Re: [asterisk-users] Re: ZapHFC quadBRI D-Channel going down randomly
I tried the same, and my Telco company told me (although sometimes it's hard to trust them, you never know what kind of guy from the call center is answering your call) that p2p lines already have l1 permanent. Nonetheless it goes down sometimes for quite long periods. I'm starting wondering whether it's some sort of kernel related problem (i.e. irq sharing settings etc.) by which the card loses packets. Henrik Woffinden ha scritto: I have the exact same problem on a normal ISDN2 BRI line. I solved it by having my Telco put layer 1 to permanent. Best regards, Henrik Woffinden Alberto Pastore wrote: asterisk ha scritto: On most traditional pabx's it's possible to set layer 1 to permanent or call. It sounds like your system is configured for permanent and your lines to call. How you would set this on asterisk I have no idea. fadge The question is: is it possible I am the only one with such problems on all asterisk boxes on different sites and different ISDN lines? I've googled around on many forums but no one seems to have this one. The old replaced PBXs had layer 1 set for call, as you say, and they showed no problems at all. With asterisk as a PBX, every 2-3 hours, you cannot dial out for 5 to 15 minutes then everything gets back to normal (no idea about what triggers the return to working state). ___ --Bandwidth 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] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M
If you have divas4linux package installed (from Eicon), you can use the Config textual gui utility, it always reports which cards (model and revision) are found in your system. Klaus Darilion ha scritto: Hi (Armin)! Does someone knows how to identify the type of the card? The delivery note says it is a V-4BRI-8M, whereas lspci reports a 4BRI-8M. What is it really? Are there any Eicon tools to identify the card type? thanks klaus :0a:03.0 Network controller: Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 (rev 01) Subsystem: Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32, Cache Line Size: 0x10 (64 bytes) Interrupt: pin A routed to IRQ 77 Region 0: Memory at fdeffc00 (32-bit, non-prefetchable) [size=256] Region 1: I/O ports at cc00 [size=256] Region 2: Memory at fc00 (32-bit, non-prefetchable) [size=16M] Region 3: Memory at fdee (32-bit, non-prefetchable) [size=64K] Capabilities: [40] Power Management version 1 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [48] #06 [0080] Capabilities: [4c] Vital Product Data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.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
[asterisk-users] Asterisk Realtime with ODBC/MySql
We are testing Asterisk Realtime configuration with ODBC/MySql. This is our extconfig.xonf extensions = odbc,asterisk,extensions In the extension.conf we have inserted this line: switch = Realtime/[EMAIL PROTECTED] where mainmenu is the context. and in the table 'extensions' we have this line: ID CONTEXT EXTEN PRIORITY APP APPDATA 1 mainmeu 666 1 voicemail (somedata) When we try to call 666 asterisk respond with the following error: WARNING[8906]: pbx.c: 2404 __ast_pbx_run: Invalid extension '6', but no rule 'i' in context 'mainmenu'. Can someone help me? Thank's, Maury ___ --Bandwidth 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] Inaccurate CDRs
This is an update on the issues of CDR inaccuracies, i hope this will help someone in need.
In order to remove the Authenticate() function and still be able to perform call accounting and authentication, we pass the authetication process through an AGI. the reason beig that the Authenticate() application always awnsers the channel first before authentication. This will cause irregularities in terms of billing since the user will have to pay for both the 'authentication' time and the call processing time. My authentication Macros calls this perl script to enable authentication without awnsering the channel.
#!/usr/bin/perl
use Asterisk::AGI;open (PIN1,'/etc/asterisk/pinset_1');open (PIN2,'/etc/asterisk/pinset_2');open (PIN3,'/etc/asterisk/pinset_3');$|=1;$AGI = new Asterisk::AGI;
exit if $#ARGV 0;chomp($dbid=$ARGV[0]);#$AGI-exec(NoOp,$dbid) ;%input = $AGI-ReadParse();$try = '0';$filename = 'agent-pass';while($try 3) { $pin = $AGI-get_data($filename,2000);
if (!defined $pin) {$try++;$filename = 'auth-incorrect';next;} if (length($pin)!=4) {$try++;$filename = 'auth-incorrect';next;} $status=check_pin($pin);
if ($status =='1'){ $AGI-exec(NoOp,$pin); $AGI-stream_file('auth-thankyou'); #$AGI-exec('Playback','auth-thankyou','noanswer');
$AGI-exec('SetAccount',$pin); exit (0); } else { $try++; $filename = 'auth-incorrect'; #$AGI-exec('Playback','') next;
}}$AGI-exec('Playback','vm-goodbye');$AGI-exec('Wait','1');$AGI-hangup();exit (0);
sub check_pin{ my $pin=$_[0]; my $file='PIN'.$dbid; seek ($file,0,0); while ($file){ if (/$pin/) {return '1';} } return '0';
}
you have to install the Asterisk::AGI module for thi script to work. i used the authentication macros from AMP (FreePBX) to pug in this agi.
Every trunk thatrequiers authentication calls this macros. The macros is as follows..
[macro-pinsets]include = macro-pinsets-customexten = s,1,GotoIf(${ARG2} = 1?cdr,1)exten = cdr,1,AGI(auth.agi|${ARG1})
; end of [macro-pinsets]
i hope this will be of use to someone
On 10/18/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote:
I have found the problem.
Before calls leave our network, thee user must supply a pin. this is a for of call accounting that we implemented. To do this, we had used AMP's Authenticate () function. This function actually and always answers the channel first before accepting pin entries. This was why there is always an answered flag on the channel. and since the channel is answered as soon as the call is made, there is no difference between the duration and the billsec. Now my problem is how do i implement an authentication AGI that uses DTMF ? i would be posting this question in another thread
Thanks for your help
On 10/17/06, Dumpolid Exeplish [EMAIL PROTECTED]
wrote:
this Cdr Record if from the Primary PBX
'2006-10-17 07:11:37', 'Admin', 'XXX, 'aa', 'from-internal', 'IAX2/[EMAIL PROTECTED]'
, 'Zap/1-1', 'ResetCDR', 'w', 10, 0, 'BUSY', 3, '', '', ''
this is the CDR record from the secondsry for the same call
'2006-10-17 13:31:57', 'Admin X', 'X', 'aa', 'from-internal', 'SIP/401-8f0c', 'IAX2/TRUNK1-2', 'Dial', 'IAX2/TRUNK1/aaa|120', 15, 15, 'ANSWERED', 3, '4147', '', ''
in this setup, the caller dropped the call after allowing it to ring for 15 seconds
On 10/17/06, Dumpolid Exeplish [EMAIL PROTECTED]
wrote:
Well I am using APM on the two boxes i have modified the srripts extensievely and i am sure that there is no Awnser befor a dial when Dialing through the PBX trunks
On 10/17/06, Steve Davies [EMAIL PROTECTED]
wrote:
On 10/17/06, Dumpolid Exeplish
[EMAIL PROTECTED] wrote: Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for
even uncompleted calls ( i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my call disposition colume. I have also noticed that both the call duration and
the billsec are always equal. here is my setup below PSTN va E1 (Primary Asterisk) =Sip and IAX trunks (Secondary PBX)
Clients are connected to the Secondary PBX. this pbx handles registration of all clents. The billing irregularities happen on the Secondary PBX. When a call is maked from the Secondary and it is routed across the trunks, call
disposition always registeres 'AWNSERED', unless the Primary PBX sends back a busy signal. the duration and billsecs are always equla. this means that the user gets billed for ring time, and calls disconnected from the
Secondary PBXCould you provide a snippet of the dialplan used on each of theprimary and secondary boxes to complete a call?For example, is the primary executing an Answer() before it does the
onward Dial() on behalf of the secondary?Cheers,Steve___--Bandwidth and
Re: [asterisk-users] random one way audio and noise betweenSIP phoneson same LAN
Hi Scott, so it seems that are polycom phones not working well... have you tried with other IP phones or only with polycom? Giorgio Incantalupo Scott Scecina wrote: Giorgio, I'll answer in reverse order: I've not had reports of noise from my users. However, when I went down to get the s/w version from the phone that seems to be acting up the most, the user reported that earlier they were actually on a call that was ok then spontaneously dropped the audio. Per my instructions (based on another similar report I read on Digium's site), my user hit a digit on the phone which brought back the caller's audio. I've also had them attempt to put the call on hold, and then resume, but that did not bring the audio back. As far as the S/W versions: One of the phones that acts up (and they all should match): Polycom 501 BootRom: 3.1.3.0131 BootBlock: 2.5.0 SIP: 1.6.6.0036 My phone, on which I've never experienced the problem: Polycom 601 BootRom: 3.1.3.0131 BootBlock: 2.6.0 SIP: 1.6.6.0036 - Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Wednesday, October 18, 2006 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] random one way audio and noise betweenSIP phoneson same LAN Hi Scott, seems that we have the same problem...I have canreinvite=no and polycom phones. I do not have cisco switch and qualify=yes but I think that is not the problem. I've got 2 questions: 1) my polycom firmware is: sip.ver: 1.6.5.0043 bootrom.ver: 2_6_2 what are yours? 2) have you got one way calls only or noise on sip calls conversations too? TIA Giorgio Incantalupo P.S.: for configuration/monitoring apps I'm still on it...I hope to find useful tools asap. In case, I'll let you know. Scott Scecina wrote: I'm having the same random problem. I have canreinvite=no on all extensions. I have qualify = yes on all non-NAT extensions. I do have several NAT extensions, but I've not had reports of problems from those. 95% of my extensions (all polycom 501/601) are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches. In all cases, the called party cannot hear the calling party. The calling party has the still ringing icon on their phone, but can hear the called party talking. I've got call monitoring turned on, and asterisk is recording both sides of the conversation. The problem occurs on SIP-SIP and Zap-SIP calls. I've tried enabling sip debug on a particular extension that seemed to be experiencing the problem more than others. However the problem did not occur when the debugging was on. Sip debug generates so much noise I've been hesitant to turn it on system-wide. Is there a way I can turn on sip debug and have all that logging go to a specific file (and not in the asterisk console)? Also, are there any other configuration/logging tricks I can try? Thank you, Scott Scecina ___ --Bandwidth 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] SIP users with Database
Hi, I'm testing Asterisk with MySql and I would want to insert sip users in a table "sip_users". After I modified extconfig.conf with "sipusers = odbc,asterisk" and I create the table sipusers, which changes must I make to sip.conf? Thank's Maury P.S.: C'è qualche utente italiano nella mailing list? ___ --Bandwidth 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] accountcode and amaflags?
Hello ppl, Can someone explain to me the meaning and use of the variables accountcode and amaflags in sip.conf,etc. Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I know, they are billing related, but not much beyond that. Any ideas? cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 - versionStamp
If I put versionStamp in cnf.xml file, how do I check it on the phone? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Digium on Dell PowerEdge 1850
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... We're running 2 TE412P's in a Dell 1850 just fine, been running like this for well around 6 months to a year now without any problems. They're not exactly 212P's but I imagine it won't be much different. On Wed, 2006-10-18 at 10:54 +0200, Tomislav ParÄŤina wrote: Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Hi Aaron! Thank you for your mail. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Got reject for frame XX, retransmitting frame XX now, updating n_r!
Hi all, What does 'Got reject for frame...' message really mean, what could be causing it, and how should one start troubleshooting it? Thanks in advance, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which is the best ?
Hi, As I am a newbie, I am going to ask a newbie question ;-) I saw Digium has TE212P (with DSP) and TE210P cards (no DSP), and Sangoma has A102 T1/E1 AFT card (no DSP)... What is the best choice of T1/E1 card (2ports) for an installation on Centos 4 (or simply what is the best choice of T1/E1 card ?) ? Digium or Sangoma cards? We would to build up a between small and medium system (and hope that it will grow to a large one ;-) :-P !) Thanks you 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] echotraining=yes in misdn.conf is invalid or out of range.
Hi Jarkko, I had the same problem..It worked with an old version of misdn-install (taken from beronet site) but not with actual mqueue-misdn-install. I tried to put it in every misdn.conf section I have without success. The updated beronet install manual doesn't mention that parameter anymore so I removed it from misdn.conf. Giorgio Incantalupo Jarkko Nevala wrote: Hi. I'm having problems with chan_mISDN configuration. Line echotraining=yes causes warning, when Asterisk is parsing misdn.conf and I'm confused why the PBX doesn't accept the setting. No matter which section I try to offer it, it is always invalid or out of range. The setting itself is supposed to be valid, it is in the sample configuration file of chan_mISDN 0.3.1. When I list the configuration of the ISDN-ports with misdn show config, I can find values for echocancel and echocacelwhenbridged, but no mention about echotraining. I'm running Asterisk 1.2.10 on OpenSuse 10.1 with chan_misdn-0.3.1-rc23. The hardware platform is HP server with Intel XEON processor and three hfcpci BRI -cards in TE -mode. - Here's the message I get: == Parsing '/etc/asterisk/misdn.conf': Found Oct 19 00:33:54 WARNING[4443]: misdn_config.c:660 _build_port_config: misdn.conf: echotraining=yes (section: default) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. - And here is my misdn.conf: [general] debug=1 bridging=no tracefile=/var/log/asterisk/misdn.trace [default] echocancel=yes echotraining=yes hold_allowed=yes screen=-1 presentation=-1 senddtmf=yes [isdn_call] ports=1,2,3 context=isdn_in msns=* - Have you got any idea what is causing this and how I could get the echo training working? Thank you for your help. Jarkko ___ --Bandwidth 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] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?
Hi, I have a sangoma PRI card on an Asterisk PBX. I have problem with outgoing caller ID: when I make an outbound call, the called party gets x1 instead of x240 where x is the my company prefix and 240 is the phone extensions I call from. I read something about usecallingpres on wiki but nothing is told about default and possible values. Is this parameter the real cause of my wrong caller id? TIA Giorgio Incantalupo ___ --Bandwidth 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] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?
Giorgio Incantalupo wrote: Hi, I have a sangoma PRI card on an Asterisk PBX. I have problem with outgoing caller ID: when I make an outbound call, the called party gets x1 instead of x240 where x is the my company prefix and 240 is the phone extensions I call from. I read something about usecallingpres on wiki but nothing is told about default and possible values. Is this parameter the real cause of my wrong caller id? How are you setting the caller id before dialing? Show us the section of your dial plan that handles this. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel not detecting Tormenta2 PRI Interface card
Hi, After upgrading to new version of Asterisk and Zaptel, my Tormenta2 card has stopped working. Zaptel doesn't detect it. What should I do to make it work again. It is enabled in /etc/sysconfig/zaptel. Where else I have to enable it so zaptel can detect it and make it work. zaptel.conf and zapata.conf are the same as I had them before. I tried googling but couldn't find any help.-- Zeeshan A Zakaria ___ --Bandwidth 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] say Asterisk to answer
Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on Accept button. Greg ___ --Bandwidth 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 gotoiftime and Macro question
Thank you all very much: it solved my problem.
exten = 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?novm,567,1)
...
[novm]
exten = _X.,1,Macro(exten-vm,novm,${EXTEN})
...
The remaining dialplan is not my specific,
is the standard dial plan provided by FreePbx, which I integrate in
_custom.conf file
thanks,
Andrea
Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.
Visitate il sito http://www.frameweb.it
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Re: [asterisk-users] say Asterisk to answer
It's not possible. The idefisk however has a button to auto answer. Zoa Gregory Duchatelet wrote: Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on “Accept” button. Greg ___ --Bandwidth 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] SIP users with Database
Hi Maurizio,http://www.voip-info.org/wiki/view/Asterisk+RealTime+SipOften you can find what you seek just spending a minute with google Cheers,Giovanni (Italiano)2006/10/19, Maurizio Pederneschi [EMAIL PROTECTED]: Hi, I'm testing Asterisk with MySql and I would want to insert sip users in a table sip_users. After I modified extconfig.conf with sipusers = odbc,asterisk and I create the table sipusers, which changes must I make to sip.conf? Thank's Maury P.S.: C'è qualche utente italiano nella mailing list? ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth 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] say Asterisk to answer
Hi Greg,Idefisk support Auto-answer only in a biz versionI suppose you got free version..You will find more details http://www.asteriskguru.com/idefisk/free/ Cheers,Giovanni2006/10/19, Gregory Duchatelet [EMAIL PROTECTED]: Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on "Accept" button. Greg ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth 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] Server power indication
I've narrowed it down to 2 configs, It wil either be a dual: Intel xeon dual core 3.73Ghz 1066 2x2MB cache or a dual: AMD Opteron DUAL CORE 285 (2.6GHz 32/64bit) So effectively there will be 4 real CPU cores to handle processes/transcoding. * Are there any numbers on how many (SIP-SIP) Alaw to G729 transcoding calls this setup will be able to handle? * Which of these two would be the better? * Would this configuration be enough to serve a TE412P to 120 channels G729 ? Kind regards, Erik Erik wrote: Hello list, I'm currently looking into building a new Asterisk server, due to some codec problems i've got to transcode most of my channels between Alaw -- G729. Is there any indication on how many channels you would be able to transcode on a certain platform? I'm looking into dual Xeon or dual Opteron configurations, which of these platforms would perform better? And how much power would be needed to transcode 120 Channels PRI to G729 (for example Digium TE412P)? Erik ___ ___ --Bandwidth 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] Electric usage of a tdm400p
Erick, It looks like the 2.5 laptop drive requires 5 watts to spin up. Adding that to the 15 watts for the Digium card, leaves about 40 watts available for the MB. It's unlikely that the system will be producing ring voltages when the drive is spinning up. It depends on how conservative you may be with real-time power management, e.g. spinning drives down when not in use, etc. I did not easily find too many ITX MB power requirements, but the one I did find, required 45 watts (peak). In the worst case, ringing 5 RENs on each Digium port and spinning up the disk, you would be overtaxing the power supply. I doubt you will have 5 RENs on each port and all ringing, but you could. In ages past, hard drives were the most vulnerable to poor power regulation, but that may have changed. With the higher cost of 2.5 drives, I would not take any chances. Beefing up the power supply would also eliminate the need for manually managing power should you need a CDROM or more power hungry drive in the future. It's also one less concern when troubleshooting the system. As Moj has pointed out, problems can occur when working close to the edge. I, too, have experienced similar problems when power was limited and have had to, temporarily, resort to a bigger power supply to get a system installed. Then fell back to a smaller one in operation. Good luck. Bob... On Wed, 2006-10-18 at 08:49 -0800, Mojo with Horan Company, LLC wrote: I set up a similar system on an VIA Epia 5000, and I had issues when I included the CDROM in the mix. I had to use another ATX power supply to complete the install, but then once I removed the CDROM drive I had no power issues. I presume you could install the OS with the CDROM drive installed and the molex power connector REMOVED from the TDM card, then when the OS was installed and you had network connectivity, power down, remove the CDROM, add the power supply for the TDM card, then install zaptel etc. Or just try it and tell us what happens, low power won't break it in my experience. Your cdrom drive might have a lower power consumption than mine. Moj Erick Perez wrote: Well Im planning to use a mini-itx, a laptop hdd and a 4fxs digium card. the mini-itx comes with a 60W DC to DC adapter (80W peak). So I need power to manage the hdd, motherboard,the tdm card. A disk cable can be made available, but is not present as a factory default. So My real concern is power. On 10/18/06, Bob Chiodini [EMAIL PROTECTED] wrote: On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote: Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? Erick, Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN ~5). This translates to 2.7 watts. Assuming a DC/DC converter efficiency of 38% (probably low), you would need about 3.7 watts, per FXS module. About 15 watts, total. What is the TDM card installed in and is a disk drive cable available? Bob... ___ --Bandwidth 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] Access Denied on a Windows share
Message: 12 Date: Tue, 17 Oct 2006 18:07:04 -0700 (PDT) From: sdgesa gaeharth [EMAIL PROTECTED] Subject: Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 None of these steps have made a difference. Any other suggestions? Here is my original post: Can anyone help me to figure out why I can not write to a public share? I was able to join the domain without a problem. I can access the share from an xp box. I just can not write: Access denied. thanks If the person or process that is trying to write to the share is a member of a group that is denied access then the write will fail with access denied. Check the effective permissions for a user on the file and look at the server security logs which may give you some additional information. Paul Gaffney LANStatus, LLC ___ --Bandwidth 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] accountcode and amaflags?
amaflags : Categorization for CDR records. Choices are default, omit, billing, documentation and choices are defaul, omit, billing, documentationaccountcode : string : Users may be associated with an accountcode (billing purpose) Cheers,Giovanni2006/10/19, Benjamin Jacob [EMAIL PROTECTED]: Hello ppl,Can someone explain to me the meaning and use of the variablesaccountcode and amaflags in sip.conf,etc.Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. Iknow, they are billing related, but not much beyond that. Any ideas?cheerz- Ben.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___ --Bandwidth 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] question about CDR command
In cdr_mysql.conf add userfield=1 under the globals setting. bp On 10/18/06, unplug [EMAIL PROTECTED] wrote: I want to set some custom data in the field of userfield in table CDRas following.exten = s,19,Set(CDR(userfield)=1234) exten = s,20,Dial(SIP/1234)However, the userfield doesn't get update after making the call.After that, I relocate the command as following.exten = s,19,Dial(SIP/1234)exten = s,20,Set(CDR(userfield)=1234) The userfield doens't get update at all.I don't know why the fieldcan't update after issuing the command.Anyone can help?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] accountcode and amaflags?
Giovanni, Appreciate your lines mate. But, Ive already read those, all over the net. my qs inline : amaflags : Categorization for CDR records. Choices are default, omit, billing, documentation and choices are defaul, omit, billing, documentation wot r these categories??wot decides these categories? accountcode : string : Users may be associated with an accountcode (billing purpose) hmm.. ive seen in quite a few places, where the pin collected is stored as the accountcode... wot duz that mean? anyway, can you give me an example of wot the association means?am a lil slow.. Cheers, Giovanni 2006/10/19, Benjamin Jacob [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello ppl, Can someone explain to me the meaning and use of the variables accountcode and amaflags in sip.conf,etc. Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I know, they are billing related, but not much beyond that. Any ideas? cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth 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] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M
Hi! lspci -nv reports: :0a:03.0 0280: 1133:e013 (rev 01) Subsystem: 1133:e013 Thus, I suspect I really got a 4BRI-8M V2 Also divactrl reports a 4BRI: bbgast01:~# /usr/lib/divas/divactrl ctrl -c 1 -CardName Diva Server 4BRI-8M 2.0 PCI Let's test faxing :-) thanks klaus Armin Schindler wrote: On Wed, 18 Oct 2006, Klaus Darilion wrote: Hi (Armin)! Does someone knows how to identify the type of the card? The delivery note says it is a V-4BRI-8M, whereas lspci reports a 4BRI-8M. What is it really? Are there any Eicon tools to identify the card type? As far as I know these cards are almost identical, but the PCI ID must be different. Maybe the pci id database doesn't have this difference... What PCI-ID does it have? 0xE012 = 4BRI-8M 0xE013 = 4BRI-8M V2 0xE016 = Voice 4BRI-8M 0xE017 = Voice 4BRI-8M V2 There is no special tool. When you load the divas driver, it should announce the cards found. And the divactrl utility uses divas to get the cards info and can tell the correct version as well, e.g.: divactrl ctrl -c 1 -CardName Armin ___ --Bandwidth 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] Bristuff qozap drivers problem
Hi, For a significant time now (since about 0.2.0-rc8n) the qozap driver has become very verbose if an ISDN line is not connected... I get the messages below every couple of seconds in the asterisk logs. The flaw in the messages is the Alarm cleared message - The alarm cannot possibly be cleared because there is no physical media connected into that port!!! (BTW - All ports are in TE mode.) Can anyone suggest a cleanup in qozap.c that will prevent it telling Asterisk that the channel is up unless it actually has come back up? I do not understand the zaptel/bristuff internals well enough to be able to find where this is occuring. I also get a solid kernel crash with no Oops if I unload the qozap module - Again this does not happen in the older versions of the qozap module. I am using Kernel 2.6.10. Many thanks for any pointers, Steve. Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 4: Red Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 4 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 5: Red Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 5 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 7: Red Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 7 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 8: Red Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 8 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 10: No Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 10 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event: Detected alarm on channel 11: No Alarm Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable to disable echo cancellation on channel 11 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 4 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 5 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 7 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 8 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 10 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm cleared on channel 11 Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 2 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 6 as D-channel anyway! Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 2 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 6 as D-channel anyway! Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 3 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 9 as D-channel anyway! Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 3 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 9 as D-channel anyway! Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 4 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 12 as D-channel anyway! Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 4 Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No D-channels available! Using Primary channel 12 as D-channel anyway! ___ --Bandwidth 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] Multiple bridge attempts
Hi, I have an Asterisk box connected via an anaogue lines(ZAP/1-1) to a Siemens PBX. I take calls off the PBX and put send it to a premicell connected via ZAP/7-1. Calls orginate from the PBX, hit Asterisk, then get sent to the premicell. Can anyone tell me why there is multiple bridge attempts? I am used to there been only one. -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, ZAP/R2/0727228489|40|L(360)) in new stack -- Called R2/0727228489 -- Zap/7-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth 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] Please help with these SIP errors
Hi, sometimes on my Asterisk 1.2.10 box I get these errors, there are about 50 active SIP channels so I dont know if calls are getting dropped or not. Should I be worried? 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! -- Executing GotoIf(SIP/sipCSC-b737f9e8, 0 ? 15) in new stack 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! == Spawn extension (iax, 0837707300, 34) exited non-zero on 'SIP/sipCSC-b73aba28' 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11347 sipsock_read: We could NOT get the channel lock for SIP/sipCSC-b73aba28! 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11348 sipsock_read: SIP MESSAGE JUST IGNORED: ACK 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11349 sipsock_read: BAD! BAD! BAD! == Spawn extension (iax, 0825905581, 24) exited non-zero on 'SIP/sipBBG-b736f910' -- thanks, yusuf -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth 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: Digium on Dell PowerEdge 1850
Hi Tomislav, I use Dell hardware for desktops only. Each time I tried to use a Dell pc with telephony cards I get problems. It works only with a TDM400 but if u plan to add something more it is a real nightmare! Giorgio Incantalupo Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... We're running 2 TE412P's in a Dell 1850 just fine, been running like this for well around 6 months to a year now without any problems. They're not exactly 212P's but I imagine it won't be much different. On Wed, 2006-10-18 at 10:54 +0200, Tomislav ParÄŤina wrote: Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Hi Aaron! Thank you for your mail. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP users with Database
Hi Giovanni, I follow step by step the documentthat you suggest. The connection between asterisk and MySql works fine, but sip users can't resgister. If I query my sipuser tablethe command realtime load family column value I have not any result... What can I check in my configuration? Can anyone give me some example of the configuration files in order to work with odbc/mysql? Thank's, Maury - Original Message - From: Giovanni Miano To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, October 19, 2006 12:15 PM Subject: Re: [asterisk-users] SIP users with Database Hi Maurizio,http://www.voip-info.org/wiki/view/Asterisk+RealTime+SipOften you can find what you seek just spending a minute with google Cheers,Giovanni (Italiano) 2006/10/19, Maurizio Pederneschi [EMAIL PROTECTED]: Hi, I'm testing Asterisk with MySql and I would want to insert sip users in a table "sip_users". After I modified extconfig.conf with "sipusers = odbc,asterisk" and I create the table sipusers, which changes must I make to sip.conf? Thank's Maury P.S.: C'è qualche utente italiano nella mailing list?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___--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] Modifying SIP Stack
Hello list, I am trying to include a new message after I receive a Register in chan_sip, at the beginning I would like to forward the same message to a fixed IP address, I have seen that fileds like p-sa.sin_addr and p-sin.sin_addr have to be with the IP address, but I am not sure about how to force these fields to be the destination that I want (192.168.1.10)... somebody knows how to do it? do you know where can I fond information about the p structure in asterisk and how to modify it manually? Thanks and Rgds, German ___ --Bandwidth 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: Digium on Dell PowerEdge 1850
Like Aaron, our asterisk systems are on Dell servers and even some Dell optiplex systems for small offices. However we use Sangoma cards to skirt compatibility issues.On 10/19/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Tomislav,I use Dell hardware for desktops only. Each time I tried to use a Dellpc with telephony cards I get problems. It works only with a TDM400 butif u plan to add something more it is a real nightmare! Giorgio IncantalupoTomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... We're running 2 TE412P's in a Dell 1850 just fine, been running like this for well around 6 months to a year now without any problems. They're not exactly 212P's but I imagine it won't be much different. On Wed, 2006-10-18 at 10:54 +0200, Tomislav ParÄŤina wrote: Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Hi Aaron! Thank you for your mail. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while
Robert La Ferla wrote: I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the analog line on the FXO, * dials all the extensions (SIP and analog.) I have a Digium card with 1 FXO and 1 FXS. Do you have callprogress=yes or busydetect=yes in your /etc/asterisk/zapata.conf ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX softphones
Guillermo Salas M. a écrit : On Wed, 2006-10-18 at 20:08 +0200, Francesco Peeters (Asterisk) wrote: On Wed, October 18, 2006 19:03, Paul Gaffney wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil Neil, www.asteriskguru.com http://www.asteriskguru.com/ lists a few of them. Try IDEFISK. Paul Gaffney LANStatus,LLC I personally like DIAX on for Windows users. Haven't yet found an IAX phone I like on Linux... Kiax works great with Gnome, KDE or Xfce. There is also a windows version. Very nice piece of software. -- Daniel ___ --Bandwidth 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] Access Denied on a Windows share
Is it an NTFS Share?On 10/19/06, Paul Gaffney [EMAIL PROTECTED] wrote: Message: 12 Date: Tue, 17 Oct 2006 18:07:04 -0700 (PDT) From: sdgesa gaeharth [EMAIL PROTECTED] Subject: Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 None of these steps have made a difference. Any other suggestions? Here is my original post: Can anyone help me to figure out why I can not write to a public share? I was able to join the domain without a problem. I can access the share from an xp box. I just can not write: Access denied. thanks If the person or process that is trying to write to the share is a member of a group that is denied access then the write will fail with "access denied". Check the "effective permissions" for a user on the file and look at the server security logs which may give you some additional information. Paul Gaffney LANStatus, LLC ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth 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] Digium Single Span Card Installation
On Thu, Oct 19, 2006 at 10:15:32AM +0530, K Y Iyer wrote: Hello Again I ran the genzaptelconf - but I cannot see any channels in asterisk CLI. I did make and install zaptel after I put in the card. What am I missing? zaptel.conf is now configured (hopefully with the correct signalling, if not, please let me know). genzaptelconf does not generate a complete /etc/asterisk/zapata.conf file. Rather, it generates /etc/asterisk/zapata-channels.conf which is a zapata.conf snippet that could be #include-d at the end of zapata.conf: echo '#include zapata-channels.conf' /etc/asterisk/zapata.conf Then you need to restart asterisk (or use 'zap restart' , if availble) Thanks very much, indeed, for all your assistance Best wishes Iyer [03 ~]# asterisk -r Asterisk 1.2.12.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Connected to Asterisk 1.2.12.1 currently running on mysql-03 (pid = 23271) 03*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault 03*CLI zap show status Description Alarms IRQbpviol CRC4 Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 03*CLI [03 ~]# ztcfg -vvv Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 133-266 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, October 18, 2006 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium Single Span Card Installation On Wed, Oct 18, 2006 at 04:28:16PM +0530, K Y Iyer wrote: Hi Also tried the following command and got the following lines. Why is that /proc/zaptel/1 reports all the 31 lines whereas ztcfg reports 18 channels? What do they mean? ztcfg reports the channels you are about to try to configure and the result of that configuration. It does not scan your system for channels. Your /etc/zaptel.conf does not include all the bchan-nnels, and thus not all of them are reported and not all were configured. selfpromo If you want to scan your system for channels and attempt to create a wonrking configuration (one that at least be able to pass ztcfg and run asterisk with), use xpp/utils/genzaptelconf. /selfpromo -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update
[asterisk-users] siemens hipath interoperability - PRI/Q.SIG - card recommendation
Hello, if somebody using this scenario in production successfully, please send me info, which ISDN card for asterisk server is usefull for me (Digium, Sangoma)? my crucial requirement is caller id name transfer/display between ISDN (Siemens PBX) and IP phone connected to asterisk I'm using PRI interface and Q.SIG signaling. thank you PJ ___ --Bandwidth 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: 1.4 on mac OSX 10.4.8
I just built 1.4.0 beta 3 on OS X 10.4.8, and it went pretty smoothly. I didn't need to install wget. Asterisk starts and runs with 0% CPU. The CLI also works, but hangs if I try to tab-complete commands. However, that might be because I don't have any working config files and/or have some legacy modules installed from 1.2. Now if only I could get app_conference to build, Asterisk would do everything I need. - .Dustin On Oct 18, 2006, at 3:20 AM, Martin Joseph wrote: On 2006-10-17 14:19:00 -0700, Daniel Salama [EMAIL PROTECTED] said: You can get wget for OSX from DarwinPorts (http:// wget.darwinports.com/) Ok, I bit the bullet and build wget. This allows me to build 1.4 branch, which does the same thing as 1.40b2. It starts up, consumes as much CPU as is available, and is not responsive to CLI commands or registrations. It's a dead duck. Anybody out there trying this stuff on OSX? 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 ___ --Bandwidth 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] rxfax problem
Did you ever get an answer to this problem ? I too am seeing this and its driving me mad !!! Jim ___ --Bandwidth 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] Occasional one-way audio - Sangoma A101
We are having an occasional one w-way audio problem that occurs about every 25 - 30 calls on a system configured as follows: Asterisk 1.2.12.1 Sangoma A101 w/wanpipe beta9 Polycom 500s w 1.5.3 This happens only on inbound calls from the PRI. The external caller can hear our customer answer and say hello, however, our customer cannot here their caller. Typically, the caller calls right back and all is fine. There is no discernable pattern as I can tell. Anyone have, and hopefully fix, a similar issue? Thanks, Mike Clark ___ --Bandwidth 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: 1.4 on mac OSX 10.4.8
I'm a Certified Apple Sys Admin - lots of experience with Macs and Mac servers. However, when setting up an asterisk server, I'm still thinking a Dell box with linux is the best direction - to get the full reliability and full support of this group. Am I mistaken? Or is using a Mac box just as convenient and reliable? Or is traditional linux 'strongly' recommended for asterisk? I'm looking at a solely IP based system - no digium cards thanks Todd ___ --Bandwidth 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] Occasional one-way audio - Sangoma A101
Hi Mike, Sounds like you're having about the same problem Giorgio and I are having. I'd be really surprised if you don't start having the same problem from SIP-SIP calls to. I also have a Sangoma card, and originally thought it was only on calls coming from a PRI. But as time has moved forward, the issue really appears to be between the Polycoms and Asterisk. The next time it happens, try hitting a digit (like 5) on the polycom and see if the audio becomes available. BTW - our other discussion on this is called random one way audio and noise between SIP phoneson same LAN - Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Thursday, October 19, 2006 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Occasional one-way audio - Sangoma A101 We are having an occasional one w-way audio problem that occurs about every 25 - 30 calls on a system configured as follows: Asterisk 1.2.12.1 Sangoma A101 w/wanpipe beta9 Polycom 500s w 1.5.3 This happens only on inbound calls from the PRI. The external caller can hear our customer answer and say hello, however, our customer cannot here their caller. Typically, the caller calls right back and all is fine. There is no discernable pattern as I can tell. Anyone have, and hopefully fix, a similar issue? Thanks, Mike Clark ___ --Bandwidth 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] How do I configure Asterisk if I need to run Mysql server on second Linux
Hi List: Please someone help me to point out where I can get the idea to configure Asterisk for mysql server running on different Linux. Many thanks, Tielin ___ --Bandwidth 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] Occasional one-way audio - Sangoma A101
We use almost all Polycoms, several hundred had one way audio with 1.6.4 or 5, forget which 1.66 and 2.01 seem to be ok We did have a few phones (2-3) that had random one way for a long time, replaced everything feeding them and it still happend. A month ago I replaced the phones and have not had a complaint since then. On Oct 19, 2006, at 11:32 AM, Scott Scecina wrote: Hi Mike, Sounds like you're having about the same problem Giorgio and I are having. I'd be really surprised if you don't start having the same problem from SIP-SIP calls to. I also have a Sangoma card, and originally thought it was only on calls coming from a PRI. But as time has moved forward, the issue really appears to be between the Polycoms and Asterisk. The next time it happens, try hitting a digit (like 5) on the polycom and see if the audio becomes available. BTW - our other discussion on this is called random one way audio and noise between SIP phoneson same LAN - Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Thursday, October 19, 2006 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Occasional one-way audio - Sangoma A101 We are having an occasional one w-way audio problem that occurs about every 25 - 30 calls on a system configured as follows: Asterisk 1.2.12.1 Sangoma A101 w/wanpipe beta9 Polycom 500s w 1.5.3 This happens only on inbound calls from the PRI. The external caller can hear our customer answer and say hello, however, our customer cannot here their caller. Typically, the caller calls right back and all is fine. There is no discernable pattern as I can tell. Anyone have, and hopefully fix, a similar issue? Thanks, Mike Clark ___ --Bandwidth 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] Errors in console in every call made when using 1.4b3
When using Asterisk 1.4b3, everytime I make a call I get continuously (around 20 times a second) error messages like the following upon the call connecting (or getting to ring): [Oct 19 17:23:22] WARNING[28682]: channel.c:767 ast_queue_frame: Unable to write to alert pipe on IAX2/gradwell-7, frametype/subclass 2/256 (qlen = 96): Bad file descriptor! [Oct 19 17:23:29] WARNING[28685]: chan_iax2.c:1834 __attempt_transmit: Max retries exceeded to host 84.9.159.76 on IAX2/acron-9 (type = 6, subclass = 1, ts=7, seqno=0) [Oct 19 17:26:14] WARNING[31424]: channel.c:767 ast_queue_frame: Unable to write to alert pipe on IAX2/office-1, frametype/subclass 2/256 (qlen = 21): Bad file descriptor! This did not occur with 1.4b2, Is anyone else experiencing this? ___ --Bandwidth 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 not detecting Tormenta2 PRI Interface card
On Thu, Oct 19, 2006 at 05:38:46AM -0400, Zeeshan Zakaria wrote: Hi, After upgrading to new version of Asterisk and Zaptel, my Tormenta2 card has stopped working. Zaptel doesn't detect it. What should I do to make it work again. It is enabled in /etc/sysconfig/zaptel. Where else I have to enable it so zaptel can detect it and make it work. zaptel.conf and zapata.conf are the same as I had them before. I tried googling but couldn't find any help. Is the module loaded? lsmod | grep ^zaptel Has it detected a span? cat /proc/zaptel/* What do you have on /etc/zaptel.conf and on /etc/asterisk/zapata-channels.conf ? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modifying SIP Stack
I don't have access to the sip code right now, but from past network code I've writen you could try this: inet_aton(192.168.1.10, p-sin.sin_addr); If they are just wrapping the struct sockaddr_in as sin in p. Worth a try... Ryan Hello list, I am trying to include a new message after I receive a Register in chan_sip, at the beginning I would like to forward the same message to a fixed IP address, I have seen that fileds like p-sa.sin_addr and p-sin.sin_addr have to be with the IP address, but I am not sure about how to force these fields to be the destination that I want (192.168.1.10)... somebody knows how to do it? do you know where can I fond information about the p structure in asterisk and how to modify it manually? Thanks and Rgds, German ___ --Bandwidth 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] 3way calling / codec problem
Right - I get the error on the console - I just can't tell how many transcodes are occuring at any given point in time... On 10/18/06, Thomas Kenyon [EMAIL PROTECTED] wrote: Mr. Jones wrote: Is there some way I can tell? On 10/16/06, Thomas Kenyon [EMAIL PROTECTED] wrote: Mr. Jones wrote: I'm having problems with conference calls (3-way) when I have my codec forced to g729 in sip.conf. I'm using Grandstream 2000s. If enable both g711 and g729 then 3 way calling and transfers work. I'm not sure why this would matter? Here's the error: Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs! Any help is greatly appreciated! Are you out of licences? From memory when in a console each channel needs to be able to be transcoded to SLIN. (where it is mixed and transcoded back again). I meant conference (not console). You can show g729 or have a console open with verbosity set (probably to 3) and it should tell you on the console output (usually several times). ___ --Bandwidth 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] say Asterisk to answer
The latest X-lite version has autoanswer button on the front.. marked AA Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi Greg, Idefisk support Auto-answer only in a biz version I suppose you got free version.. You will find more details http://www.asteriskguru.com/idefisk/free/ Cheers, Giovanni 2006/10/19, Gregory Duchatelet [EMAIL PROTECTED]: Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on Accept button. Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth 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] Re: 1.4 on mac OSX 10.4.8
Our company's PBX is running Asterisk 1.2 under OS X Server and it's been pretty reliable for the 50+ extensions that we have. The system has an uptime of 245 days, and no one has ever reported dropped calls or any other disturbing behavior. The reason we're using OS X instead of Linux is that it's much easier to administer in our environment. We have a data center full of xserves, and it would be a pain to have one oddball Linux box worry about. At the very least, if you started off with a xeon-based xserve and OS X didn't work out, you could always just install Linux on it. The only glaring hole in our implementation is that we haven't found a good conferencing solution (no timing source = no meetme). Right now, we are paying a fee for a third-party service to handle that for us. It would be nice to get an Asterisk module working for this, or possibly some sort of standalone SIP conferencing server. There are some other issues that are not a big problem for us; last I checked, realtime was not working for OS X ...and you're out of luck if you want to buy Digium hardware since they haven't developed any OS X drivers. - .Dustin On Oct 19, 2006, at 11:30 AM, Todd- Asterisk wrote: I'm a Certified Apple Sys Admin - lots of experience with Macs and Mac servers. However, when setting up an asterisk server, I'm still thinking a Dell box with linux is the best direction - to get the full reliability and full support of this group. Am I mistaken? Or is using a Mac box just as convenient and reliable? Or is traditional linux 'strongly' recommended for asterisk? I'm looking at a solely IP based system - no digium cards thanks Todd ___ --Bandwidth 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] plainvoip - down ???
Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails -- #Joseph ___ --Bandwidth 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] plainvoip - down ???
Joseph wrote: Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails I can get there just fine. Your routes might be toasted [EMAIL PROTECTED] ~]# ping -c 10 plainvoip.com PING plainvoip.com (66.199.240.2) 56(84) bytes of data. ... --- plainvoip.com ping statistics --- 10 packets transmitted, 10 received, 0% packet loss, time 9013ms rtt min/avg/max/mdev = 75.531/78.550/80.349/1.418 ms, pipe 2 Depending on your location thought, there are issues with GBLX possibly due to a fiber cut either in VA or DC. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I configure Asterisk if I need to run Mysql server on second Linux
If you are using MySQL for storing CDR's, this is what I use (slightly modified, of course): cat /etc/asterisk/cdr_mysql.conf ; [global] dbname = hostname= password= port= 3306 sock= /tmp/mysql.sock table = cdrs user= userfield = 1 On Thu, 19 Oct 2006, Tielin Xu wrote: Hi List: Please someone help me to point out where I can get the idea to configure Asterisk for mysql server running on different Linux. Many thanks, Tielin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] plainvoip - down ???
Joseph wrote: Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails Since when is the asterisk-users list second level support for VoIP providers? If they are down, I am sure they are well aware of it. Jeremy McNamara ___ --Bandwidth 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] T1 pricing in Oz
Im looking at getting a T1 into a location in Melbourne, Australia and was wondering if anyone has a good source and pricing for this. Cheers, 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] T1 pricing in Oz
On Thursday 19 October 2006 14:43, Forum wrote: I'm looking at getting a T1 into a location in Melbourne, Australia and was wondering if anyone has a good source and pricing for this. I think your looking for a E1 Australia follows the European standard last i looked. I never did any voice stuff when i lived in Oz. but you could try Telstra, Optus, etc, or look in the phone book for a reseller. Someone probably has alot better experience than me. Dennis ___ --Bandwidth 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] Embedded Asterisk
I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Thanks Cory Andrews ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 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] Embedded Asterisk
Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Thanks Cory Andrews ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory Cory, Brian Capouch is very interested in running Asterisk on Mipsel hardware like the Linksys WRT, etc. His latest favorite is the Netgear WGT634U (it has a USB port for extra storage and a miniPCI slot for a wifi radio). It's also dirt cheap (just the way he likes it). Unfortunately, Netgear has discontinued this model (about a year ago - I think), so you better get them while they're hot! Brian will be helping people flash the the Netgear's with the latest version of the firmware + Asterisk. He is very dedicated to reading the list, I'm sure he'll get back to you too. -- Kristian Kielhofner ___ --Bandwidth 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 / Silence issues
I am now running 1.4 beta3 I have an ongoing issue that it does not recognize my DTMF key press. I will call and press as many numbers and the background message still plays. I am also having an issue with transfers NOTICE[30930]: chan_sip.c:13289 handle_request_invite: Unable to create/find SIP channel for this INVITE happens everytime Any ideas. I tried to go back to 1.2 and the modules would not show up. Thanks Jason ___ --Bandwidth 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] considering purchasing a t1 card, any recommendations?
CanI now 5th it ? All this makes me wonder why Digium dosent work harder. I have mainly only seen others praise Sangoma over Digium. - Original Message - From: Tom Vile To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, October 18, 2006 4:22 PM Subject: Re: [asterisk-users] considering purchasing a t1 card,any recommendations? I 4th it. On 10/18/06, Matthew Thompson [EMAIL PROTECTED] wrote: On 17 Oct 2006, at 22:09, Richard wrote: I would have to second the Sangoma buy. Their tech support is second to none and more then helpful. I've never had any problems with their products that wasn't my own fault. Thirded - I've just done another install with a Sangoma A102 - the setup guides you through all the way and takes no more than 30 minutes (Including recompiling zaptel, which it does for you) [EMAIL PROTECTED] :o) -- Matthew Thompson [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Fax: 518-631-2856 ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Findme problem
I have a backup of a working version on my server
some where. When i find it I will post it.
Dovid
- Original Message -
From:
Eric
Jacksch
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 18, 2006 4:20
PM
Subject: [asterisk-users] Findme
problem
Greetings all,
I've been working on having Asterisk put a call
through to two different numbers, and give the call to the first one that
acknowledges by pressing the 1 key. I found an example on the wiki, but
I can't get it working.
When I answer the call I hear the message telling
me to press 1 to connect, and as soon as the message is done, the call is
connected. In other words, it is not waiting for me to press a
key.
I'm sure this is a forehead slapper, but I just
can't see it...can anyone help? Here's the relevant portion of the
dialplan, It executes the NoOp(Waiting) and then the macro seems to
immediately exit and the call is connected.
[default]exten =
_XX,1,Dial(SIP/provider/${EXTEN:4},40,M(screen))exten =
_XX,2,Hangup
[macro-screen]exten =
s,1,Wait(1)exten = s,2,Set(TIMEOUT(digit)=5)exten =
s,3,Set(TIMEOUT(response)=10)exten = s,4,Background(press-1)exten
= s,5,NoOp(Waiting)
exten = 1,1,NoOp(Caller
accepted)
exten = i,1,NoOp(Invalid
response)
exten =
i,2,Set(MACRO_RESULT=CONTINUE)
exten = t,1,NoOp(Timeout)exten =
t,2,Set(MACRO_RESULT=CONTINUE)
[find-eric]exten =
s,1,Playback(pls-wait-connect-call)exten =
s,n,Dial(LOCAL/6135551212LOCAL/6135551313,40,m)
(I have replaced the phone numbers with bogus
ones).
Thanks,
Eric
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Re: [asterisk-users] Findme problem
Ooops. I read the email wrong. The macro I created called one number. If the person didnt accept the call or if they didnt pick up then it tried the second person. Let me know if you still want it. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] siemens hipath interoperability - PRI/Q.SIG - cardrecommendation
Hi, we have tested the Digium-Cards, they work fine, but don't expect to much! Only segmentation 1 in Ecma (it is not a digium-problem) The Name ist displayed, but only in Hex-Code (this is due to the Libpri/Zaptel Drivers but I didn't fint a way to display it in *) There is also very less documentation, on Asterisk.org (Features) there is non Q.Sig Support offered. Also very less documentation through google available. ;-( If you find some hints, i'm also interested! Regards wendy - Original Message - From: Pavel Jezek [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, October 19, 2006 5:22 PM Subject: [asterisk-users] siemens hipath interoperability - PRI/Q.SIG - cardrecommendation Hello, if somebody using this scenario in production successfully, please send me info, which ISDN card for asterisk server is usefull for me (Digium, Sangoma)? my crucial requirement is caller id name transfer/display between ISDN (Siemens PBX) and IP phone connected to asterisk I'm using PRI interface and Q.SIG signaling. thank you PJ ___ --Bandwidth 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] considering purchasing a t1 card, any recommendations?
On Thu, 2006-10-19 at 22:39 +0200, Dovid B wrote: Can I now 5th it ? All this makes me wonder why Digium dosent work harder. I have mainly only seen others praise Sangoma over Digium. I strongly suspect digium is painfully aware of the problems with some combinations of mboards and their cards, but given limited resources, what would you focus development efforts on? Creating echo-cancellers and bri cards or designing a PCI interface that works even with the most broken motherboard chipset? Most people who have digium cards working seem to have them working extremely well and some can't get them to work at all. Frankly, I have seen so many motherboards that I consider outright broken, I wouldn't blame a digium card for not working in any of those. In fact, those motherboards tend to have problems with most cards, it just happens to show much more when you try and do voice traffic (e.g. isdn) rather than say IP traffic. I'm not qualified to judge digium vs sangoma isdn facing interfaces, but I'd be interested how that compares, if someone can shed some light on that, I'd read it with interest. E.g., how do digium cards perform on substandard isdn lines? How do they handle faults on the line? How compatible (compared to sangoma) are they with different exchange equipment? Last but not least, might that be why digium secured some funding recently, to increase the resources for development on these cards? just my 2 pence worth ;) Conrad ___ --Bandwidth 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] Occasional one-way audio - Sangoma A101
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Thursday, October 19, 2006 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Occasional one-way audio - Sangoma A101 We are having an occasional one w-way audio problem that occurs about every 25 - 30 calls on a system configured as follows: Asterisk 1.2.12.1 Sangoma A101 w/wanpipe beta9 Polycom 500s w 1.5.3 This happens only on inbound calls from the PRI. The external caller can hear our customer answer and say hello, however, our customer cannot here their caller. Typically, the caller calls right back and all is fine. There is no discernable pattern as I can tell. Anyone have, and hopefully fix, a similar issue? Thanks, Mike Clark Scott Scecina wrote: Hi Mike, Sounds like you're having about the same problem Giorgio and I are having. I'd be really surprised if you don't start having the same problem from SIP-SIP calls to. I also have a Sangoma card, and originally thought it was only on calls coming from a PRI. But as time has moved forward, the issue really appears to be between the Polycoms and Asterisk. The next time it happens, try hitting a digit (like 5) on the polycom and see if the audio becomes available. BTW - our other discussion on this is called random one way audio and noise between SIP phoneson same LAN - Scott So, do newer versions of Polycom firmware, like 1.6.7, help? I saw Asterisk 1.2.13 is now available. Have you tried that. The info on the Asterisk site says numerous bugs have been fixed. Mike ___ --Bandwidth 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] T1 pricing in Oz
Thanks Dennis, I am an Australian living in Canada and like yourself have done no voice work back home -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dennis Gilmore Sent: Thursday, October 19, 2006 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 pricing in Oz On Thursday 19 October 2006 14:43, Forum wrote: I'm looking at getting a T1 into a location in Melbourne, Australia and was wondering if anyone has a good source and pricing for this. I think your looking for a E1 Australia follows the European standard last i looked. I never did any voice stuff when i lived in Oz. but you could try Telstra, Optus, etc, or look in the phone book for a reseller. Someone probably has alot better experience than me. Dennis ___ --Bandwidth 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] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
Bristuff has been updated; http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz -- Vidar ___ --Bandwidth 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 logging
If I set the logging.conf to log DTMF it only seems to log dtmf messages that are bridged through the * server. If the call goes into a menu the DTMF dont get logged. Is the intended behavior? Scott England ___ --Bandwidth 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] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
On 23:04, Thu 19 Oct 06, Vidar wrote: Bristuff has been updated; http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz Thanks for the information. It's a shame we need to read this here and not see it on their website. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Occasional one-way audio - Sangoma A101
Mike, I'm not able to download the newer polycom software(s) 'cause I'm not certified in anything. I just saw the announcement on the Asterisk updates today. I'll get them installed tonight... - Scott Scott Scecina wrote: Hi Mike, Sounds like you're having about the same problem Giorgio and I are having. I'd be really surprised if you don't start having the same problem from SIP-SIP calls to. I also have a Sangoma card, and originally thought it was only on calls coming from a PRI. But as time has moved forward, the issue really appears to be between the Polycoms and Asterisk. The next time it happens, try hitting a digit (like 5) on the polycom and see if the audio becomes available. BTW - our other discussion on this is called random one way audio and noise between SIP phoneson same LAN - Scott So, do newer versions of Polycom firmware, like 1.6.7, help? I saw Asterisk 1.2.13 is now available. Have you tried that. The info on the Asterisk site says numerous bugs have been fixed. Mike ___ --Bandwidth 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] considering purchasing a t1 card, any recommendations?
I think the recent Digium and Sangoma cards are quite similar. (and about the same price) I didn't try sangoma so far, never had any issues with the digium cards, I have no clue how the digium helpdesk is, i never needed to call them. (well not really correct i did call them once, years ago for a firmware problem with their first te410p revision, causing a crash once every few months they had the distributor send me replacement cards right away, before i returned the old ones, so that i could swap them without having to shut down the server for a week). Configuration and installation for the cards is pretty straightforward, all you need to do is compile the kernel modules for your kernel. I personally installed at least 20 digium pri cards, all on different hardware without problems related to the digium hardware. (sometimes i did have bad cables, bad pri's, oh and my embedded pc didn't provide enough power for FXO ports). You will probably find more people on the list with problems with digium than people with problems with sangoma. This might be because a lot more people seem to use the digium cards with asterisk than sangoma cards with asterisk. (Based on the people i speak to, i'd guess 1 to 5% use sangoma?). The biggest choice you need to make is if you want onboard echo cancellation or not, you might not need it and if you want it its going to cost you a lot more than without. (both for sangoma and digium hardware). - They both seem to use exactly the same Octasic echo cancellation module. If you need on board echo cancellation but don't need 4 ports, digium is the only choice with their 2 port card with Octasic echo cancellation module. (Afaik sangoma doesn't have such a 2 port board with on board E.C. but i could be wrong.) Btw, there are more options, dialogic has compatible cards and so does eicon. (you will need deeper pockets though, the eicon retails at +/- 12000 euro for a quad span i think - people who buy these for asterisk usually do so for hardware faxing or interconnection to different carriers at the same time.) Some people prefer digium over sangoma because they sponsor the asterisk development that way. I'm not one of them, i buy digium cards (or tell my customers to buy them) because i'm happy with their product. Dislaimer: I know some of the people within Digium quite well, so maybe i get exceptional support or they ship me handpicked gold plated, overclocked versions of their cards (not really since i just buy them from a reseller). Cheers, Zoa. Dovid B wrote: Can I now 5th it ? All this makes me wonder why Digium dosent work harder. I have mainly only seen others praise Sangoma over Digium. - Original Message - *From:* Tom Vile mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Wednesday, October 18, 2006 4:22 PM *Subject:* Re: [asterisk-users] considering purchasing a t1 card,any recommendations? I 4th it. On 10/18/06, *Matthew Thompson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 17 Oct 2006, at 22:09, Richard wrote: I would have to second the Sangoma buy. Their tech support is second to none and more then helpful. I've never had any problems with their products that wasn't my own fault. Thirded - I've just done another install with a Sangoma A102 - the setup guides you through all the way and takes no more than 30 minutes (Including recompiling zaptel, which it does for you) [EMAIL PROTECTED] :o) -- Matthew Thompson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth 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:
[asterisk-users] PRI Card
We are looking at migrating our office from a Samsung PBX to an Asterisk PBX. I am looking at ordering a PRI with 12 Channels for now (we currently have 8 analog lines) and need to know what PRI card you guys would recommend that we use. I have seen some with Echo Cancellation and so on, but don't know which one would be best to get. K ___ --Bandwidth 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] Electric usage of a tdm400p
Further, I should have mentioned my resolution. As Bob mentioned, upgrade the power supply now to give you options and peace of mind in the future. I am using a picoPSU-120 in the aforementioned itx box, for example, http://www.mini-box.com/s.nl/it.A/id.417/.f?category=13 and have been quite happy with it :) Moj Bob Chiodini wrote: Erick, It looks like the 2.5 laptop drive requires 5 watts to spin up. Adding that to the 15 watts for the Digium card, leaves about 40 watts available for the MB. It's unlikely that the system will be producing ring voltages when the drive is spinning up. It depends on how conservative you may be with real-time power management, e.g. spinning drives down when not in use, etc. I did not easily find too many ITX MB power requirements, but the one I did find, required 45 watts (peak). In the worst case, ringing 5 RENs on each Digium port and spinning up the disk, you would be overtaxing the power supply. I doubt you will have 5 RENs on each port and all ringing, but you could. In ages past, hard drives were the most vulnerable to poor power regulation, but that may have changed. With the higher cost of 2.5 drives, I would not take any chances. Beefing up the power supply would also eliminate the need for manually managing power should you need a CDROM or more power hungry drive in the future. It's also one less concern when troubleshooting the system. As Moj has pointed out, problems can occur when working close to the edge. I, too, have experienced similar problems when power was limited and have had to, temporarily, resort to a bigger power supply to get a system installed. Then fell back to a smaller one in operation. Good luck. Bob... On Wed, 2006-10-18 at 08:49 -0800, Mojo with Horan Company, LLC wrote: I set up a similar system on an VIA Epia 5000, and I had issues when I included the CDROM in the mix. I had to use another ATX power supply to complete the install, but then once I removed the CDROM drive I had no power issues. I presume you could install the OS with the CDROM drive installed and the molex power connector REMOVED from the TDM card, then when the OS was installed and you had network connectivity, power down, remove the CDROM, add the power supply for the TDM card, then install zaptel etc. Or just try it and tell us what happens, low power won't break it in my experience. Your cdrom drive might have a lower power consumption than mine. Moj Erick Perez wrote: Well Im planning to use a mini-itx, a laptop hdd and a 4fxs digium card. the mini-itx comes with a 60W DC to DC adapter (80W peak). So I need power to manage the hdd, motherboard,the tdm card. A disk cable can be made available, but is not present as a factory default. So My real concern is power. On 10/18/06, Bob Chiodini [EMAIL PROTECTED] wrote: On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote: Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? Erick, Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN ~5). This translates to 2.7 watts. Assuming a DC/DC converter efficiency of 38% (probably low), you would need about 3.7 watts, per FXS module. About 15 watts, total. What is the TDM card installed in and is a disk drive cable available? Bob... ___ --Bandwidth 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 !DSPAM:500,453763ec326532002735277! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] plainvoip - down ???
Joseph wrote: Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails This is starting to sound like a rerun of Livevoip. Remember that company? ___ --Bandwidth 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] Embedded Asterisk
On Thu, 19 Oct 2006 16:10:23 -0400, Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Cory, I've been running Astlinux on an H-P T5700 thin client. These are available, not dirt cheap, but I received box of them that were due to be recycled. I see them on Ebay for around $80-150. That's less than a Soekris Net4801 new in a case. They have the advantage of some built-in flash in an IDE DOM card, then 4 USB ports as well as traditional mouse, keyboard, VGA ports. Also, the CPUs which are in the 800 MHz - 1.2 GHz range handle G.729 codecs better than the Net4801 or WRAP boards. Michael Graves ___ --Bandwidth 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] considering purchasing a t1 card, any recommendations?
Conrad Wood wrote: On Thu, 2006-10-19 at 22:39 +0200, Dovid B wrote: Can I now 5th it ? All this makes me wonder why Digium dosent work harder. I have mainly only seen others praise Sangoma over Digium. I strongly suspect digium is painfully aware of the problems with some combinations of mboards and their cards, but given limited resources, what would you focus development efforts on? Creating echo-cancellers and bri cards or designing a PCI interface that works even with the most broken motherboard chipset? Sangoma seems to accomplish that quite well. In fact they clearly state their cards will work Digium says try another motherboard That answer is an insult Especially when the same motherboard/machine/software works with the Sangoma Their answer really should be buy a Sangoma Perhaps their efforts would be better spent as a Sangoma reseller. Just my opinion. Worth what you paid for it John Novack ___ --Bandwidth 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] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
On Thu, Oct 19, 2006 at 11:27:07PM +0200, Michiel van Baak wrote: On 23:04, Thu 19 Oct 06, Vidar wrote: Bristuff has been updated; http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz Thanks for the information. It's a shame we need to read this here and not see it on their website. Also note that the changelog entry for 0.3.0-PRE-1u is missing from the CHANGES file. Nevertheless, that is a version for Asterisk 1.2.13 , Zaptel 1.2.10 and libpri 1.2.4 . -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FS: Sangoma A200 10 port FXO card
Hi folks, I have a Sangoma A200 10 port FXO card for sale. US$500 secures plus shipping. Thanks 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
Re: [asterisk-users] question about CDR command
Thanks!! Just one more question. Can I do the same add fieldname=1 if I add a field fieldname in the cdr table to perform the same action? On 10/19/06, William Piper [EMAIL PROTECTED] wrote: In cdr_mysql.conf add userfield=1 under the globals setting. bp On 10/18/06, unplug [EMAIL PROTECTED] wrote: I want to set some custom data in the field of userfield in table CDR as following. exten = s,19,Set(CDR(userfield)=1234) exten = s,20,Dial(SIP/1234) However, the userfield doesn't get update after making the call. After that, I relocate the command as following. exten = s,19,Dial(SIP/1234) exten = s,20,Set(CDR(userfield)=1234) The userfield doens't get update at all. I don't know why the field can't update after issuing the command. Anyone can help? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] T1 pricing in Oz
E1 is readily available in Australia, and very easy to set up. later, PaulH On Thu, 2006-10-19 at 15:05 -0500, Dennis Gilmore wrote: On Thursday 19 October 2006 14:43, Forum wrote: I'm looking at getting a T1 into a location in Melbourne, Australia and was wondering if anyone has a good source and pricing for this. I think your looking for a E1 Australia follows the European standard last i looked. I never did any voice stuff when i lived in Oz. but you could try Telstra, Optus, etc, or look in the phone book for a reseller. Someone probably has alot better experience than me. Dennis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk to asterisk DID extentions
On Fri, 2006-10-13 at 01:55 +1000, Matt wrote:
Avi Miller wrote:
On 04/10/2006, at 1:55 AM, Matt wrote:
How can I make * aware of the other ext on the remote box so the DID
caller can access them like he can with the local box?
On each box, define the other range:
Box A:
exten = _9XX,1,Dial(IAX2/BoxB/${EXTEN})
Box B:
exten = _8XX,1,Dial(IAX2/BoxA/${EXTEN})
Thanks Avi but it didnt work. It still says not a valid extention
from the indial :(
-Matt
Is IAX working fine between the boxes?
Or is this a dial-in issue (8XXX,1,Dial(IAX2/BoxA/${EXTEN:5}) ???
later,
PaulH
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Re: [asterisk-users] T1 pricing in Oz
Dennis, I work for a company that can provide E1's in a secure data center in Melbourne CBD. Contact me off list for a quote etc. Paul Hales wrote: E1 is readily available in Australia, and very easy to set up. later, PaulH On Thu, 2006-10-19 at 15:05 -0500, Dennis Gilmore wrote: On Thursday 19 October 2006 14:43, Forum wrote: I'm looking at getting a T1 into a location in Melbourne, Australia and was wondering if anyone has a good source and pricing for this. I think your looking for a E1 Australia follows the European standard last i looked. I never did any voice stuff when i lived in Oz. but you could try Telstra, Optus, etc, or look in the phone book for a reseller. Someone probably has alot better experience than me. Dennis ___ --Bandwidth 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: 1.4 on mac OSX 10.4.8
On 2006-10-19 08:51:01 -0700, Dustin Wenz [EMAIL PROTECTED] said: I just built 1.4.0 beta 3 on OS X 10.4.8, and it went pretty smoothly. I didn't need to install wget. Asterisk starts and runs with 0% CPU. The CLI also works, but hangs if I try to tab-complete commands. However, that might be because I don't have any working config files and/or have some legacy modules installed from 1.2. Now if only I could get app_conference to build, Asterisk would do everything I need. ok, I don't really get that, unless the make file was changed... When you say it runs with 0% of the CPU, you mean it's not eating the whole CPU right? Mine is 0% available. I did kill all the warned about modules... So I am not clear why I have no joy with 1.4b3. I am going to keep building 1.4 branch for a while and see if it resolves itself, This was a fully working 1.2.12 box that had been a spare for me. Oh well... Please do let us know if you configure it and are able to accepts registrations and make it work? Thanks, 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] Re: 1.4 on mac OSX 10.4.8
On 2006-10-19 09:30:14 -0700, Todd- Asterisk [EMAIL PROTECTED] said: I'm a Certified Apple Sys Admin - lots of experience with Macs and Mac servers. However, when setting up an asterisk server, I'm still thinking a Dell box with linux is the best direction - to get the full reliability and full support of this group. The support thing is the strongest argument above. Am I mistaken? Or is using a Mac box just as convenient and reliable? My old imac G3/400 has been chugging along for months without any issues. It doesn't handle a lot of calls (about 50 a day), but it doesn't have any problems either. It has about 6 extensions, some SIP and some IAX2 as well as PSTN gateway (wellgate 3701a). I have some echo across the gateway (I have a long loop) but that resolves within second of a call begining and seems to be a standard issue around these parts (mac or Linux). I also have 3 IAX call terminators for US 48 AND Canada, as well as one inbound DID (just for testing really). Or is traditional linux 'strongly' recommended for asterisk? I'm looking at a solely IP based system - no digium cards If you don't need Zaptel for hardware support, it seems to me that Asterisk 1.09-1.2.12 has been solid on OSX (up till 1.4). Plus, if you buy Apple hardware new, it's firkin' intel hardware anyhow, so you could always boot it as a linux box if need be ;~) ___ --Bandwidth 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] question about asterisk txFAX
I have installed the libtiff(3.5.7),spandsp-0.0.2pre24,app_txfax and
app_rxfax,asterisk-1.2.12.1 on the CentOS 4.2.
I set sip.conf like this:
[sip_local]
host=192.168.2.111
type=friend
dtmfmode=rfc2833
canreinvite=no
insecure=very
I set extensions.conf like this:
[test]
exten=_9001,1,Set(CALLERID=${EXTEN})
exten=_9001,n,dial(SIP/[EMAIL PROTECTED])
exten=_9001,n,hangup
[fax]
exten=_9002,n,SetVar(FAXFILE=/var/spool/asterisk/fax/test.tif)
exten=_9002,n,SetVar(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLEEID})
exten=_9002,n,txfax(${FAXFILE})
hen I dial 9001,it's OK,but when I dial 9002,it faild fax,I
thought that maybe I had not set the sip_local in the [fax] config
session,but how can I set that?
I search on the google net, but have no idears about this,how can I
send the fax out?
Thanks for any hints!
fangh.
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[asterisk-users] Polycom boot error
I am having the same issue as below. Has this issue been solved or does anyone know an answer? This error recently began and we have multiple phones out of commission. PLEASE HELP!! http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere.-Original Message-From: Dovid Bender [mailto:asteriskusers at dovid.net]Sent: Tuesday, August 15, 2006 11:16 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom upgrade issueI believe 468* resets the phone but dosent return it to the orig. firmware. Also try to name the files with the phones mac id and see what happens. I am doing this with 1.6.6 and its working fine.- Original Message - From: Curt mailto:cshaffer at gmail.com Shaffer To: 'Asterisk Users Mailing List - mailto:asterisk-users at lists.digium.com Non-Commercial Discussion' Sent: Tuesday, August 15, 2006 10:07 PMSubject: [asterisk-users] Polycom upgrade issueOK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log 0527180621|cfg |4|00|Could not get all 512 bytes of the header.0527181013|cfg |4|00|Could not get all 512 bytes of the header.0527181014|app1 |6|00|Error application is not present.0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006 I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there? Thanks! Curt Clint Neider Email Administrator [EMAIL PROTECTED] Alta Resources | IT Application Services | 120 N Commercial St | Neenah, WI 54956 | Office (920) 751-5800 x 7472 | This email message is intended only for the addressee(s) and contains information that may be confidential and/or copyright. If you are not the intended recipient please notify the sender by reply email and immediately delete this email. Use, disclosure or reproduction of this email by anyone other than the intended recipient(s) is strictly prohibited. No representation is made that this email or any attachments are free of viruses. Virus scanning is recommended and is the responsibility of the recipient. ___ --Bandwidth 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: Sip Trunks
On 2006-10-18 12:34:43 -0700, [EMAIL PROTECTED] said: Hello, well, I need to configure two asterisk box like SIP trunks to send sip calls from one asterisk to the other and visceversa. So How I setup confi g files to get this working?.Thanks. You can do it via IAX2, there was a recipe posted here very recently that made this quite simple. Plus IAX2 saves bandwidth for trunked calls. http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtptimeout ___ --Bandwidth 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] Getting started with sample dial plans
Okay, I have Asterisk up and running on Fedora Core 5 with a TDM400 board with one FXO and FXS module. Zap is up and running and * is functioning with the modules. Oh yeah, and I have some soft phones configured and have them working as well. Now I'm ready to begin playing with dial plans and am having a difficult time getting started. I'm looking for some simple samples that might demonstrate basic functionality such as running the inside phone extension when an incoming call is received. Or, a simple Dial 9 for an outside line plan where whatever number is dialed (after the 9) is simply dialed via the Zap Line. In other words, something that makes * nearly transparent to begin with. Then, I'd like to slowly add to the dial plan as I learn more of the commands. I have not found any good samples of dial plans (other than the defaults built with *) that demonstrate basics like this. Are there some online references I could work through? I've been to Asterisk.com and their respective documentation, and Asteriskguru.com and not found what I'm looking for (maybe I overlooked it?). Any guidance is sincerely appreciated. -- Mitch ___ --Bandwidth 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] Embedded Asterisk
On Thursday 19 October 2006 14:10, Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. I happen to know that November's Linux Journal will have an article about running Linux/Asterisk on a Linsys WRTGS54SL router. Nothing too technical, but I hope you enjoy it. Mike Diehl. ___ --Bandwidth 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] /dev/zap/channel ownership
* is having permission problems accessing /dev/zap/channel. When I look, these devices (everything in /dev/zap) shows root.root for uid and gid. If I start Asterisk from the command line, it runs fine (running as Root). When I start it as a service, I get Oct 19 23:02:55 WARNING[10587] chan_iax2.c: Unable to open IAX timing interface: Permission denied Oct 19 23:02:55 WARNING[10587] chan_zap.c: Unable to open '/dev/zap/channel': Permission denied Oct 19 23:02:55 ERROR[10587] chan_zap.c: Unable to open channel 1: Permission denied So ... I changed ownership on /dev/zap/* to asterisk.asterisk and now everything seems to be running fine. My question is ... how would the ownership on these devices have changed? (I've not yet rebooted, but I'm suspicious that they'll revert back to root.root). -- Mitch ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
