =>7003,1,Dial(SIP/7003,20,r) exten =>7004,hint,SIP/7004 exten =>7004,1,Dial(SIP/7004,20,r) exten =>7005,hint,SIP/7005 exten =>7005,1,Dial(SIP/7005,20,r) exten =>7006,hint,SIP/7006 exten =>7006,1,Dial(SIP/7006,20,r) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061005/f4499d b5/attachment-0001.htm ------------------------------ Message: 2 Date: Thu, 05 Oct 2006 07:10:00 -0600 From: Steve Murphy <murf@digium.com> Subject: [asterisk-users] Re: AEL2 #include madness in Asterisk 1.4 - Murf? To: asterisk-users@lists.digium.com Message-ID: <1160053800.3638.88.camel@monster> Content-Type: text/plain; charset="us-ascii" On Thu, 2006-10-05 at 01:08 -0700, dgarstang@oneeighty.com wrote:> Asterisk 1.4 beta2. > > My top level /etc/asterisk/extensions.ael has the following > two lines: > > #include "include/syst/extensions.ael" > #include "include/btck/extensions.ael" > > Here is the console output on Asterisk load. > > app_system.so => (Generic System() application) > [Oct 4 15:48:15] NOTICE[1143]: pbx_ael.c:3798 > pbx_load_module: Starting AEL load process. > [Oct 4 15:48:15] NOTICE[1143]: pbx_ael.c:3805 > pbx_load_module: AEL load process: calculated config file name > '/etc/asterisk/extensions.ael'. > [Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex: > --Read in included > file /etc/asterisk/include/syst/extensions.ael, 4130 chars > [Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex: > --Read in included file /etc/asterisk/include/syst/macros.ael, > 1463 chars > [Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex: > --Read in included > file /etc/asterisk/include/syst/dundiapps.ael, 758 chars > [Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex: > --Read in included file /etc/asterisk/include/syst/rdapps.ael, > 275 chars > [Oct 4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex: > --Read in included > file /etc/asterisk/include/btck/extensions.ael, 1385 chars > [Oct 4 15:48:15] NOTICE[1143]: pbx_ael.c:3808 > pbx_load_module: AEL load process: parsed config file name > '/etc/asterisk/extensions.ael'. > [Oct 4 15:48:15] ERROR[1143]: pbx_ael.c:1162 check_goto: > Error: file /etc/asterisk/include/syst/extensions.ael, line > 157-157: goto: no label remote exists in the current > extension! > [Oct 4 15:48:15] ERROR[1143]: pbx_ael.c:1162 check_goto: > Error: file /etc/asterisk/include/syst/extensions.ael, line > 159-159: goto: no label local exists in the current > extension! > [Oct 4 15:48:15] ERROR[1143]: pbx_ael.c:3821 pbx_load_module: > Sorry, but 0 syntax errors and 2 semantic errors were > detected. It doesn't make sense to compile. > pbx_ael.so => (Asterisk Extension Language Compiler) > > Here's the context > from /etc/asterisk/include/syst/extensions.ael, that contains > lines 157 that the parser is complaining about: > > 148 context syst_Route { > 149 > 150 _[*0123456789]. => { > 151 NoOp(*** Originated call ${CALLERID} -> > ${EXTEN}); > 152 Set(TMP=${CALLERID(number)}); > 153 &SysLogger(This is a test message); > 154 &FastAGIConnectGet(CALLERID); > 155 ChanIsAvail(SIP/${EXTEN}); > 156 if ("${AVAILCHAN}" = "") { > 157 goto remote; > 158 } else { > 159 goto local; > 160 } > 161 remote: > 162 NoOp(REMOTE); > 163 Set(PATH> ${DUNDILOOKUP(3254103,DUNDIRegistr)}); > 164 //Set(PATH> ${DUNDILOOKUP(${EXTEN},DUNDIRegistr)}); > 165 Dial(${PATH}); > 166 Hangup(); > 167 local: > 168 NoOp(LOCAL); > 169 Dial(SIP/${EXTEN}); > 170 Hangup(); > 171 > 172 } > 173 } > > As you can quite clearly see, labels 'remote' and 'local' DO > exist in the syst_Route context. > > Now, if I switcheroo the two includes around in the top > level /etc/asterisk/extensions.ael, to: > > #include "include/btck/extensions.ael" > #include "include/syst/extensions.ael" > > and reload Asterisk, I get: > > [Oct 4 15:57:28] NOTICE[1202]: pbx_ael.c:3813 > pbx_load_module: AEL load process: compiled config file name > '/etc/asterisk/extensions.ael'. > [Oct 4 15:57:28] NOTICE[1202]: pbx_ael.c:3816 > pbx_load_module: AEL load process: merged config file name > '/etc/asterisk/extensions.ael'. > [Oct 4 15:57:28] WARNING[1202]: pbx.c:6194 > ast_context_verify_includes: Context 'syst_PSTNStart' tries > includes nonexistent context 'syst_AppACDQueue' > [Oct 4 15:57:28] WARNING[1202]: pbx.c:6194 > ast_context_verify_includes: Context 'btck_CallStart' tries > includes nonexistent context 'syst_ACD' > [Oct 4 15:57:28] NOTICE[1202]: pbx_ael.c:3819 > pbx_load_module: AEL load process: verified config file name > '/etc/asterisk/extensions.ael'. > pbx_ael.so => (Asterisk Extension Language Compiler) > > There are no errors about nonexistent labels in the syst_Route > extension. I would not have thought that #include order made > any difference, since all we are doing is pulling a bunch of > contexts into a global context space. > > Anyone? Mr Murpy, care to take a shot at it? :) > > Doug.Doug-- I cannot reproduce the problems, given just the one context. There is something magical about your data, that the code trips over it, and to find the bugs, I will need your files! Is this possible? As to order, you are correct, it should not make a difference what order the files are included in the data. I did note that in the above output, you got the error messages: [Oct 4 15:57:28] WARNING[1202]: pbx.c:6194 ast_context_verify_includes: Context 'syst_PSTNStart' tries includes nonexistent context 'syst_AppACDQueue' [Oct 4 15:57:28] WARNING[1202]: pbx.c:6194 ast_context_verify_includes: Context 'btck_CallStart' tries includes nonexistent context 'syst_ACD' These messages do not come from the AEL compiler, but rather, are complaints from the bowels of the asterisk engine: somewhere, it's not finding some included contexts... which may mean yet one more bug in the AEL code: why didn't AEL make note of it first? murf -- Steve Murphy Software Developer Digium -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3227 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20061005/e682b5 e6/smime-0001.bin ------------------------------ Message: 3 Date: 05 Oct 2006 15:31:54 +0200 From: Benny Amorsen <benny+usenet@amorsen.dk> Subject: [asterisk-users] Re: Bandwidth requirements To: asterisk-users@lists.digium.com Message-ID: <m3lknuubw5.fsf@ursa.amorsen.dk> Content-Type: text/plain; charset=us-ascii>>>>> "rJ" == raphael Jacquot <sxpert@sxpert.org> writes:rJ> ATM cell tax is actually 10% as there's 5 header bytes for each 53 rJ> bytes cell, For VoIP the cell tax is much larger. In the example, each RTP packet contains 20 useful bytes and 40 bytes IP overhead. 60 bytes doesn't fit in one cell, so you end up with 106 bytes at the ATM layer to transport 20 bytes of G.729. The ATM-caused overhead is thus 46 bytes per voice packet, thereby making the needed bandwidth 77% larger. All in all VoIP over ADSL adds 430% overhead, when using G.729 and 20ms packets. Lovely, isn't it? /Benny ------------------------------ Message: 4 Date: Thu, 5 Oct 2006 06:38:10 -0700 (PDT) From: sdgesa gaeharth <pollux1234567890@yahoo.com> Subject: RE: [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute To: asterisk-users@lists.digium.com Message-ID: <20061005133811.67985.qmail@web50808.mail.yahoo.com> Content-Type: text/plain; charset="iso-8859-1" Below is the text of my original post. I am not sure what Codec we are using. The "Codec Preferences" phone setting shows, in order of preference, G.711u, G.711A, G.729AB We are running asterisk-1.2.4 with zaptel-1.2.7 on Fedora Core 4-2.6.14-1.1656_FC4smp. It is installed on a Dell PE 2500 with 2x900 MHz processors and 1 Gb RAM and 1 SCSI Disk. The server has a Digium TDM400P card which is connected to 4 POTS lines. The server is also connected to a 100MB switched LAN where we have about 20 Polycom 501 phones with the latest firmware updates. Nothing else runs on the server except an ftp daemon which is never used except when a phone reboots. For about 20% of the calls to the outside world, the voice on the other end of an outside line is incredibly choppy. Enough to where we have to hang up and call on a cell phone. It is always the same numbers that are choppy. The funny thing is, if I press mute while talking on a choppy call, the choppiness goes away completely. I have tried: turning off ACPI, turning off APCI, moving the card to another PCI slot, changing the RX/TX gains. There are no shared IRQs. I have tested the lines by unplugging them from the asterisk server and plugging them directly into an analogue phone. Using "cat /proc/interrupts; sleep 10 ; cat /proc/interrupts" I see that there are about 1,000 interrupts per seconds between the card and the CPU. I do not think it is a network congestion problem as intra-office communications as well as voicemail retrieval are always perfect. The Voip does not go over any routers, just a max of 2 switches with a 1GB trunk. This happens even off-hours when the network isnt being used at all. There are never more than 2 people on the phone at the same time and it is definitely not an over-utilized processor. I have trying to figure this out for 2 months on and off with no success any help is appreciated. Thanks Andrew Shelton <andrew.shelton@stemnetworks.co.uk> wrote: v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);} st1\:*{behavior:url(#default#ieooui) } What codec are you using? How many phone? What load is the server under? --------------------------------- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of sdgesa gaeharth Sent: 05 October 2006 13:22 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute 1)Can anyone tell me how to do this on a Polycom 501? 2)Can you explain why you think this any why it ony happens on some calls? Thanks Andres <andres@telesip.net> wrote: > > > For about 20% of the calls to the outside world, the voice on the > other end of an outside line is incredibly choppy. Enough to where > we have to hang up and call on a cell phone. It is always the same > numbers that are choppy. The funny thing is, if I press mute while > talking on a choppy call, the choppiness goes away completely. > > > Maybe you have silence suppression enabled on your phones. Try to disable it and see if it helps. >------------------------------------------------------------------------ > > > -- Andres Technical Support http://www.telesip.net _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2"/min or less. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --------------------------------- Get your email and more, right on the new Yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061005/e87282 02/attachment-0001.htm ------------------------------ Message: 5 Date: Thu, 05 Oct 2006 09:38:07 -0400 From: "J. Oquendo" <sil@infiltrated.net> Subject: Re: [asterisk-users] Re: Bandwidth requirements To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <45250ABF.8070506@infiltrated.net> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Benny Amorsen wrote:>>>>>> "rJ" == raphael Jacquot <sxpert@sxpert.org> writes: >>>>>> > > rJ> ATM cell tax is actually 10% as there's 5 header bytes for each 53 > rJ> bytes cell, > > For VoIP the cell tax is much larger. In the example, each RTP packet > contains 20 useful bytes and 40 bytes IP overhead. 60 bytes doesn't > fit in one cell, so you end up with 106 bytes at the ATM layer to > transport 20 bytes of G.729. The ATM-caused overhead is thus 46 bytes > per voice packet, thereby making the needed bandwidth 77% larger. > >CRTP solves this issue (40byte waste) -- ===================================================J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ------------------------------ Message: 6 Date: Thu, 5 Oct 2006 08:42:16 -0500 From: "Joe Pukepail" <pukepail@gmail.com> Subject: Re: [asterisk-users] snom 360: how to make record button working ? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <7b3aa3a40610050642l41c41980xa283758f0072e742@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" There was a patch to get this working, looks like it has been abandoned, though. Should give you a starting point to get it working, or perhaps a bounty would get someone interested in getting it usable and committed. http://bugs.digium.com/view.php?id=4845 On 10/4/06, Joel Hill <jhill@asteriskit.com.au> wrote:> > Hi Noro, > > Depending on what firmware you have this is the way to go. > Go to the Functions keys page, then look for the Record button, Change > the type to DTMF and in number put in *1 which is the default Asterisk > recording function. > > Hope this helps > > Cheers, > > Joel > Asterisk IT > www.asteriskit.com.au > > > noro kamen wrote: > > Hi, > > > > I'd like to make record button working on snom 320/360 + asterisk. > > > > As I learned from wireshark output, the phone produces SIP info > > message "Record: on", while record button pressed. > > > > Can anybody give me an advice, how to teach asterisk to understand > > that SIP info message and start recording ? > > > > TIA > > noro > > _______________________________________________ > > --Bandwidth 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061005/1b923b b0/attachment-0001.htm ------------------------------ Message: 7 Date: Thu, 5 Oct 2006 08:43:26 -0500 From: "James" <jltaylor@metrotel.net> Subject: Re: [asterisk-users] TNT Max Password reset To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <014d01c6e884$3bd21a70$2c05a8c0@table> Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=original I have five MAX TNT's runnig with SIP and g.729. They will do E1's, T1's, T3's. James Taylor 1-903-793-1956 ----- Original Message ----- From: "Steve Kennedy" <steve-asterisk@gbnet.net> To: <asterisk-users@lists.digium.com> Sent: Thursday, October 05, 2006 4:28 AM Subject: Re: [asterisk-users] TNT Max Password reset> On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote: > >> On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote: >> > Anyone have happen know how to reset the password on a TNT Max? >> > Thanks. >> Does your asking here suggest that the the MAX's can do, say, voice >> gateway service? Protocols? Codecs? > > Ascent TNT's with the right software and hardware can do SIP, E1 > termination/origination, and all sorts of codecs. > > Similar functionality to Cisco AS5200'ish. > > > Steve > > -- > NetTek Ltd UK mob +44-(0)7775 755503 > UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 > Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net > Euro Tech News Blog http://eurotechnews.blogspot.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 >------------------------------ Message: 8 Date: Thu, 05 Oct 2006 09:50:54 -0400 From: jose diaz <ing.josediaz@verizon.net.do> Subject: [asterisk-users] two asterisk and one NBX system To: asterisk-users@lists.digium.com Message-ID: <45250DBE.7020300@verizon.net.do> Content-Type: text/plain; charset=ISO-8859-1; format=flowed We have three servers: Two asterisk and one NBX 3COM. The connection between Asterisk1 and Asterisk2 is with IAX2. The connection between Asterisk2 and NBX is with a Digium analog TDM400P (2FXO and 2 FXS) The dial plan Asterisk1: 3XXX The dial plan Asterisk2: 2XXX The dial plan NBX: 1XXX The system work well, but the call from Asterisk1 to NBX fail. I'm using the IAX2 protocol to call from asterisk1 to asterisk2, i need to trasnfer the call to the NBX. How i can to make that? Regards, Jose Diaz ------------------------------ Message: 9 Date: Thu, 5 Oct 2006 10:01:43 -0400 From: "Noah Miller" <noahisaacmiller@gmail.com> Subject: Re: [asterisk-users] Video Conference To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <8699dcab0610050701m32130387w56a2299b4965d702@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi Bilal -> We need to apply Video conference, can asterisk > support this?No. Asterisk supports video calls between two end points, but not video conferences with three or more participants. There is a bounty for someone to add this feature, but nobody has successfully implemented it yet.> What I need for that?Something else. You can get video conferencing software, or if you have the right hardware you can use it. There are many hardware video conferencing units available from Polycom, Tandberg, Sony, etc. - Noah ------------------------------ Message: 10 Date: Thu, 5 Oct 2006 16:04:55 +0200 From: "asterisk" <asterisk@sirtem.fr> Subject: RE: [asterisk-users] TNT Max Password reset To: "'James'" <jltaylor@metrotel.net>, "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <200610051404.k95E4iku025944@ra.sirtem.fr> Content-Type: text/plain; charset="iso-8859-1" Hello james, I have 1 max with pri, only used for incomming data call. It is a old box, where to find firmware for this unit ? If a can use it for voice.... Ps: i leave in France.. Many thanks... -----Message d'origine----- De : asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] De la part de James Envoyi : jeudi 5 octobre 2006 15:43 @ : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] TNT Max Password reset I have five MAX TNT's runnig with SIP and g.729. They will do E1's, T1's, T3's. James Taylor 1-903-793-1956 ----- Original Message ----- From: "Steve Kennedy" <steve-asterisk@gbnet.net> To: <asterisk-users@lists.digium.com> Sent: Thursday, October 05, 2006 4:28 AM Subject: Re: [asterisk-users] TNT Max Password reset> On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote: > >> On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote: >> > Anyone have happen know how to reset the password on a TNT Max? >> > Thanks. >> Does your asking here suggest that the the MAX's can do, say, voice >> gateway service? Protocols? Codecs? > > Ascent TNT's with the right software and hardware can do SIP, E1 > termination/origination, and all sorts of codecs. > > Similar functionality to Cisco AS5200'ish. > > > Steve > > -- > NetTek Ltd UK mob +44-(0)7775 755503 > UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 > Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net > Euro Tech News Blog http://eurotechnews.blogspot.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 ------------------------------ Message: 11 Date: Thu, 5 Oct 2006 10:04:11 -0400 From: "Don" <sales@xwebfactor.com> Subject: Re: [asterisk-users] TNT Max Password reset To: "James" <jltaylor@metrotel.net>, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <02e401c6e887$23506c60$1d01a8c0@shizznit2000> Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=response We used to use em... I believe you can just use a serial connection to them and reset them... Could be mistaken been a couple years now... ----- Original Message ----- From: "James" <jltaylor@metrotel.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Thursday, October 05, 2006 9:43 AM Subject: Re: [asterisk-users] TNT Max Password reset>I have five MAX TNT's runnig with SIP and g.729. > They will do E1's, T1's, T3's. > > James Taylor > 1-903-793-1956 > > > ----- Original Message ----- > From: "Steve Kennedy" <steve-asterisk@gbnet.net> > To: <asterisk-users@lists.digium.com> > Sent: Thursday, October 05, 2006 4:28 AM > Subject: Re: [asterisk-users] TNT Max Password reset > > >> On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote: >> >>> On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote: >>> > Anyone have happen know how to reset the password on a TNT Max? >>> > Thanks. >>> Does your asking here suggest that the the MAX's can do, say, voice >>> gateway service? Protocols? Codecs? >> >> Ascent TNT's with the right software and hardware can do SIP, E1 >> termination/origination, and all sorts of codecs. >> >> Similar functionality to Cisco AS5200'ish. >> >> >> Steve >> >> -- >> NetTek Ltd UK mob +44-(0)7775 755503 >> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 >> Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net >> Euro Tech News Blog http://eurotechnews.blogspot.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 > > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.1.407 / Virus Database: 268.12.13/463 - Release Date: 10/4/2006 > >------------------------------ Message: 12 Date: Thu, 05 Oct 2006 16:07:14 +0200 From: Michael Neuhauser <mike@firmix.at> Subject: Re: [asterisk-users] Re: extensions.conf strangeness To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Cc: Brian Candler <B.Candler@pobox.com> Message-ID: <1160057234.9804.13.camel@willow.firmix.at> Content-Type: text/plain; charset="us-ascii" On Thu, 2006-10-05 at 11:12 +0100, Brian Candler wrote:> Is there a debug mode which can say: > > "dialplan: trying to match 611 against pattern _1XXXXX: failed > dialplan: trying to match 611 against pattern _2XXXXX: failed > dialplan: trying to match 611 against pattern _6X.: matched"No, there isn't (I assume to keep this central part as fast as possible, i.e., even "if (option_debug) ..." costs time and pollutes the cache). I've created and attached a one line patch (for 1.4 branch, r44464) that should give you the info you need (sort of). But be aware that I haven't tested it on 1.4 (only on 1.2, but things are different there). Only use this patch on a test system as it will generate massive amounts of output and will considerably slow down call handling. -- Dr. Michael Neuhauser mailto:mike@firmix.at Firmix Software GmbH sip:mike@firmix.at Vienna/Austria/Europe tel:+43-1-7890849-30 Linux Development and Services http://www.firmix.at/ -------------- next part -------------- A non-text attachment was scrubbed... Name: matchdebug.patch Type: text/x-patch Size: 559 bytes Desc: Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20061005/5e45c0 e7/matchdebug.bin ------------------------------ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 27, Issue 23 **********************************************
K Y Iyer
2006-Oct-06 01:19 UTC
[asterisk-users] Asterisk Configuration Complete Newbie question
Hello Am starting on my Asterisk journey - am getting a single span Digium card to connect Asterisk to our Alcatel 4400 EPABX and install about 100 VoIP instruments. The Asterisk VoIP extensions and Alcatel digital extensions have to talk to each other. Am I right in understanding that IN ASTERRISK : I have to create a config with either all Asterisk and Alcatel extensions - which config files? extensions.conf for both with the two types of extensions in different contexts? Would that be the correct way? IN ALCATEL : List of Asterisk extensions and the PRI card to which the calls have to be delivered. Is that broadly correct? Thanks very much Best wishes Iyer
Lacy Moore - Aspendora
2006-Oct-06 03:07 UTC
[asterisk-users] Asterisk Configuration Complete Newbie question
On 10/6/06, K Y Iyer <Iyer@ndtv.com> wrote:> > Is that broadly correct?Yes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061006/0976d559/attachment.htm
K Y Iyer
2006-Oct-06 03:13 UTC
[asterisk-users] Asterisk Configuration Complete Newbie question
Thanks very much - let me see how far I can take it now. Best wishes Iyer -----Original Message----- From: asterisk-users-bounces@lists.digium.com on behalf of Lacy Moore - Aspendora Sent: Fri 10/6/2006 03:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Configuration Complete Newbie question On 10/6/06, K Y Iyer <Iyer@ndtv.com> wrote:> > Is that broadly correct?Yes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061006/80f5b851/attachment.htm