Re: [Asterisk-Users] IP Phone Recommendation
Anders Svensson wrote: We use Grandstream GPX2000 for this. It works ok. Support 11 lines in basic. Anders I also use this phone, have read about the 11 lines, but how does one 'manage' these lines? The first 4 are easy, you have buttons for that, but how can you use the 'others' ? (incoming/outgoing) ___ --Bandwidth 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] Cisco 7940 Reboot
Kristian Kielhofner ha scritto: Or you can keep using the phones with SIP and use sip_notify. I think Ciscos support it. In my last try it was not doing it on cisco sip phones. Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-3000: Dual Registrations?
I'm wondering if there's anyone out there who has successfully gotten an SPA-3000 to register, as its documentation would indicate, on both ports 5060 (for standard client FXS service) and 5061 (for the purpose of originating calls via SIP from the PSTN interface on the box). I can get one or the other to register, but with the current firmware (3.1.7) so far I haven't been able to get both. The second ones gives me an error: chan_sip.c:10823 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.113' - Username/auth name mismatch I have checked the settings 1000 times; spa3000 is what I have in both the SIP stanza name as well as the username parameter, and that is the name I'm using in the SPA config screen for User It works all right, even though, according to the average of the many conflicting explanations as to how these things are to be configured, it shouldn't. Thx. B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Language
René Enskat [Teamware GmbH] ha scritto: -- Executing Set(SCCP/1000131-0006, Language()=de) edit your sccp.conf and in the general section set language=de; Default language setting Sergio Chersovani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP Phone Recommendation
You can use the speeddial buttons. They are configurable Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristof Hardy Sent: den 13 december 2005 09:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Phone Recommendation Anders Svensson wrote: We use Grandstream GPX2000 for this. It works ok. Support 11 lines in basic. Anders I also use this phone, have read about the 11 lines, but how does one 'manage' these lines? The first 4 are easy, you have buttons for that, but how can you use the 'others' ? (incoming/outgoing) ___ --Bandwidth 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] SPA-3000: Dual Registrations?
Brian Capouch wrote: I'm wondering if there's anyone out there who has successfully gotten an SPA-3000 to register, as its documentation would indicate, on both ports 5060 (for standard client FXS service) and 5061 (for the purpose of originating calls via SIP from the PSTN interface on the box). I can get one or the other to register, but with the current firmware (3.1.7) so far I haven't been able to get both. The second ones gives me an error: chan_sip.c:10823 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.113' - Username/auth name mismatch I have checked the settings 1000 times; spa3000 is what I have in both the SIP stanza name as well as the username parameter, and that is the name I'm using in the SPA config screen for User It works all right, even though, according to the average of the many conflicting explanations as to how these things are to be configured, it shouldn't. Thx. B. B, I have had it working perfectly with 2.0.11 for a while now, which I know is ancient... But hey, it works! -- 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] chan_capi AVM C2
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all, currently i running * 1.0.9 with chan_capi 0.3.5 my first problem is: in incoming call, when BCHAN is full in contr1 incoming call on contr2 are not answered with error : chan_capi.c:1953 in capi_handle_msg: received a call waiting CONNECT_IN but if use different msn in capi.conf incoming call works on both controler - -- Stephane Plichon | HASGARD jabber: [EMAIL PROTECTED] ~ -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDnpAYMI/jEEfAy/4RAgqhAJ9w7x+org8dQtiK2Ke5E3NPBg2AeQCfVAos 2uO9vsdVaZDvt9zK4H2X9uU= =ptnp -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AGI GET Variable Problem
Dear All,
Never Mind, I have solved the problem. It seems that you should clear the buffer for any 'waiting' response or else you will be getting an empty '200 result=1' response. So be sure to read, before you write in php agi script to ensure that you will get a proper response.
Regards,
Kengie
On 12/13/05, Kenige Ho [EMAIL PROTECTED] wrote:
Dear All,
I am trying to get a variable via AGI GET VARIABLE , but using AGI DEBUG I actually do see the variable get return but somehow my retrieving the variable via php. I don't get the value of the variable. Below is my code and my results. Please help. thank you.
Coding:
#!/usr/bin/php -q?phpob_implicit_flush(true);set_time_limit(6);$in = fopen(php://stdin,r);$stdlog = fopen(/var/log/asterisk/my_agi.log, w);
// toggle debugging output (more verbose)$debug = false;
// Do function definitions before we start the main loopfunction read() { global $in, $debug; $input = str_replace(\n, , fgets($in, 4096)); if ($debug) fputs($stdlog, read: $input\n);
return $input;}
function errlog($line) { global $err; echo VERBOSE \$line\\n;}
function write($line) { global $debug; if ($debug) fputs($stdlog, write: $line\n); echo $line.\n;}
// parse agi headers into arraywhile ($env=read()) { $s = split(: ,$env); // $agivar[str_replace(agi_,,$s[0])] = trim($s[1]); // errlog($s[0].,.$s[1]);
$agivar[$s[0]] = trim($s[1]); if(($endid.phpv==) || ($env==\n)) { break; }}
// main programecho VERBOSE \fone-check\\n;$tmp = GET VARIABLE x;write($tmp);errlog(Temp Var is . $tmp);$result = read();errlog(Before Strip Result is . $result);
$result = trim(ereg_replace(200 result=1,,$result));$result = trim(ereg_replace(\(,,$result));$result = trim(ereg_replace(\),,$result));
errlog(After Strip Result is . $result);
// clean up file handlers etc.fclose($in);fclose($stdlog);
exit;?
Results:
AGI Debugging EnabledAGI Tx agi_request: fone-check.agiAGI Tx agi_channel: SIP/1234-addaAGI Tx agi_language: enAGI Tx agi_type: SIPAGI Tx agi_uniqueid:
1134460079.22AGI Tx agi_callerid: 1234AGI Tx agi_calleridname: 1234AGI Tx agi_callingpres: 0AGI Tx agi_callingani2: 0AGI Tx agi_callington: 0AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 1233AGI Tx agi_rdnis: unknownAGI Tx agi_context: testAGI Tx agi_extension: 1233AGI Tx agi_priority: 11AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode: testAGI Tx AGI Rx VERBOSE fone-check fone-check.agi: fone-checkAGI Tx 200 result=1AGI Rx GET VARIABLE foneAGI Tx 200 result=1 (55)
AGI Rx VERBOSE Temp Var is GET VARIABLE fone fone-check.agi: Temp Var is GET VARIABLE foneAGI Tx 200 result=1AGI Rx VERBOSE Before Strip Result is 200 result=1
fone-check.agi: Before Strip Result is 200 result=1AGI Tx 200 result=1AGI Rx VERBOSE After Strip Result is fone-check.agi: After Strip Result is AGI Tx 200 result=1
Regards,
Kengie
___
--Bandwidth 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: CDR MySQL
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Dec 12 18:03:33 WARNING[7237]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/cdr_addon_mysql.so: undefined symbol: ast_load Dec 12 18:03:33 WARNING[7237]: loader.c:554 load_modules: Loading module cdr_addon_mysql.so failed! I can't remember now what was my problem. Can you check do you have cdr_addon_mysql.so file in /usr/lib/asterisk/modules directory? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info request from Sangoma users
Hi :) I have an A104 and wondered if other owners could confirm the strange behaviour I'm seeing.. it's best seen on an idle system, thus eliminating asterisk or other factors.. Very simply, just let 'vmstat 1' run for a few minutes and watch the output, specifically the 'sy' column... On the 2.4G Xeon machine I'm using, the system CPU usage sits very low for a minute or two, and then spikes up to 100 for a few seconds, before tailing off again - this happens all the time :( Interestingly, the 'load average' as reported with 'w' always stays at zero even with this high 'system load'... I moved the card to another PCI slot (and bus) and get the same thing, but now much more frequently, but for a much shorter length of time... Now bringing Asterisk into the picture, I can't use Monitor() because once I get even 5 simultaneous recordings, the real 'load average' on the machine spikes up to 2 and greater, and calls become stuttered as the machine fails to keep up with whatever it's doing.. The machine is SCSI, with a decent LSI Logic onboard controller and fast disks - so it's nothing to do with enabling DMA - hdparm shows 70MB/sec with minimal load increase. Can anyone confirm this behaviour? Cheers, Gavin ___ --Bandwidth 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] chan_capi AVM C2
Hi! currently i running * 1.0.9 with chan_capi 0.3.5 Try chan_capi-cm instead and see if it helps. Cheers, Philipp ___ --Bandwidth 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] Softphone with Hint support?
Hi! Are there any Windows-based softphones (SIP or IAX based) that support the new Hint system in Asterisk 1.2? I don't mind evaluating commercial options, if they're available. Try the SNOM softphone: http://www.snom.com/snom360softphone.html The only other softphone I am aware if is EyeBeam. Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Disconnecting
Hi all, I have really very serious problem. I installed G.729 and G.7231 from the Intel. And I got it is registered with asterisk. Registered translator 'g723tolin' from format g723 to slin, cost 1 Registered translator 'lintog723' from format slin to g723, cost 7 Registered translator 'g729tolin' from format g729 to slin, cost 1 Registered translator 'lintog729' from format slin to g729, cost 7 But when I am calling using g729 or g7231 the called party is not able to hear anything after 10 seconds, and in 10 seconds he is able to hear well. And when asterisk did not receive any frame from the called party, within 30 seconds, the call is hang-up with the following warning: Dec 13 10:34:07 DEBUG[20979]: channel.c:3248 ast_generic_bridge: Didn't get a frame from channel: SIP/123456-cffc Please help me to void this problem. -- Thank You, Code Lover ___ --Bandwidth 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] queue_log Vs show queue abandon calls discrepancy
Hi, Yesterday was the first day my call center operated under Asterisk 1.2.1. At the end of the day, I ran a "show queue queuename" and saw that the value of abandoned calls was 45. This morning, after updating my database with data from queue_log file, I saw, through Asterisk Guru Queue Stat, that I had only 33 abandoned calls. I tend to believe that queue_log and AsteriskGuru are more correct, because on some of the several times I tested the queue and abandoned it before being answered, I realized that the "show queue queuename" A: counter was incremented by 2. Has anyone realized such a problem? Thank you Dov ___ --Bandwidth 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] OOH323 - IAX2 : no sound
Hi, I use Asterisk 1.2. My configuration is: ooh323.conf: [general] port=1720 bindaddr=0.0.0.0 allow=all context=office tos=lowdelay iax.conf: [general] disallow=all allow=gsm bindport=4569 bindaddr=0.0.0.0 codecpriority=reqonly language=en jitterbuffer=yes tos=lowdelay [test] type=friend host=dynamic username=test secret=testpwd context=office callerid=Test User test extensions.conf: [general] static=yes writeprotect=yes [globals] [default] [office] exten = s,1,Dial(IAX2/test) Client test is connected by idefisk. On every call from outside I see on Asterisk console: == Starting OOH323/10.10.10.10-ba85 at office,2687988,1 failed so falling back to exten 's' -- Executing Dial(OOH323/10.10.10.10-ba85, IAX2/test) in new stack -- Called test -- Call accepted by 192.168.46.99 (format gsm) -- Format for call is gsm -- IAX2/test-2 is ringing -- IAX2/test-2 answered OOH323/10.10.10.10-ba85 At this moment I hear call sound in caller phone and hear nothing in answerer phone, so connection is not established. Why? -- Thanks, Eugene Prokopiev ___ --Bandwidth 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] No outgoing sound...sometimes
Verify communication between protocols. SIP ou IAX2. Jason Frisch [EMAIL PROTECTED] Enviado Por: [EMAIL PROTECTED] 13/12/05 00:13 Favor responder a Asterisk Users Mailing List - Non-Commercial Discussion Para:asterisk-users@lists.digium.com cc: Assunto:[Asterisk-Users] No outgoing sound...sometimes - Hi All, I have been having trouble with my asterisk box since last week. It was going fine until then and I can't remember changing anything.. nothing that I haven't put back anyway. The issue is with that about half of the calls received or placed, the outside party cannot hear my voice; I can hear the other end fine. I have checked the logs and nothing is different for the calls that fail. I thought it was the phones, but the messages played from asterisk itself also have the same problem. The native bridge in the below sections seems strange as I though this was disabled with canreinvite=no. denwa*CLI -- Executing Goto(SIP/10.129.46.102-0853ec38, sip|1000|1) in new stack -- Goto (sip,1000,1) -- Executing SetVar(SIP/10.129.46.102-0853ec38, CALLFILENAME=000-20051213-110514) in new sta ck -- Executing GotoIfTime(SIP/10.129.46.102-0853ec38, 18:00-10:00|mon-fri|*|*?24hour|s|1) in n ew stack -- Executing GotoIfTime(SIP/10.129.46.102-0853ec38, *|sat-sun|*|*?24hour|s|1) in new stack -- Executing Dial(SIP/10.129.46.102-0853ec38, SIP/2201SIP/2202|180|tTH) in new stack -- Called 2201 -- Called 2202 -- SIP/2201-afc3 is ringing -- SIP/2202-4367 is ringing -- SIP/2201-afc3 answered SIP/10.129.46.102-0853ec38 -- Attempting native bridge of SIP/10.129.46.102-0853ec38 and SIP/2201-afc3 == Spawn extension (sip, 1000, 4) exited non-zero on 'SIP/10.129.46.102-0853ec38' - conf file: sip.conf [general] port=5060 realm=ocn.ne.jp context=sip [EMAIL PROTECTED]:secret:[EMAIL PROTECTED]/number disallow=all allow=ulaw [number] type=friend host=voip-ca35323.ocn.ne.jp username=username secret=secret fromuser=number fromdomain=ocn.ne.jp port=5060 dtmfmode=inband disallow=all allow=ulaw nat=yes canreinvite=no context=sip [snip] If anybody has any idea where I should look, it would be most appreciated. 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 ___ --Bandwidth 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] Dlink DI-102 QOS Thingy?
Mojo Jojo wrote: Anyone using one of these as a QOS device in an Asterisk environment? If so, does it work well? No, I don't use one of these myself. However... Do you know what exactly it prioritizes? SIP only? IAX? ...during my recent DCE course, this product (or one extremely similar to it) was discussed. The QoS in these products are *extremely* simplistic and from a VoIP perspective covers only SIP. The configuration is little more than turning it on or off - and possibly (if my memory serves correctly) giving more priority to some IP addresses on your LAN than others. Unfortunately my DCE manual is at work, so if you want more information on the unit, send me an email off-list and I'll dig the manual up for you. I don't think this thing is going to work as I hoped (a simple/cheap device that will give priority to SIP and IAX). If the product description was accurate (and it should be) then it will only be of assistance to SIP calls. At best IAX calls will be unaffected - at worst, they may be marginally worse. If you need true QoS, you're better off looking at other routers. DLink have other products that should fit the bill, but they will be more expensive. Personally, I'm using a SonicWALL at home, but they are several orders of magnitude more expensive. ___ --Bandwidth 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] ericsson pabx and digium card TE110P
tir, 08,.11.2005 kl. 17.08 +0100, skrev Olivier Perrin: According your conf, you are in France, so i answer in french :-) That's really not very polite, since most people on this list won't understand a word you're saying. Other people read this list too, you know... -- Henning Kilset Pedersen ___ --Bandwidth 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] NAT Issues?
Hi All I am having various problems that I am convinced is NAT related. I have a Vega box on public IP talking to an Asterisk box on a public IP address. Calls from the Asterisk to the Vega and back are fine. I have 2 VoIP phones in a NAT network registered to the Asterisk box. The problems I am having appear to be intermitent and adding quality to the phone config I can see that they are constantly changing from reachable to not. Incoming calls come from Vega to Asterisk fine and then dials the extension they should end up at: Executing Dial(SIP/xx.xx.xx.xx-0816dbf8, SIP/111|20|tr) in new stack -- Called 111 Sometimes the call will go and the extension will dial immediately but more often than not it will just sit and not do anything or go straight to voicemail. Another problem is when you make an outgoing call from the phones they are passed to the Asterisk and then to the Vega, when the person answers the Vega and Asterisk shows the call as connected but the VoIP phone continues to ring. Also when a call does make it all the way incoming and outgoing when you hang up the VoIP phone still thinks its connected! All the above happens when trying to call VoIP phone to VoIP phone as well! Any advice would be most appreciated, email me off list if you wish. Many Thanks Scott Pinhorne ___ --Bandwidth 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] Info request from Sangoma users
Hello, Can you post what firmware your board is and what wanpipe driver version you are using? We do up to 50 concurrent recordings on our systems and they do not have recording issues. We use MegaRAID 320-1 cards as well. MATT--- On 12/13/05, Gavin Hamill [EMAIL PROTECTED] wrote: Hi :) I have an A104 and wondered if other owners could confirm the strange behaviour I'm seeing.. it's best seen on an idle system, thus eliminating asterisk or other factors.. Very simply, just let 'vmstat 1' run for a few minutes and watch the output, specifically the 'sy' column... On the 2.4G Xeon machine I'm using, the system CPU usage sits very low for a minute or two, and then spikes up to 100 for a few seconds, before tailing off again - this happens all the time :( Interestingly, the 'load average' as reported with 'w' always stays at zero even with this high 'system load'... I moved the card to another PCI slot (and bus) and get the same thing, but now much more frequently, but for a much shorter length of time... Now bringing Asterisk into the picture, I can't use Monitor() because once I get even 5 simultaneous recordings, the real 'load average' on the machine spikes up to 2 and greater, and calls become stuttered as the machine fails to keep up with whatever it's doing.. The machine is SCSI, with a decent LSI Logic onboard controller and fast disks - so it's nothing to do with enabling DMA - hdparm shows 70MB/sec with minimal load increase. Can anyone confirm this behaviour? Cheers, Gavin ___ --Bandwidth 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] g729 translation to zap (ISDN) doesn´t work
Hi Asterisk Users, i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a Debian 3.1. With a quadbri card installad, wich is running on the bristuff drivers. Everything seems to be fine so far. but now i wanted to use the g.729A Codec. I bought 5 licences and installed them: asterisk3*CLI show g729 0/0 encoders/decoders of 5 licensed channels are currently in use When i do sip to sip calls, everything is working fine (from a snom 190 wich is running with that codec to a sip phone with g.711a), asterisk is translating correct. the output on the CLI is: asterisk3*CLI show g729 1/0 encoders/decoders of 5 licensed channels are currently in use But if i try to call a zap channel from that sip phone (snom 190) wich runs that g729 Codec, i don´t hear anything on the ISDN Phone. the output on the CLI: asterisk3*CLI show g729 1/1 encoders/decoders of 5 licensed channels are currently in use Here is the output of the show channel command for the SIP Channel and the ZAP Channel: asterisk3*CLI show channel SIP/71-d293 -- General -- Name: SIP/71-d293 Type: SIP UniqueID: asterisk-2204-1134137006.49 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 256 WriteFormat: 256 ReadFormat: 64 1st File Descriptor: 31 Frames in: 7949 Frames out: 7956 Time to Hangup: 0 Elapsed Time: 0h2m39s -- PBX -- Context: default Extension: 329 Priority: 2 Call Group: 0 Pickup Group: 0 Application: Dial Data: Zap/g1/329 Stack: 0 Blocking in: ast_waitfor_nandfds asterisk3*CLI show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: asterisk-2204-1134137006.50 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 72 WriteFormat: 64 ReadFormat: 256 1st File Descriptor: 13 Frames in: 8255 Frames out: 8246 Time to Hangup: 0 Elapsed Time: 0h0m0s -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: Bridged Call Data: SIP/71-d293 Stack: -1 Blocking in: ast_waitfor_nandfds I don´t know what i can do on this problem and would be pleased to get some help. Thank you very much! -- Mit freundlichen Grüßen With kind regards Klaus Peras begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth 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] IAX2 show channels show Channel (NONE)
Hi all! I have got a bit strange output from iax2 show channels: Med venlig hilsen ComX Networks A/S Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-2600 Glostrup Denmark Phone: +45 70 25 74 74 Fax: +45 70 25 73 74 Web: www.comx.dk Dmitry Zhukovski Direct: +45 32 87 73 90 E-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 show channels show Channel (NONE)
One of my things also does very strange things, does somebody know what could be wrong with those things ? Maybe the other guy (you know, the one with the hair and the two or less eyes and two legs) could help me ? Please, at least give us some info... What are you referring to ? Zoa Dmitry Zhukovski wrote: Hi all! I have got a bit strange output from iax2 show channels: Med venlig hilsen ComX Networks A/S Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-2600 Glostrup Denmark Phone: +45 70 25 74 74 Fax: +45 70 25 73 74 Web: www.comx.dk Dmitry Zhukovski Direct: +45 32 87 73 90 E-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] IAX2 show channels show Channel (NONE)
Hi all! Sorry for last message. I have got a bit strange output from iax2 show channels: x*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None)xx.xx.xx.xx x 1/00318 00131/00162 1ms 0004ms 0036ms alaw IAX2/[EMAIL PROTECTED]/2 xx.xx.xx.xx x 2/00233 00022/00024 6ms ms 0021ms alaw IAX2/[EMAIL PROTECTED]/6 xx.xx.xx.xx x 6/00114 00025/00028 1ms 0002ms 0036ms alaw There is strange channel (None) which (probably) blocks another two from releasing. At least one of them are for 20 hours long. The server becomes more and more slowly and top shows top - 13:50:12 up 23:36, 1 user, load average: 3.16, 3.12, 2.90 Tasks: 54 total, 2 running, 52 sleeping, 0 stopped, 0 zombie Cpu(s): 50.0% us, 0.0% sy, 0.0% ni, 49.6% id, 0.4% wa, 0.0% hi, 0.0% si Mem: 2075600k total, 583220k used, 1492380k free, 131760k buffers Swap:0k total,0k used,0k free, 259308k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 1399 root 11 -5 41640 5448 2556 S 99.9 0.3 1439:20 asterisk 1 root 16 0 680 248 216 S 0.0 0.0 0:01.22 init 2 root RT 0 000 S 0.0 0.0 0:00.00 migration/0 Any ideas? Thank you in advance, Dmitry Med venlig hilsen ComX Networks A/S Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-2600 Glostrup Denmark Phone: +45 70 25 74 74 Fax: +45 70 25 73 74 Web: www.comx.dk Dmitry Zhukovski Direct: +45 32 87 73 90 E-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 show channels show Channel (NONE)
On Tue, December 13, 2005 13:47, Dmitry Zhukovski said: Hi all! I have got a bit strange output from iax2 show channels: Med venlig hilsen ComX Networks A/S Dmitry Zhukovski System developer Adding some info might be helpful? -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --Bandwidth 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] Info request from Sangoma users
On Tue, 2005-12-13 at 07:24 -0500, Matt Florell wrote: Hello, Can you post what firmware your board is and what wanpipe driver version you are using? Hi Matt :) I've already been through all this with Sangoma's support - just looking for external opinions from real-life installs - so thank you for the response :) I've seen this behaviour with everything from the first 2.3.2 Asterisk-compatible wanpipe to the latest 2.3.3-beta18. We do up to 50 concurrent recordings on our systems and they do not have recording issues. We use MegaRAID 320-1 cards as well. That's what I thought - I mean the amount of disk IO is absolutely nothing at all :( What kind of CPUs are you using? Also, single or dual (or a single with hyperthreading ?) What onboard L2 cache do they have? My last hope is to try a P4 machine with 1MB cache, since the others I've used have 512K.. They're all Dell machines - and I know the reaction that usually evokes when dealing with Digium hardware (been there, seen that...) - I thought someone like Sangoma with many more years in the business would be more immune to things like this :( Cheers, Gavin. ___ --Bandwidth 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] NAT/Qualify/RTP bug
Got a really wierd problem her. Maby it's a bug. But before i report it, i'll try my luck here. I have one asterisk server on public ip. I have two identical hardphones on two different LAN's. The firewall are different. Both are configured in asterisk with nat=yes and qualify=yes. For one phone everything works. SIP and audio is sent to the global address of the client. But for the other it's a bit different. SIP messages are sent to the global address of the client. You can call in and out. But the audio (RTP) is sent to the local address found in the SIP packets. The only thing that is different is the firewalls. How can a firewall, or anything else, tell asterisk to use the ipaddress in the sip packets instead of the global address, when i have told asterisk nat=yes Is this a bug? Or something i've missed. PS: i'v tried nat=route, same results Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000: Dual Registrations?
Brian Capouch wrote: I'm wondering if there's anyone out there who has successfully gotten an SPA-3000 to register, as its documentation would indicate, on both ports 5060 (for standard client FXS service) and 5061 (for the purpose of originating calls via SIP from the PSTN interface on the box). I can get one or the other to register, but with the current firmware (3.1.7) so far I haven't been able to get both. The second ones gives me an error: chan_sip.c:10823 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.113' - Username/auth name mismatch I have checked the settings 1000 times; spa3000 is what I have in both the SIP stanza name as well as the username parameter, and that is the name I'm using in the SPA config screen for User It works all right, even though, according to the average of the many conflicting explanations as to how these things are to be configured, it shouldn't. Yes, have had it working through many sipura firmware updates including the latest, and through many cvs-head updates over the last year or so. I'm out of town today and can't supply any sample config info, but it was very straight forward. I used different userid/secrets for the two registrations. Multiple associates and isp's (that I assist) also have it working fine. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: CDR MySQL
In article 3bf71fa80512121816u6928839cg2dfcf14d3ffb2c04 @mail.gmail.com, [EMAIL PROTECTED] says... I believe you are missing 2 variables in your conf file: table=cdr (the table your cdrs should be stored) sock=/var/lib/mysql/mysql.sock (the location to your mysql.sock) I didn't use those two with Asterisk 1.0.9 and it worked fine. Do I have to use them in 1.2.x or it's optional? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR MySQL
Instead of hostname=localhost, it would be hostname=IP address of MySQL server.On 12/12/05, Innocent Evil [EMAIL PROTECTED] wrote: I was also following this thread. Would anybody please tell, what would be configuration file if mysql is a different machine than asterisk box? Thanks, --You don't have any choice, you already made it before you came here. -Original Message-From: [EMAIL PROTECTED]Sent: Mon, 12 Dec 2005 21:16:23 -0500 To: asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] CDR MySQL I believe you are missing 2 variables in your conf file:table=cdr(the table your cdrs should be stored)sock=/var/lib/mysql/mysql.sock(the location to your mysql.sock)try something like this: [global]hostname=localhostdbname=dbasterisktable=cdrpassword=dbpassworduser=dbusersock=/var/lib/mysql/mysql.sockuserfield=1 On 12/12/05, Juanjo Portela [EMAIL PROTECTED] wrote: My cdr_mysql.conf is the same I was using for version.1.0.9 and it is as follow[global] hostname=localhostdbname=dbasteriskpassword=dbpassworduser=dbuseruserfield=1Any ideas?Thank you in advance, Juanjo___--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 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] Sip behind the NAT
On Fri, Dec 09, 2005 at 03:23:31PM -0500, Tom Rymes wrote: On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi: i added these two lines to my general context ,but nothing happened the same result the sound came in one way for 3 seconds and stopped but it didnt hangup. --- Jeffery Chen [EMAIL PROTECTED] wrote: If your Astersik server behind NAT too, your need modify SIP.conf like this externalIP= x.x.x.x localnet= x.x.x. hope this can help you Make sure that you have ports 5060 and ports 1-2 UDP forwarded to your Asterisk server. (Asterisk uses UDP for SIP, not TCP!!!) Also, in addition to the externip and localnet entries in sip.conf, You need to add a nat=yes entry I have a similar problem with a client's system. They have * 1.0.x behind a NAT with all the SIP phones on the local network. Their VoIP provider is outside the NAT (a Metaswitch at their ISP, connected to the phone lines from there). Their network guy has the firewall passing traffic on ports 5060 and 1-2 to the * system. I have externalIP and localnet set, but nat=no (default) is the case for this one. Occasionally they will place outgoing calls and the other party does not hear anything. Usually another attempt at the call will pass audio normally. One person who makes about 100 calls a day remembers having this happen on about 7 calls one day. No one recalls this ever happening on incoming calls (though this client primarily makes outgoing calls, I believe). Apparently this has been happening for a while and they just now mentioned it to me. Would nat=yes in the general section of sip.conf make a difference in this case? Is there anything else I could look at that might alleviate this problem? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth 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] Info request from Sangoma users
On Tue, 2005-12-13 at 12:59 +, Gavin Hamill wrote: [snip] What kind of CPUs are you using? Also, single or dual (or a single with hyperthreading ?) What onboard L2 cache do they have? My last hope is to try a P4 machine with 1MB cache, since the others I've used have 512K.. They're all Dell machines - and I know the reaction that usually evokes when dealing with Digium hardware (been there, seen that...) - I thought someone like Sangoma with many more years in the business would be more immune to things like this :( Been a while since I used Asterisk on a Dell box but I remember I had to turn off HT. Have you tried that? Think you can do it either in the BIOS or booting the kernel with noht. On Dell boxes I have also seen some funky NMI received for unknown reason. Dazed and confused messages in /var/log/messages. There is some boot option called nmi_watchdog that can be set at 0 or 1 that perhaps solves that one. When things get really weird try reseating the memory modules. And if you have a dual Xeon box and only one cpu shows up when booting Linux try reseating the processors too. While you are at it reseat everything you can find :) As a test you can also disable the onboard nic and stick in a quality nic on its own interrupt to see if that helps. And off course disable in the BIOS everything that you do not use (serial/parallel/usb etc.). Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Patch zaptel.init to support debian
Karl O. Pinc wrote: I foolishly made this patch against the zaptel 1.2 branch rather than trunk, although I did check that the trunk has the problem. It'll probably apply This script is completely unnecessary on Debian; just add the modules you wish to load into /etc/modules and they will be loaded at boot time. ___ --Bandwidth 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] ENUM For Presence
Douglas Garstang wrote: Then again updates are sent to the master DNS server, which filters them down to the slave DNS server, and you do queries to the slave... might take a few minutes to become effective. The bigger issue would be caching on the client ends; unless you set the TTL on these records to some ridiculously low value (which causes constant hits on the DNS servers and excess network traffic), the client resolvers will keep the records around. ___ --Bandwidth 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] 1.2.1 has broken voicemail realtime switching
It seems that version 1.2.1 has broken Asterisks ability to use realtime in the voicemail.conf file. It appears that the statement: switch = Realtime/@ is not read properly by Asterisk. -- Executing Voicemail(mailto:Local/[EMAIL PROTECTED],2, mailto:[EMAIL PROTECTED]) Dec 13 14:20:09 WARNING[7208]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '0625034077' n Executing hangup(mailto:Local/[EMAIL PROTECTED],2, ) The mailbox exists in the voicemail_users table, but Asterisk never even looks there: mysql select * from voicemail_users where mailbox='0625034077'; +--+-+-++--+--+---+---+-+--++ | uniqueid | customer_id | context | mailbox | password | fullname | email | pager | stamp | language | pwdset | +--+-+-++--+--+---+---+-+--++ | 11 | 0625034077 | default | 0625034077 | 11 | | | | 2005-11-29 07:47:00 | de | 1 | +--+-+-++--+--+---+---+-+--++ 1 row in set (0.00 sec) Asterisk is connected to MYSQL: AST-VM*CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username asterisk for 11 minutes, 20 seconds. AST-VM*CLI I also have no problems with extensions coming from the DB using the same switch statement. extcongif.conf is also correct. Falling back to 1.2 to see if the problem disappears. I believe it will since we already had this server running 1.2 and everything worked as expected. Regards to all, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No outgoing sound...sometimes
On Tuesday 13 December 2005 01:27, Jason Frisch wrote: I see. How would I go about checking such conflicts (for the future) With the old NIC in and everything running normal, type cat /proc/interrupts /tmp/ints-oldnic.txt Now with the new NIC in and everything running normal, type cat /proc/interrupts /tmp/ints-newnic.txt. You can then see the difference in interrupt routing. You may also want to capture the dmesg output for both cases as well. (dmesg /tmp/dmesg-oldnic.txt and /tmp/dmesg-newnic.txt). Have you put your old NIC back in and confirmed the problem comes back? -A. ___ --Bandwidth 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] No outgoing sound...sometimes
On Tuesday 13 December 2005 06:20, Mario Evangelista-Silva wrote: Verify communication between protocols. SIP ou IAX2. I get it with both protocols, but it's far more infrequent... one call in a hundred maybe. I've verified (with IAX2 at least) that both sides are seeing each other's packets, and that I am indeed seeing IAX2 control frames for RINGING and ANSWER. One thing I have noticed is that my * box is not recording CDRs for the one-way-audio calls. It records CDRs for all other calls just fine. -A. ___ --Bandwidth 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] Small / embedded system recommendations
On Tuesday 13 December 2005 02:11, Chris Mason (Lists) wrote: At sixty concurrent calls, you are not looking at a small embedded machine. Rack mount dual P3 or P4 in a small form factor I could see. I have to wonder if a CF card based system can be adequate for this kind of work, I have tended to move up to mirrored drives and a couple of slots for that type of installation. I dunno... I know you can terminate 192 concurrent calls (eight T1s) in a dual P4 Dell system, 60 calls is less than 1/3 of that. I realize that this doesn't scale linearly and that it also depends on what you're going to be doing CPU-wise... If you're doing external echo cancellation and limiting your transcoding (stick to ulaw or something light on the CPU) I could see sixty concurrent calls on smaller hardware. DTMF detection isn't all that CPU intensive. You do, of course, need to test to know for sure. There are far too many factors to armchair quarterback this kind of decision. :-) -A. ___ --Bandwidth 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] 1.2.1 has broken voicemail realtime switching
Joseph Rothstein wrote: It seems that version 1.2.1 has broken Asterisk's ability to use realtime in the voicemail.conf file. It appears that the statement: switch = Realtime/@ is not read properly by Asterisk. (Could you use a little more whitespace next time? G) What does this statement have to do with voicemail.conf? 'switch' statements are used in extensions.conf _only_, since they are used to look up dialplan extensions. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR MySQL
Thank you Traci,I put this two variables in my .conf file and it works!!!Well, It seems that this variables are not necessaries in old versions, but in newest ones.Thank you again,Juanjo I believe you are missing 2 variables in your conf file:table=cdr(the table your cdrs should be stored)sock=/var/lib/mysql/mysql.sock(the location to your mysql.sock )try something like this:[global]hostname=localhostdbname=dbasterisktable=cdrpassword=dbpassworduser=dbusersock=/var/lib/mysql/mysql.sock userfield=1 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Turning off hardware echo can on TE411P
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William K. Volkman Sent: Monday, December 12, 2005 9:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Turning off hardware echo can on TE411P Hello, On Mon, 2005-12-12 at 15:42 -0600, Kevin P. Fleming wrote: Eric Bishop wrote: Anyone know if Asterisk 1.2.1 supports turning off the hardware echo canceller WITHOUT recompiling the driver like I had to in 1.0.X? Add 'vpmsupport=0' to your modprobe.conf or equivalent. OK, so is there a way to have hardware echo canceling and have DTMF digits go out correctly? We bought the expensive hardware echo canceling card however it appears that we have to have vpmsupport=0 in order to get DNIS digits correctly (see my thread about ADIT and DNIS digits earlier). Clarifications about what to tweak appreciated. Thanks, William. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Turning off hardware echo can on TE411P
I didn't write this below. I replied with a blank line by mistake. I am truly sorry if you were confused by that. -Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, December 13, 2005 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Turning off hardware echo can on TE411P Jason Brashear wrote: OK, so is there a way to have hardware echo canceling and have DTMF digits go out correctly? We bought the expensive hardware echo canceling card however it appears that we have to have vpmsupport=0 in order to get DNIS digits correctly (see my thread about ADIT and DNIS digits earlier). Clarifications about what to tweak appreciated. Thanks, William. Is your name Jason or William? Very confusing. Please take this issue up with Digium tech support. ___ --Bandwidth 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] calls forwarded to busy agent
Hi We have a call queue setup with several agents using agentcallbacklogin. If one of the agent is logged in and is talking on the phone with another employee the queue application doesn't see that the phone is busy and continues to forward incoming calls to him. Since the agent cannot answer, the calls go to the agent's voicemail. in the show queues I see Agent/108 (Not in use) I did the show queues while talking to the agent in question. Is this normal behaviour ? Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work
Hi, i just figured out, that there is also a problem by going in a conference with the sip phone that runs the g729a codec. Could it be, that i have timing problems? I don´t have digium hardware installed, but i have ztdummy: asterisk3:/etc/asterisk# lsmod | grep ztdummy ztdummy 3748 0 zaptel225540 24 ztdummy,qozap Does anybody have a advice for me? Mit freundlichen Grüßen With kind regards Klaus Peras Klaus Peras schrieb: Hi Asterisk Users, i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a Debian 3.1. With a quadbri card installad, wich is running on the bristuff drivers. Everything seems to be fine so far. but now i wanted to use the g.729A Codec. I bought 5 licences and installed them: asterisk3*CLI show g729 0/0 encoders/decoders of 5 licensed channels are currently in use When i do sip to sip calls, everything is working fine (from a snom 190 wich is running with that codec to a sip phone with g.711a), asterisk is translating correct. the output on the CLI is: asterisk3*CLI show g729 1/0 encoders/decoders of 5 licensed channels are currently in use But if i try to call a zap channel from that sip phone (snom 190) wich runs that g729 Codec, i don´t hear anything on the ISDN Phone. the output on the CLI: asterisk3*CLI show g729 1/1 encoders/decoders of 5 licensed channels are currently in use Here is the output of the show channel command for the SIP Channel and the ZAP Channel: asterisk3*CLI show channel SIP/71-d293 -- General -- Name: SIP/71-d293 Type: SIP UniqueID: asterisk-2204-1134137006.49 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 256 WriteFormat: 256 ReadFormat: 64 1st File Descriptor: 31 Frames in: 7949 Frames out: 7956 Time to Hangup: 0 Elapsed Time: 0h2m39s -- PBX -- Context: default Extension: 329 Priority: 2 Call Group: 0 Pickup Group: 0 Application: Dial Data: Zap/g1/329 Stack: 0 Blocking in: ast_waitfor_nandfds asterisk3*CLI show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: asterisk-2204-1134137006.50 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 72 WriteFormat: 64 ReadFormat: 256 1st File Descriptor: 13 Frames in: 8255 Frames out: 8246 Time to Hangup: 0 Elapsed Time: 0h0m0s -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: Bridged Call Data: SIP/71-d293 Stack: -1 Blocking in: ast_waitfor_nandfds I don´t know what i can do on this problem and would be pleased to get some help. Thank you very much! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth 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] Turning off hardware echo can on TE411P
Jason Brashear wrote: OK, so is there a way to have hardware echo canceling and have DTMF digits go out correctly? We bought the expensive hardware echo canceling card however it appears that we have to have vpmsupport=0 in order to get DNIS digits correctly (see my thread about ADIT and DNIS digits earlier). Clarifications about what to tweak appreciated. Thanks, William. Is your name Jason or William? Very confusing. Please take this issue up with Digium tech support. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM 190 using 2 lines
I have a Snom 190 and setup two lines one for the local Asterisk and the Other for a remote asterisk. I can see that both likes register and in the web interface say they are ok. My problem is that line 1 takes precedence. I am not sure how to use line 2. If I go to the main setup page in the web browser for the snom I can change the Outgoing Identity: Line 2 and that works but its like switching me to the other network. What I was hoping to do was to setup my Function Keys to dial out one wither line 1 or line 2. Is that Possible? Am I missing anything? P1 is set to line : sip:[EMAIL PROTECTED];user=phone P2 is set to line : sip:[EMAIL PROTECTED];user=phone But this seems to have no effect. The Function Key 2 seem to still default to line 1. Any ideas? -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] Polycom 501 remapping keys
Yeah I just got in a 301 to test and I can configure a key (for example in sip.cfg key.IP_300.2.function.prim=Messages/ and then when I hit the line 2 key it drops me right into VM (since I have that configured too) Still playing around, I noticed that if you get the soft keys (the menu ones under the LCD) then it ALWAYS is that function... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Sent: Friday, December 09, 2005 9:06 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 remapping keys There has been a fair amount of converstaion about this, but I'm not sure anyone really has this working. I had exactly the same problem that the button got remapped to a volume up function. The only button remapping I got working was to map the Transfer button to the # key so that when you hit Transfer it started and Asterisk based transfer. I would love to hear from someone who has this working. Matthew O'Connor [EMAIL PROTECTED] wrote: I've tried to configure the services-key on my Polycom 501 to run a SpeedDial-entry in [MACADRESS]-directory.xml (which would call a asterisk-extension that starts SayUnixTime) but i have not been able to accomplish my goal. Whe configuring the SpeedDial-function in sip.cfg VolUp is started when i press the Services-Key. Also some other possible functions listed under 4.6.1.15 in the SIP 1.6 Administrator Guide fail. Some of them were working with the expected function, some where not giving any response at all but some where starting totally different functions, e.g. configuring Redial as the function starts Settings, function Messages starts Redial, SpeedDialMenu starts VolUp, VolUp starts Line1 :-[ I've seen that other failed as well (http://lists.digium.com/pipermail/asterisk-users/2005-October/130129.ht ml) - anyone ever got this working? Maybe with BootROM 3.0/3.1? Or should i downgrade to 1.5 where there was a ipmid-file for remapping-info...? I'm running Firmware 1.6.2.0041/BootROM 2.6.2.0032 regards Christian ___ --Bandwidth 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] Testing 10.0.0.203 with 10.0.0.0
FC4, Asterisk 1.0.9 and SjPhone softphone. On CLI I get this message every 20 sec. # Testing 10.0.0.203 with 10.0.0.0 10.0.0.203 is the IP of softphone and 10.0.0.0 is the network defind in sip.conf. Asterisk server is on 10.0.0.26 address. Why do I get this message? sip.conf [general] externip = 123.123.143.254 fromdomain=lama.hr localnet=10.0.0.0/255.255.255.0 port=5060 bindaddr=0.0.0.0 context=sip srvlookup=yes dtmfmode=rfc2833 disallow=all allow=gsm allow=ulaw allow=alaw musicclass=default useragent=PBX [201] type=friend username=201 secret=myswc host=dynamic defaultip=10.0.0.83 mailbox=201 canreinvite=yes [211] type=friend username=211 secret=mysec host=dynamic defaultip=10.0.0.203 mailbox=211 canreinvite=yes -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info request from Sangoma users
On Tue, 2005-12-13 at 14:32 +0100, Patrick wrote: Been a while since I used Asterisk on a Dell box but I remember I had to turn off HT. Have you tried that? For sure, I've tried HT on and off, with 2.4 and 2.6 kernels :) or booting the kernel with noht. On Dell boxes I have also seen some funky NMI received for unknown reason. Dazed and confused messages in /var/log/messages. Yes I had those with the Digium card (before I returned it, obviously :), although Digium support managed to solve those in the driver. While you are at it reseat everything you can find :) Feel the build quality :)) As a test you can also disable the onboard nic and stick in a quality nic on its own interrupt to see if that helps. And off course disable in the BIOS everything that you do not use (serial/parallel/usb etc.). All very sage advice - I have another box to try it on yet before curling up in a corner and crying - I'll report back if I find anything spectacularly wrong :) Cheers, Gavin. ___ --Bandwidth 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] extensions and regular expressions ( probablyan easy question )
Or you can treat everything as a 10 digit number retaining in a variable
whether the user dialed one or not
exten = _1NXXNXX,1,SetVar(ONPRESSED=TRUE) *** skip this step if you
don't care whether the one was pressed in any of your dialplans
exten = _1NXXNXX,2,Goto(${CONTEXT},${EXTEN:1},1)
exten = 8661234567,1,Goto(800-in)
Can be your thing or not, depending on what you're using it for.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: December 10, 2005 8:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extensions and regular expressions (
probablyan easy question )
Or, just do...
exten = 18661234567,1,Goto(800-in)
exten = 8661234567,1,Goto(800-in)
It's kind of tough to truly understand what you are trying to accomplish (or
ask for). Apparently you've got something more in mind that words are making
it through the list. Reading between the lines, it would appear from the
800-in that calls are coming in from some external source, and you trying to
do something with them. Can you be a little more explicit.
Hi Dan,
Thanks for the info, but what I'm after is the ability to match a
digit/character 0 or 1 times at the beginning of the string. If I'm
reading your example right, it'll match anything starting with 866,
which doesn't work for me. I am trying to match:
18661234567 and 8661234567
Sean
ps: The pdf doesn't have a good explaination of this either, although
it occurs to me that this might not be possible with * if I'm having
such a hard time finding it.
Daniel Wright wrote:
Sean Kennedy wrote:
Hi all,
I'm having a hard time finding information related to the regular
expressions that can be used in a dialplan, specifically as an
extension. For example, I have an 800 number which I'd like to jump
directly to if my users dial it, instead of going over my pstn
termination. Currently, it looks like this:
exten = 8661234567,1,Goto(800-in)
However, I'd like 1866123456 to match as well. I can't find in the
wiki or sample configs how to say match this 0 or 1 times.
Can anybody provide a link that would go over this? Again, I've
been digging through the wiki, but I seem to be missing it.
Thanks
Sean
You could do it like this:
exten = _866.,1,GoTo(800-in)
The period means match one or more characters.
You can find reference to expressions and how they work in this pdf
book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip
___
--Bandwidth 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: 1.2.1 has broken voicemail realtime
'searchcontexts=yes' added to my voicemail.conf file solved the problem. Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Partial PRI pass thru?
I want to put a * server in front of our legacy phone system. Currently this legacy system is connected to the CO with an ISDN PRI interface. With a dual PRI card in the * server can I only pass thru a certain number of channels to the legacy phone system and then leave the other half of the channels for Asterisk to use for a Meet Me conference bridge configuration? Can I make the * server nearly invisible to the legacy phone system? The legacy phone system would only be able to use, for example, channels 16-23 from the ISDN PRI coming out of the * server. CO---[ISDN PRI]--Asterisk--[ISDN PRI]--legacy phone system Thanks! -- Matt Burleigh Senior Systems Engineer Enterprise Integration, Inc. eiisolutions.com 703-236-0790 ___ --Bandwidth 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] Partial PRI pass thru?
Matt Burleigh wrote: The legacy phone system would only be able to use, for example, channels 16-23 from the ISDN PRI coming out of the * server. You cannot make it invisible, because the D-channel cannot be shared. However, PRI channels are allocated dynamically, so doing what you want to do is trivial using channel group assignments and count checking before initiating a dial operation in either direction. Just a matter of defining the logic and writing some dialplan magic to do it :-) The trick is understanding that the limitation is not a specific group of channels, but a maximum number of simultaneous channels in use. ___ --Bandwidth 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: 1.2.1 has broken voicemail realtime
This same thing happened to me last night, I'll have to try this out and see if it works for us too :) Aaron Joseph Rothstein wrote: 'searchcontexts=yes' added to my voicemail.conf file solved the problem. Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 408 Request Timeout vs. 403 Forbidden
Please correct me if I am wrong, but if a SIP call goes unanswered, shouldn't the proper response be a '408 Request Timeout', and not a 403 Forbidden? Anyone care to comment? Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXOTUNE Error on channel 2
What does this error mean when running fxotune on my TDM04B could not fill input buffer on channel 2 Thanks -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 978-203-3848 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
RE: [Asterisk-Users] SPA-3000: Dual Registrations?
We just posted an updated guide to the SPA-3000 a few days ago. The example uses AMP but all the settings are there: http://voipspeak.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Tuesday, December 13, 2005 5:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000: Dual Registrations? Brian Capouch wrote: I'm wondering if there's anyone out there who has successfully gotten an SPA-3000 to register, as its documentation would indicate, on both ports 5060 (for standard client FXS service) and 5061 (for the purpose of originating calls via SIP from the PSTN interface on the box). I can get one or the other to register, but with the current firmware (3.1.7) so far I haven't been able to get both. The second ones gives me an error: chan_sip.c:10823 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.113' - Username/auth name mismatch I have checked the settings 1000 times; spa3000 is what I have in both the SIP stanza name as well as the username parameter, and that is the name I'm using in the SPA config screen for User It works all right, even though, according to the average of the many conflicting explanations as to how these things are to be configured, it shouldn't. Yes, have had it working through many sipura firmware updates including the latest, and through many cvs-head updates over the last year or so. I'm out of town today and can't supply any sample config info, but it was very straight forward. I used different userid/secrets for the two registrations. Multiple associates and isp's (that I assist) also have it working fine. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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 book feedback
Just curious what everyone (as in, the people that have read it or use it) thinks about the O'Reilly Asterisk book. I'd really like to delve into the nitty gritty of Asterisk, but I'm getting kinda tired of swimming through forums and Google results. I've been reading the wiki off and on for about a week now, but I'm wondering if a book would be the way to go to get a solid foundation. My IT career for the past 10 years has been based off of learn-as-I-go methods, but I'd really like to learn asterisk the right way. I have a couple Asterisk servers up and running and in use, but they're very small systems (~10 extensions, connected to 3 or 4 pots lines). I have some clients that want to use VOIP, but they're bigger businesses, and I'm not yet comfortable enough to roll out a bigger system. So if there are any other methods for learning Asterisk that I should consider, please do tell! Any opinions (on the book or otherwise) appreciated. Thanks! -ross ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] calls forwarded to busy agent
Yes, it is correct. The best way to handle this problem (on 1.2) is to pause the agent before the outbound call and the unpause him when he's done. Yours l. On Tue, 13 Dec 2005 15:20:56 +0100, Patrick Fortin [EMAIL PROTECTED] wrote: Hi We have a call queue setup with several agents using agentcallbacklogin. If one of the agent is logged in and is talking on the phone with another employee the queue application doesn't see that the phone is busy and continues to forward incoming calls to him. Since the agent cannot answer, the calls go to the agent's voicemail. in the show queues I see Agent/108 (Not in use) I did the show queues while talking to the agent in question. Is this normal behaviour ? Thanks Patrick -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Partial PRI pass thru?
I want to put a * server in front of our legacy phone system. Currently this legacy system is connected to the CO with an ISDN PRI interface. With a dual PRI card in the * server can I only pass thru a certain number of channels to the legacy phone system and then leave the other half of the channels for Asterisk to use for a Meet Me conference bridge configuration? Yes. Can I make the * server nearly invisible to the legacy phone system? Yes providing you set the settings on whatever span is connected to your PBX the same as your carrier's and no configuration changes are required on the PBX. The legacy phone system would only be able to use, for example, channels 16-23 from the ISDN PRI coming out of the * server. You could assign specific zap channels but there is no need. Just define a number of channels and let asterisk decide which channels to use. CO---[ISDN PRI]--Asterisk--[ISDN PRI]--legacy phone system Thanks! -- Matt Burleigh Senior Systems Engineer Enterprise Integration, Inc. eiisolutions.com 703-236-0790 ___ --Bandwidth 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] extension seen as busy when it is not
Every few days our receptionist's phone will not take calls on one of the extensions. We have an extension 118 going to the first two lines of her phone and extension 101 going to the other. If we try to dial 118 it goes to voicemail even though she is not on the phone. Asterisk is thinking she is not logged on or something because the message in the log stays there is congestions calling that extension: dialparties.agi: extnum: 118 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: Extension 118 has call waiting enabled2 dialparties.agi: get_dial_string: extnum=[118] -- dialparties.agi: get dial string 118, SIP/118 -- dialparties.agi: DbSet CALLTRACE/118 to 101 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(SIP/101-dc56, SIP/118|25|tTwWr) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing GotoIf(SIP/101-dc56, 0?s-CHANUNAVAIL|1) in new stack -- Executing GotoIf(SIP/101-dc56, 0?s-CHANUNAVAIL|1) in new stack -- Executing NoOp(SIP/101-dc56, Sending to Voicemail box 118) in new stack What can I look at to see why this is happening? Thanks, James ___ --Bandwidth 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 book feedback
On Tue, 13 Dec 2005 09:45:09 -0600 Ross C [EMAIL PROTECTED] wrote: Just curious what everyone (as in, the people that have read it or use it) thinks about the O'Reilly Asterisk book. I'd really like to delve into the nitty gritty of Asterisk, but I'm getting kinda tired of swimming through forums and Google results. I've been reading the wiki off and on for about a week now, but I'm wondering if a book would be the way to go to get a solid foundation. My IT career for the past 10 years has been based off of learn-as-I-go methods, but I'd really like to learn asterisk the right way. I have a couple Asterisk servers up and running and in use, but they're very small systems (~10 extensions, connected to 3 or 4 pots lines). I have some clients that want to use VOIP, but they're bigger businesses, and I'm not yet comfortable enough to roll out a bigger system. So if there are any other methods for learning Asterisk that I should consider, please do tell! Any opinions (on the book or otherwise) appreciated. Thanks! Well, the book is freely available for download as a pdf, so you can check it out yourself and see what you think. The general consensus here seemed to be that the book was an excellent resource. If you find the pdf version as useful as I think you will, I would strongly suggest purchasing a hard copy. The price is good for what you get, and the authors put a LOT of work into it. Regards, Ozz. (Not affiliated with the book in any way) pgpQweVkjHDSP.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Partial PRI pass thru?
Thanks for the responses. I guess the next step is to get a Digium TE210P. Are there any other 2 port PRI cards anyone would recommend for *? -- Matt Burleigh Senior Systems Engineer Enterprise Integration, Inc. eiisolutions.com 703-236-0790 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, December 13, 2005 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Partial PRI pass thru? I want to put a * server in front of our legacy phone system. Currently this legacy system is connected to the CO with an ISDN PRI interface. With a dual PRI card in the * server can I only pass thru a certain number of channels to the legacy phone system and then leave the other half of the channels for Asterisk to use for a Meet Me conference bridge configuration? Yes. Can I make the * server nearly invisible to the legacy phone system? Yes providing you set the settings on whatever span is connected to your PBX the same as your carrier's and no configuration changes are required on the PBX. The legacy phone system would only be able to use, for example, channels 16-23 from the ISDN PRI coming out of the * server. You could assign specific zap channels but there is no need. Just define a number of channels and let asterisk decide which channels to use. CO---[ISDN PRI]--Asterisk--[ISDN PRI]--legacy phone system Thanks! -- Matt Burleigh Senior Systems Engineer Enterprise Integration, Inc. eiisolutions.com 703-236-0790 ___ --Bandwidth 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] Info request from Sangoma users
We use all Asus motherboards now, with single P4 processors(some with 512k, 1024k and 2048k L2 caches) We run most of them with HT on, no issues there. Also, if you are using the on-board RAID, it's not really a complete LSILogic RAID, They(LSILogic) won't support it because Dell does modifications to the hardware and firmware to optimize it's performance. Many calls to Dell and LSILogic left me very frustrated about this. I now personally avoid Dell servers at almost all costs. (I've even refused a free one offered to me) I've just had too many issues with them in the past(and Compaq too). Now we build all of our servers ourselves and can't be hapier about it. And with the money we save we buy replacement parts to keep on hand and have a spare server ready to replace any of our production servers at a moment's notice. Good luck, MATT--- On 12/13/05, Gavin Hamill [EMAIL PROTECTED] wrote: On Tue, 2005-12-13 at 14:32 +0100, Patrick wrote: Been a while since I used Asterisk on a Dell box but I remember I had to turn off HT. Have you tried that? For sure, I've tried HT on and off, with 2.4 and 2.6 kernels :) or booting the kernel with noht. On Dell boxes I have also seen some funky NMI received for unknown reason. Dazed and confused messages in /var/log/messages. Yes I had those with the Digium card (before I returned it, obviously :), although Digium support managed to solve those in the driver. While you are at it reseat everything you can find :) Feel the build quality :)) As a test you can also disable the onboard nic and stick in a quality nic on its own interrupt to see if that helps. And off course disable in the BIOS everything that you do not use (serial/parallel/usb etc.). All very sage advice - I have another box to try it on yet before curling up in a corner and crying - I'll report back if I find anything spectacularly wrong :) Cheers, Gavin. ___ --Bandwidth 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] chan_capi AVM C2
Philipp von Klitzing wrote: Hi! currently i running * 1.0.9 with chan_capi 0.3.5 Try chan_capi-cm instead and see if it helps. Cheers, Philipp compiling 0.5.4 when there was more than 2 call i got : ERROR[6060]: chan_capi.c:2324 capi_handle_connect_indication: received a call waiting CONNECT_IND :-( -- Stephane Plichon | HASGARD jabber: [EMAIL PROTECTED] ~ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mISDN Caller ID problem
Hello everyone, I am trying mISDN driver with asterisk 1.2.1 but when i call from SIP to mISDN and from mISDN to SIP, the caller ID appears always with a leading 0 (0X). I think the problem is with nationalprefix. How can I remove that zero Here is my config. [general] debug=0 trace_calls=false trace_dir=/var/log/ bridging=yes stop_tone_after_first_digit=yes append_digits2exten=yes l1_info_ok=yes clear_l3=no method=standard dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=default language=en nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 te_choose_channel=no dialplan=0 use_callingpres=yes ;always_immediate=no ;immediate=no ;hold_allowed=yes ;callgroup=1 ;pickupgroup=1 ;presentation=not_screened ;echocancel=no echocancelwhenbridged=no echotraining=yes [group1] ports=1 context=bri_card_1 msns=* Thanks in advance Pedro Nunes ___ --Bandwidth 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: Odd DTMF issue over PRI
It looks like http://bugs.digium.com/view.php?id=5266 is the problem here. My CDR shows as not answered for the tool free number. The local number answers and call forwards. Questions: It says it was committed on 10-04-05. How do I know which versions that was? I am currently running: asterisk stable 1.0.9 zaptel stable 1.2.1 libpri stable 1.0.9 I was told that zaptel and asterisk versions do not have to match. What about libpri? Can I go to libpri 1.2.1 and stay with asterisk 1.0.9? Should I just patch 1.0.9? (I would have to figure out which version the patch was for) -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Steven [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] This is an outbound issue that affects SIP and Zap (T1 from another PBX) channels going out our PRI to Telco. I have two ATT conference number that will take the conference access codes. (in theory) (214) 622 4991 (866) 340 2763 If we dial the toll free one, the menus time out because they are not recieving any DTMF. If I wait and connect to the conference receptionist/tech(?) they can do a three way call back in and my DTMF works. (they then tell me there is no problem) If I call the 214 number it works without issue. The odd thing here is that I receive DTMF back from them when it first answers the line. ref: Dec 6 10:28:21 VERBOSE[1448]: -- Called g0/12146224991 Dec 6 10:28:21 DEBUG[1448]: Ooh, format changed from unknown to ulaw Dec 6 10:28:24 DEBUG[1448]: DTMF digit: * on Zap/2-1 Dec 6 10:28:24 DEBUG[1448]: DTMF digit: 8 on Zap/2-1 Dec 6 10:28:24 DEBUG[1448]: Enabled echo cancellation on channel 2 Is this something that they are sending to test/set some DTMF setting on my side, or might I just be hearing them call forward to some other number? The thing that really confuses me is the 866 number. If there is something wrong with my setup, then why does my DTMF work if they 3 way back in. I am still on the same call and do not think any settings on my side would change because of what they do on the other side. But I still think the Issue IS on my side, because if the main toll free ATT Conference number has this problem, I think they would know and would have addressed it. zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs em=25-48 loadzone = us defaultzone = us zapata.conf: context=from-pstn switchtype=national priindication = outofband signalling=pri_cpe rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 faxdetect=no group=0 callgroup=1 pickupgroup=1 immediate=no accountcode=I musiconhold=default channel = 1-23 -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- ___ --Bandwidth 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] Skips and Pops in Call Recordings
Matt Florell wrote: Hello, Need some more information here: - hardware specs (including what kind of hard drives?) The Asterisk server is a Dell PowerEdge 6850 with the following specs. Please note that we are NOT recording to the hard drive. We are recording to a RAM disk as detailed here (http://voip-info.org/wiki/view/Asterisk+Dimensioning) under the heading 512 simultaneous SIP-to-SIP calls with Digital Recording. Unfortunately, the scalability tests we did at that time assumed that if call quality was good, so was the quality of the recording. Processor: Quad Intel Xeon 3.16GHz/1MB Cache Memory: 20 GB DDR2 400MHZ Single Ranked DIMMs (4 GB System / 16 GB RAM Disk) Hard Drive: Two 73GB, U320, SCSI, 1IN 15K Configured in a RAID 1 (Mirrored) Hard Drive Controller: Embedded RAID - PERC4 Integrated (Driver: megaraid_mm, megaraid_mbox) Everything else: http://www.dell.com/downloads/global/products/pedge/en/PE6850_specs.pdf - Linux kernel version 2.6.12-1.1376_FC3smp (Fedora Core 3). - running Xwindows? No. - Asterisk version ABE-A.2-beta (Asterisk Business Edition A.2 beta). - kind of calls you are recording (Zap, SIP, IAX, Meetme, ...) Calls originate on the PSTN and are handled by a Cisco AS5400 Universal Gateway that is a SIP peer of Asterisk. The AS5400 converts the calls from TDM channels to VoIP (SIP) channels before sending them to Asterisk. The Asterisk dialplan then routes them to one of our agents, who are using SNOM 320 VoIP (SIP) phones. Essentially all of our calls are SIP-to-SIP, with absolutely no protocol bridging or transcoding occurring on the Asterisk server. The Asterisk server handles the following major tasks: - Routing calls through the dialplan to (dynamic) agents in the appropriate queues. - Adding/removing agents to/from queues via AddQueueMember and RemoveQueueMember (NO static agents!). - Recording calls via the Monitor application directly to RAM disk. Calls are moved to a remote machine for mixing. - ChanSpy-based quality assurance of calls. Neither ChanSpy nor the quality of the calls themselves is affected by the problem. - how many recordings at once Anywhere from 5 to 30 concurrent recordings. This is not our planned peak, but it's where we've experienced the problem so far. We have not yet determined if the number of concurrent recordings is an issue, but we are considering it. We also haven't determined if the problem gets worse as the number of recordings increases, but it definitely exists throughout that entire range. In my experience, HyperThreading does not cause recording problems, it's usually a disk issue. When we had issues, switching to fast SCSI drives on a MegaRAID 320-1 with the megaraid2 linux driver solved all of our problems(skips and clicks/pops) The disk issues also directly interfere with call quality, as our previous scalability tests showed. Digium seems to think that the issue is scaling (some resource contention that causes a bit of audio to be unavailable when the write occurs). I see their point, but given our hardware and the current call volume I'm not completely sold on it. Could it be a configuration issue (file handles, interrupts, etc...)? MATT--- Colin Anderson wrote: Matt Roth: FWIW, I am recording 1000-1500 calls a day (as of 8:52PM today, 1482 calls!) of various length on my Netfinity with the onboard IBM RAID controller in RAID 5 Ultra 320 SCSI with suprisingly good quality. As the other Matt indicated, maybe what is needed here is an intelligent controller to offload some of the chore. No definite solution here, but at least it's another data point to compare. I appreciate any information contributed by list users. It's by far the most valuable resource available to me. On 12/12/05, Matt Roth [EMAIL PROTECTED] wrote: List users, I'm using the Monitor application to record calls. Most of the recordings are audible, but contain skips accompanied by a popping sound. Sometimes they are isolated, sometimes they appear in groups. Call quality is excellent and seems unaffected by whatever is causing this problem. If anyone has experienced this problem before, I'd appreciate if you'd share what the source was and any tips on eliminating it. I contacted Digium tech support and they suggested turning off hyperthreading. I have done that, but I won't know if it improved things until tomorrow. The machine is running at a moderate call volume and is always at least 90% idle. I'm not seeing any Avoided deadlock messages in the logs. If you need any more information, I'd be happy to provide it. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth 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: Odd DTMF issue over PRI
I was wrong. This patch is for channels/chan_zap.c I have been hesitant to go to 1.2.1 without config testing. Should I have any negative issues going from 1.0.9 to 1.0.10? ( I have to see if the changes are in the 1.0.10 version of channels/chan_zap.c) -- -- Steven It looks like http://bugs.digium.com/view.php?id=5266 is the problem here. My CDR shows as not answered for the tool free number. The local number answers and call forwards. Questions: It says it was committed on 10-04-05. How do I know which versions that was? I am currently running: asterisk stable 1.0.9 zaptel stable 1.2.1 libpri stable 1.0.9 I was told that zaptel and asterisk versions do not have to match. What about libpri? Can I go to libpri 1.2.1 and stay with asterisk 1.0.9? Should I just patch 1.0.9? (I would have to figure out which version the patch was for) ___ --Bandwidth 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] SPA-3000: Dual Registrations?
Are you trying to register both lines to the same user account in *? That wont work, a user can only be registered once at any time. Kerry Garrison wrote: We just posted an updated guide to the SPA-3000 a few days ago. The example uses AMP but all the settings are there: http://voipspeak.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Rich Adamson Sent: Tuesday, December 13, 2005 5:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000: Dual Registrations? Brian Capouch wrote: I'm wondering if there's anyone out there who has successfully gotten an SPA-3000 to register, as its documentation would indicate, on both ports 5060 (for standard client FXS service) and 5061 (for the purpose of originating calls via SIP from the PSTN interface on the box). I can get one or the other to register, but with the current firmware (3.1.7) so far I haven't been able to get both. The second ones gives me an error: chan_sip.c:10823 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.113' - Username/auth name mismatch I have checked the settings 1000 times; spa3000 is what I have in both the SIP "stanza" name as well as the "username" parameter, and that is the name I'm using in the SPA config screen for "User" It works all right, even though, according to the average of the many conflicting explanations as to how these things are to be configured, it shouldn't. Yes, have had it working through many sipura firmware updates including the latest, and through many cvs-head updates over the last year or so. I'm out of town today and can't supply any sample config info, but it was very straight forward. I used different userid/secrets for the two registrations. Multiple associates and isp's (that I assist) also have it working fine. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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 book feedback
The book is a great *starting* point, IMHO. If you've spent a considerable amount of time reading other sources, you probably won't find much new information in the book. OTOH, you may find that its organized approach helps consolidate what you've read. And if it clears up a couple of key concepts about dial plans, AGI, configuration, ZAP, or whatever, which you might be fuzzy about, it's probably worth the price. In addition, the appendices are a useful reference guide. Ross C wrote: Just curious what everyone (as in, the people that have read it or use it) thinks about the O'Reilly Asterisk book. I'd really like to delve into the nitty gritty of Asterisk, but I'm getting kinda tired of swimming through forums and Google results. I've been reading the wiki off and on for about a week now, but I'm wondering if a book would be the way to go to get a solid foundation. My IT career for the past 10 years has been based off of learn-as-I-go methods, but I'd really like to learn asterisk the right way. I have a couple Asterisk servers up and running and in use, but they're very small systems (~10 extensions, connected to 3 or 4 pots lines). I have some clients that want to use VOIP, but they're bigger businesses, and I'm not yet comfortable enough to roll out a bigger system. So if there are any other methods for learning Asterisk that I should consider, please do tell! Any opinions (on the book or otherwise) appreciated. Thanks! -ross ___ --Bandwidth 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] Bonded ethernet ports and *
Hey all - I'm sure this has been done before, but I'm curious about how well it works.. Typically we have all our servers setup for dual fast/gig ethernet failover... I.e. bond0 slaves eth0 and eth1 and fails over between the two. This together with dual p/s and raid1'd(at least) drives provides for a pretty safe solution(aside from building up a second server). So I'm courious thoughts/expectations/issues with doing network failover... Probably is a moot point, but I thought I'd ask. Thanks!! Rolf Brusletto Denver, Co. ___ --Bandwidth 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] Patch zaptel.init to support debian
On 12/13/2005 07:32:10 AM, Kevin P. Fleming wrote: This script is completely unnecessary on Debian; just add the modules you wish to load into /etc/modules and they will be loaded at boot time. FYI the list. Using debian with linux 2.6 you don't do anything, the requsite module information is installed in /etc/modprobe.d/zaptel and it just works. Karl [EMAIL PROTECTED] Free Software: You don't pay back, you pay forward. -- Robert A. Heinlein ___ --Bandwidth 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] talking about : mISDN Caller ID problem
wath is the list of isdn cards supported by asterisk? anybody have the list or the link about that? __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth 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 book feedback
On 12/13/2005 09:45:09 AM, Ross C wrote: Just curious what everyone (as in, the people that have read it or use it) thinks about the O'Reilly Asterisk book. I am just getting started. The book works for me. My gripe is the license. I can't submit improvements where I ran into gotchas, so I don't run into them again. I know that by the next time I set things up I'll have forgotten most of what I did wrong. Karl [EMAIL PROTECTED] Free Software: You don't pay back, you pay forward. -- Robert A. Heinlein ___ --Bandwidth 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 book feedback
We made a review of it a while ago, if you wonder if you will like it, why not download the pdf and have a look for yourself ? http://www.asteriskguru.com/review.php Zoa. John Biundo wrote: The book is a great *starting* point, IMHO. If you've spent a considerable amount of time reading other sources, you probably won't find much new information in the book. OTOH, you may find that its organized approach helps consolidate what you've read. And if it clears up a couple of key concepts about dial plans, AGI, configuration, ZAP, or whatever, which you might be fuzzy about, it's probably worth the price. In addition, the appendices are a useful reference guide. Ross C wrote: Just curious what everyone (as in, the people that have read it or use it) thinks about the O'Reilly Asterisk book. I'd really like to delve into the nitty gritty of Asterisk, but I'm getting kinda tired of swimming through forums and Google results. I've been reading the wiki off and on for about a week now, but I'm wondering if a book would be the way to go to get a solid foundation. My IT career for the past 10 years has been based off of learn-as-I-go methods, but I'd really like to learn asterisk the right way. I have a couple Asterisk servers up and running and in use, but they're very small systems (~10 extensions, connected to 3 or 4 pots lines). I have some clients that want to use VOIP, but they're bigger businesses, and I'm not yet comfortable enough to roll out a bigger system. So if there are any other methods for learning Asterisk that I should consider, please do tell! Any opinions (on the book or otherwise) appreciated. Thanks! -ross ___ --Bandwidth 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] chan_capi AVM C2
stéphane plichon wrote: Hi all, currently i running * 1.0.9 with chan_capi 0.3.5 my first problem is: in incoming call, when BCHAN is full in contr1 incoming call on contr2 are not answered with error : chan_capi.c:1953 in capi_handle_msg: received a call waiting CONNECT_IN but if use different msn in capi.conf incoming call works on both controler ok, now working, but i get only ring on third call -- Stephane Plichon jabber: [EMAIL PROTECTED] ~ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Partial PRI pass thru?
Matt Burleigh schrieb: Thanks for the responses. I guess the next step is to get a Digium TE210P. Are there any other 2 port PRI cards anyone would recommend for *? Yes - there are 2xPRI cards based on the CologneChip HFC-E1 chip and the A102u cards from the Canada based manufacturer SANGOMA: I personally prefer the Sangoma cards because of the good support (wich we needed only very little compared to our problems with other brands) and the stability we experienced in our setups. On top of this they are field upgradeable and work in 3,3v and 5v PCI slots and low profile cases. You can find more information on www.sangoma.com Chris ___ --Bandwidth 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] Skips and Pops in Call Recordings
What codec are the calls? What codec are you recording in? I would try some non-Dell hardware, I would also try a less bloated Linux Distro, something like Slackware, just to see if that had any effect. And make sure you use the megaraid2 linux drivers. MATT--- On 12/13/05, Matt Roth [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, Need some more information here: - hardware specs (including what kind of hard drives?) The Asterisk server is a Dell PowerEdge 6850 with the following specs. Please note that we are NOT recording to the hard drive. We are recording to a RAM disk as detailed here (http://voip-info.org/wiki/view/Asterisk+Dimensioning) under the heading 512 simultaneous SIP-to-SIP calls with Digital Recording. Unfortunately, the scalability tests we did at that time assumed that if call quality was good, so was the quality of the recording. Processor: Quad Intel Xeon 3.16GHz/1MB Cache Memory: 20 GB DDR2 400MHZ Single Ranked DIMMs (4 GB System / 16 GB RAM Disk) Hard Drive: Two 73GB, U320, SCSI, 1IN 15K Configured in a RAID 1 (Mirrored) Hard Drive Controller: Embedded RAID - PERC4 Integrated (Driver: megaraid_mm, megaraid_mbox) Everything else: http://www.dell.com/downloads/global/products/pedge/en/PE6850_specs.pdf - Linux kernel version 2.6.12-1.1376_FC3smp (Fedora Core 3). - running Xwindows? No. - Asterisk version ABE-A.2-beta (Asterisk Business Edition A.2 beta). - kind of calls you are recording (Zap, SIP, IAX, Meetme, ...) Calls originate on the PSTN and are handled by a Cisco AS5400 Universal Gateway that is a SIP peer of Asterisk. The AS5400 converts the calls from TDM channels to VoIP (SIP) channels before sending them to Asterisk. The Asterisk dialplan then routes them to one of our agents, who are using SNOM 320 VoIP (SIP) phones. Essentially all of our calls are SIP-to-SIP, with absolutely no protocol bridging or transcoding occurring on the Asterisk server. The Asterisk server handles the following major tasks: - Routing calls through the dialplan to (dynamic) agents in the appropriate queues. - Adding/removing agents to/from queues via AddQueueMember and RemoveQueueMember (NO static agents!). - Recording calls via the Monitor application directly to RAM disk. Calls are moved to a remote machine for mixing. - ChanSpy-based quality assurance of calls. Neither ChanSpy nor the quality of the calls themselves is affected by the problem. - how many recordings at once Anywhere from 5 to 30 concurrent recordings. This is not our planned peak, but it's where we've experienced the problem so far. We have not yet determined if the number of concurrent recordings is an issue, but we are considering it. We also haven't determined if the problem gets worse as the number of recordings increases, but it definitely exists throughout that entire range. In my experience, HyperThreading does not cause recording problems, it's usually a disk issue. When we had issues, switching to fast SCSI drives on a MegaRAID 320-1 with the megaraid2 linux driver solved all of our problems(skips and clicks/pops) The disk issues also directly interfere with call quality, as our previous scalability tests showed. Digium seems to think that the issue is scaling (some resource contention that causes a bit of audio to be unavailable when the write occurs). I see their point, but given our hardware and the current call volume I'm not completely sold on it. Could it be a configuration issue (file handles, interrupts, etc...)? MATT--- Colin Anderson wrote: Matt Roth: FWIW, I am recording 1000-1500 calls a day (as of 8:52PM today, 1482 calls!) of various length on my Netfinity with the onboard IBM RAID controller in RAID 5 Ultra 320 SCSI with suprisingly good quality. As the other Matt indicated, maybe what is needed here is an intelligent controller to offload some of the chore. No definite solution here, but at least it's another data point to compare. I appreciate any information contributed by list users. It's by far the most valuable resource available to me. On 12/12/05, Matt Roth [EMAIL PROTECTED] wrote: List users, I'm using the Monitor application to record calls. Most of the recordings are audible, but contain skips accompanied by a popping sound. Sometimes they are isolated, sometimes they appear in groups. Call quality is excellent and seems unaffected by whatever is causing this problem. If anyone has experienced this problem before, I'd appreciate if you'd share what the source was and any tips on eliminating it. I contacted Digium tech support and they suggested turning off hyperthreading. I have done that, but I won't know if it improved things until tomorrow. The machine is running at a moderate call volume and is always at least 90% idle. I'm not seeing any Avoided deadlock messages in the logs. If you need any more information, I'd be
[Asterisk-Users] Tellabs manuals
Does anybody have a Tellabs manual for: * 253c shelf. the complete model number is: 81.0253c * 2572 Echo Canceller card, complete model number is: 81.2572 I know the wiki has got lots of info on it, but I'm trying to get the original docs from Tellabs. Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk book feedback
Ross C wrote: Just curious what everyone (as in, the people that have read it or use it) thinks about the O'Reilly Asterisk book. I'd really like to delve into the nitty gritty of Asterisk, but I'm getting kinda tired of swimming through forums and Google results. I've been reading the wiki off and on for about a week now, but I'm wondering if a book would be the way to go to get a solid foundation. My IT career for the past 10 years has been based off of learn-as-I-go methods, but I'd really like to learn asterisk the right way. I have a couple Asterisk servers up and running and in use, but they're very small systems (~10 extensions, connected to 3 or 4 pots lines). I have some clients that want to use VOIP, but they're bigger businesses, and I'm not yet comfortable enough to roll out a bigger system. So if there are any other methods for learning Asterisk that I should consider, please do tell! Any opinions (on the book or otherwise) appreciated. Thanks! Another resource you might want to consider is Ted Wallingford's Switching to VoIP: http://www.oreilly.com/catalog/switchingvoip/ It uses Asterisk extensively in examples and provides good coverage of concepts like QoS, codecs, etc. that are important considerations in many Asterisk deployments. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca www.gabcast.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] mISDN chan_misdn on Fedora Core 4 - problems
Hi Everyone, I'm trying to get chan_misdn working with asterisk. Currently I'm using two seperate * boxes with chan_capi and one AVM Fritz card per box and I'd love to get one box doing the job (plus I'm hoping that echo cancellation is better in chan_misdn). I have this error when I start asterisk with chan_misdn configured: - [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) mISDN_close: fid(8) isize(131072) inbuf(0xb7dae008) irp(0xb7dae008) iend(0xb7dae008) == Parsing '/etc/asterisk/misdn.conf': Found P[ 0] Got: 1 from get_ports Segmentation fault (core dumped) So far I've managed to do this: Install the 2.6.14.3 kernel Install the mISDN kernel patches and compile the kernel with the latest mISDN (via make menuconfig, etc) Compile and install the latest CVS mqueue mISDN and mISDN code (only the mqueue release seemed to work with GCC-4 which comes with Fedora Core 4) Compile and install the latest CVS chan_misdn and install it Loaded the mISDN modules (nice stuff showing in dmesg), did the /etc/rc.d/init.d/misdn-init start (after the config argument) and did the funny mkdir -p /dev/capi, mount /dev/capi after defining the capifs stuff in fstab. I haven't a clue what I'm doing here with capifs - beats me! I did get an error after I ran the misdn-init start - its says FATAL: Error inserting mISDN_dsp (/lib/modules/2.6.14.3/extra/mISDN_dsp.ko): Unknown symbol in module, or unknown parameter (see dmesg). I don't know why I'm getting this error because everything was installed with this kernel - no old stuff lying around I think. When I start asterisk with asterisk -vvvgc I get the text and the segmentation fault as above. Anyone any ideas? (apart from go buy other hardware :). Thanks very much! Derek -- Derek Conniffe Rivertower Ltd DID Number: 01 440 1806 (International: 00 353 1 440 1806) Ireland: (Freephone) 1800 719 400 Ireland: (Local) 01 440 1800 United Kingdom: 0870 068 2368 International: 00 353 1 440 1800 Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 201 0085 (International: 00 353 1 201 0085) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.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] Asterisk Feature Request: app_bridgeme
Hi all,
I'm currently involved in a project where the meetme application is used
extensively forbridging calls between an operator and 2 or more parties. One
of the features that we require is the ability to pass DTMF signals from any
party in the bridge to a pre-specified bridge connected channel.
I will explain this using the following scenario:
1. User A calls the Asterisk box and is put into the bridge
2. An operator is notified that a user is waiting in the bridge and connects to
the bridge.
3. Now, the operator originates 1 or more calls that would be connected to the
bridge. One of these calls is designated with an environment variable saying
${OUTBOUND_BRIDGE}.
4. Any channel connected to the bridge, when pressing a DTMF key would then have
that DTMF signal transmitter to the ${OUTBOUND_BRIDGE} channel, resulting in
the ability to bridge several users into a single outbound channel and to
proxy DTMF's to it.
I'm aware that is a fairly funky usage for an application, but if someone has
a better way of doing this, I'm willing to learn. Oh, btw, one small remark, if
you were about to say: use queues and call park, my answer would be: I
can't, I have no control over the extensions. I basically interconnect via a
PRI to an external Avaya CTI system, thus, I have no way of implementing queues
in the system - due to constraints by the Avaya CTI system.
Regards,
Nir Simionovich
___
--Bandwidth 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] Very high memory consumption when high number of calls are processed
We are running a number of hosted SIP-only PBXs, and we do have memory problems with some of them. The servers have typically 512 MB RAM, and in some of the servers the Asterisk usage goes up from a couple of percent (at restart in the morning) to more than 82% after periods with a high number of calls processed. At a call rate of 100 calls an hour, the memory consumption growth is around 50MB an hour. At the end of the busy period we sometimes get Fork failed: Cannot allocate memory error, and calls to the server are rejected. At that time, several hundreds of MB of the virtual memory has often been taken into use, and the so-called free memory is down at a few MB or even kB. After restart, the memory is freed up. The dial plan is complex using OBCD calls to a MySQL astdb table. Most calls are queued. We use mechanisms such as hint, qualify and setGroup. We have a separate Flash Operators Panel server, which communicates with the server through the manager API. We have tried to change the unixOBCD driver, but the memory consumption did not change. The version used is 1.0.10, STABLE. Running on Debian RC 2. We use a FLASH disk with 4GB capacity. My main issue is: can we avoid these problems by changing design parameters somewhere? . or do we just have to put more RAM into the servers? ___ --Bandwidth 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] chan_capi AVM C2
On Tue, 13 Dec 2005, stéphane plichon wrote: stéphane plichon wrote: Hi all, currently i running * 1.0.9 with chan_capi 0.3.5 my first problem is: in incoming call, when BCHAN is full in contr1 incoming call on contr2 are not answered with error : chan_capi.c:1953 in capi_handle_msg: received a call waiting CONNECT_IN but if use different msn in capi.conf incoming call works on both controler ok, now working, but i get only ring on third call Make sure you have 2 separate sections in your capi.conf, one for each controller. 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
[Asterisk-Users] CID name number contain unwanted quotes in CDR
On a recent install
of ast 1.2 (b1) I noticed something strange in the CDR records (in mysql).
The caller ID name and number contained extra quotes for calls outbound
(inbound was fine).
Below is an example
of the extensions.conf excerpt, and an excerpt from my sql. Can anyone
explain how to avoid the extra quotes in the CDR record?
EXTENSIONS.CONF:
[globals]
MYNAME="Bob
Steve"
MYNUMBER="123456789"
exten = _123,1,Set(CALLERID(name)=${MYNAME})exten
= _123,2,Set(CALLERID(number)=${MYNUMBER})
MYSQL
CDR:
|
|
| |
"11" |22 |
menuhome | ""Bob Steve"" "1234567898"
|
SIP/290-0334
| IAX2/UNLIMITEL4-4 |
Dial
| IAX2/UNLIMITEL4/22|60|r
| 2005-12-13 12:47:03 | 482
| 465 | ANSWERED
| 3 |
Notice the
""Bob Steve"" ? Any ideas?
Thanks,
Michele
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_addon_mysql can't find libmysqlclient.so
Hello list! I had a problem while trying to build asterisk-addons, but noticed some paths specified in the Makefile didn't fit my system. So I modified Makefile for it to look for MySQL includes and libs on the following locations: /usr/local/mysql/include/mysql /usr/local/mysql/lib/mysql Now when trying "make" it works fine, and "make install" too. ;) But when I add cdr_addon_mysql.so on modules.conf for Asterisk to load it, Asterisk refuses tu come up saying: Dec 13 12:19:29 WARNING[4112]: loader.c:325 __load_resource: libmysqlclient.so.15: cannot open shared object file: No such file or directoryDec 13 12:19:29 WARNING[4112]: loader.c:499 load_modules: Loading module cdr_addon_mysql.so failed! Is there any other place in which I should specify the diffetent locations for my system? How can I fix this? Thanks for your help. Cheers! ___ --Bandwidth 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] WIFI Phones
I'm looking for iax2 wifi phones, do you know where i can buy them? Thanks Mario ___ --Bandwidth 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] Skips and Pops in Call Recordings
Matt, The calls are u-Law. The format of the recordings is PCM. Is this correct to prevent transcoding the recording? We've noloaded all other codecs, so I don't believe that transcoding is occurring. I've only ever seen "show translation" generate the following output: immlx15*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - - - - - - - - - - ulaw - - - - - - 1 - - - - alaw - - - - - - - - - - - g726 - - - - - - - - - - - adpcm - - - - - - - - - - - slin - - 1 - - - - - - - - lpc10 - - - - - - - - - - - g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc - - - - - - - - - - - Any suggestions on hardware? Are you talking the entire server or components? I'll look into the megaraid2 drivers, but I'm interested in knowing how they come into play when recording to a RAM disk. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Matt Florell wrote: What codec are the calls? What codec are you recording in? I would try some non-Dell hardware, I would also try a less bloated Linux Distro, something like Slackware, just to see if that had any effect. And make sure you use the megaraid2 linux drivers. MATT--- On 12/13/05, Matt Roth [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, Need some more information here: - hardware specs (including what kind of hard drives?) The Asterisk server is a Dell PowerEdge 6850 with the following specs. Please note that we are NOT recording to the hard drive. We are recording to a RAM disk as detailed here (http://voip-info.org/wiki/view/Asterisk+Dimensioning) under the heading "512 simultaneous SIP-to-SIP calls with Digital Recording". Unfortunately, the scalability tests we did at that time assumed that if call quality was good, so was the quality of the recording. Processor: Quad Intel Xeon 3.16GHz/1MB Cache Memory: 20 GB DDR2 400MHZ Single Ranked DIMMs (4 GB System / 16 GB RAM Disk) Hard Drive: Two 73GB, U320, SCSI, 1IN 15K Configured in a RAID 1 (Mirrored) Hard Drive Controller: Embedded RAID - PERC4 Integrated (Driver: megaraid_mm, megaraid_mbox) Everything else: http://www.dell.com/downloads/global/products/pedge/en/PE6850_specs.pdf - Linux kernel version 2.6.12-1.1376_FC3smp (Fedora Core 3). - running Xwindows? No. - Asterisk version ABE-A.2-beta (Asterisk Business Edition A.2 beta). - kind of calls you are recording (Zap, SIP, IAX, Meetme, ...) Calls originate on the PSTN and are handled by a Cisco AS5400 Universal Gateway that is a SIP peer of Asterisk. The AS5400 converts the calls from TDM channels to VoIP (SIP) channels before sending them to Asterisk. The Asterisk dialplan then routes them to one of our agents, who are using SNOM 320 VoIP (SIP) phones. Essentially all of our calls are SIP-to-SIP, with absolutely no protocol bridging or transcoding occurring on the Asterisk server. The Asterisk server handles the following major tasks: - Routing calls through the dialplan to (dynamic) agents in the appropriate queues. - Adding/removing agents to/from queues via AddQueueMember and RemoveQueueMember (NO static agents!). - Recording calls via the Monitor application directly to RAM disk. Calls are moved to a remote machine for mixing. - ChanSpy-based quality assurance of calls. Neither ChanSpy nor the quality of the calls themselves is affected by the problem. - how many recordings at once Anywhere from 5 to 30 concurrent recordings. This is not our planned peak, but it's where we've experienced the problem so far. We have not yet determined if the number of concurrent recordings is an issue, but we are considering it. We also haven't determined if the problem gets worse as the number of recordings increases, but it definitely exists throughout that entire range. In my experience, HyperThreading does not cause recording problems, it's usually a disk issue. When we had issues, switching to fast SCSI drives on a MegaRAID 320-1 with the megaraid2 linux driver solved all of our problems(skips and clicks/pops) The disk issues also directly interfere with call quality, as our previous scalability tests showed. Digium seems to think that the issue is scaling (some resource contention that causes a bit of audio to be unavailable when the write occurs). I see their point, but given our hardware and the current call volume I'm not completely sold on it. Could it be a configuration issue (file handles, interrupts, etc...)? MATT--- Colin Anderson wrote: Matt Roth: FWIW, I am recording 1000-1500 calls a day (as of 8:52PM today, 1482 calls!) of various length on my Netfinity with the onboard IBM RAID controller in RAID 5 Ultra 320 SCSI with suprisingly good quality. As the other Matt indicated, maybe what is needed here is an intelligent controller to offload some of the chore. No definite
Re: [Asterisk-Users] Tellabs manuals
I have a 253A manual out of the big three manuals I have, but not the echo canceller. - James C F wrote: Does anybody have a Tellabs manual for: * 253c shelf. the complete model number is: 81.0253c * 2572 Echo Canceller card, complete model number is: 81.2572 I know the wiki has got lots of info on it, but I'm trying to get the original docs from Tellabs. Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] queues music on hold
Hello list,
I have the following problem.
The behavior of music on hold is not constant on my
queues... Sometimes it plays well, sometimes it becomes mute in the middle of
the wait and sometimes it doesn't even start.
mpg123 is installed on my server.
Is there something I am missing???
Thank you!Dov
---
queues.conf
[infocadastrais]leavewhenempty=yesjoinempty=nomusiconhold=fila
strategy=leastrecent
timeout=14
eventwhencalled=yesmaxlen=0retry=0wrapuptime=5servicelevel=45monitor-format=wav49monitor-join=yesannounce-holdtime=no
member = Agent/5132
agents.conf
[agents]
autologoff=
ackcall=no
wrapuptime=5000
musiconhold = fila
recordagentcalls=no
updatecdr=yes
group =1
agent = 5132,1234
extensions.conf
exten = cobrancainfo,1,NoOp(Ligacao para Fila
de Info Cadastrais)exten = cobrancainfo,2,SetVar(prioridade=0)exten
= cobrancainfo,3,SetCIDName(CobrancaInfoCadastrais ${CALLERIDNAME})exten
= cobrancainfo,4,SetVar(QUEUE_PRIO=${prioridade})exten =
cobrancainfo,5,Answerexten =
cobrancainfo,6,Queue(infocadastrais|tT|||45)exten =
cobrancainfo,7,Wait(3)exten = cobrancainfo,8,VoiceMail(u501)exten
= cobrancainfo,9,Hangup
musiconhold.conf
[classes]fila =
mp3:/var/lib/asterisk/mohmp3/defaultfila
[moh_files]fila
=/var/lib/asterisk/mohmp3/defaultfila,r
And on /var/lib/asterisk/mohmp3/defaultfila I have
3 valid MP3 files.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tellabs manuals
Can you please email it to me? off list. What are the other 2 manuals? Thank You On 12/13/05, James Armstrong [EMAIL PROTECTED] wrote: I have a 253A manual out of the big three manuals I have, but not the echo canceller. - James C F wrote: Does anybody have a Tellabs manual for: * 253c shelf. the complete model number is: 81.0253c * 2572 Echo Canceller card, complete model number is: 81.2572 I know the wiki has got lots of info on it, but I'm trying to get the original docs from Tellabs. Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Channel 0/1, span 1 got hangup request
Steve Totaro wrote: What are you doing in between making changes and testing the changes? After changing settings I reboot system! Really. :) Because other actions have no effect. Also reboot, too.. Thanks, Steve Just a couple guesses on things to try. Zapata.conf 1. Changing switchtype variables (doubtful but give it a try). 2. Add a variable to define pridialplan (I remember someone setting this to unknown to solve a similar issue) Try pridialplan=unknown and/or prilocaldialplan=local or some other valid option. A do this config, but no effects Zaptel.conf 1. span=1,1,5,ccs,hdb3 I think that your dial statement or the pridialplan is your issue. If you look at the debug info Here is something suspicious: -- Called g1/100 unless 100 is the number you are trying to dial outbound. If the above fails, then try below. Try tweaking your settings here like span=1,0,0,ccs,hdb3 What is the provider expecting? No effect on settings: span=1,0,0,ccs,hdb3 span=1,1,5,ccs,hdb3 span=1,2,4,ccs,hdb3 Thanks, Steve Dear Users, I have an Digium Wildcard TE110P T1/E1 Card inserted in Linux box runnig Asterisk 1.2.0 All incoming calls from E1 interface to SIP-phone goes exellent, but calls from SIP to E1 gives the errors: -- Executing Dial(SIP/anton-6cf4, Zap/g1/100) in new stack -- Making new call for cr 32775 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 05 41 6e 74 6f 6e] Display (len= 5) +)[EMAIL PROTECTED]@[EMAIL PROTECTED]@│@[ Anton ] [6c 0d 21 81 38 34 37 37 33 36 31 38 31 38 33] Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '84773618183' ] [70 04 a1 31 30 30] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '100' ] -- Called g1/100 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup request Dec 5 15:30:12 WARNING[30946]: app_dial.c:706 wait_for_answer: Unable to forward voice NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == No one is available to answer at this time (1:0/0/0) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 80 d1] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (81), class = Invalid message (5) ] -- Processing IE 8 (cs0, Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Timeout on SIP/anton-6cf4 == CDR updated on SIP/anton-6cf4 -- Executing Hangup(SIP/anton-6cf4, ) in new stack /etc/zaptel.conf span=1,1,5,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = nl defaultzone=nl /etc/asterisck/zapata.conf [trunkgroups] [channels] language=en signalling=pri_cpe switchtype=euroisdn echocancel=32 echocancelwhenbridged=yes usecallerid=yes callerid=asreceived transfer=yes overlapdial=yes cancallforward=yes group=1 context=zapata channel = 1-15,17-31 Has anybody resolve this problem? -- SY, Anton V Bakulev. MIPT-telecom. [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users
[Asterisk-Users] format_mp3 uninstalling mpg123
Hi all, In order to fix my problem with music on hold I would like to test format_mp3, that comes with asterisk-addons package. For that, the wiki says "Be sure to remove mpg123 from your system (this may attribute to 'Request to schedule in the past!?!?!' messages). Now you are set! " How do I uninstall mpg123? Thank you Dov ___ --Bandwidth 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] Skips and Pops in Call Recordings
Hello, To see if it's somehow the recording throughput that's the problem I'd suggest trying recording in GSM just as a test and see how that is. As for the hardware, just try a machine with no Dell parts in it. I've talked to many Asterisk users who's problems went away when they switched to something that wasn't a Dell. MegaRAID2 might help just because it's another reduction in the overall data that flows over the PCI bus. It's faster and more streamlined than the original megaraid driver and it can't hurt to try it. MATT--- On 12/13/05, Matt Roth [EMAIL PROTECTED] wrote: Matt, The calls are u-Law. The format of the recordings is PCM. Is this correct to prevent transcoding the recording? We've noloaded all other codecs, so I don't believe that transcoding is occurring. I've only ever seen show translation generate the following output: immlx15*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - - - - - - - - - - ulaw - - - - - - 1 - - - - alaw - - - - - - - - - - - g726 - - - - - - - - - - - adpcm - - - - - - - - - - - slin - - 1 - - - - - - - - lpc10 - - - - - - - - - - - g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc - - - - - - - - - - - Any suggestions on hardware? Are you talking the entire server or components? I'll look into the megaraid2 drivers, but I'm interested in knowing how they come into play when recording to a RAM disk. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Matt Florell wrote: What codec are the calls? What codec are you recording in? I would try some non-Dell hardware, I would also try a less bloated Linux Distro, something like Slackware, just to see if that had any effect. And make sure you use the megaraid2 linux drivers. MATT--- On 12/13/05, Matt Roth [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, Need some more information here: - hardware specs (including what kind of hard drives?) The Asterisk server is a Dell PowerEdge 6850 with the following specs. Please note that we are NOT recording to the hard drive. We are recording to a RAM disk as detailed here (http://voip-info.org/wiki/view/Asterisk+Dimensioning) under the heading 512 simultaneous SIP-to-SIP calls with Digital Recording. Unfortunately, the scalability tests we did at that time assumed that if call quality was good, so was the quality of the recording. Processor: Quad Intel Xeon 3.16GHz/1MB Cache Memory: 20 GB DDR2 400MHZ Single Ranked DIMMs (4 GB System / 16 GB RAM Disk) Hard Drive: Two 73GB, U320, SCSI, 1IN 15K Configured in a RAID 1 (Mirrored) Hard Drive Controller: Embedded RAID - PERC4 Integrated (Driver: megaraid_mm, megaraid_mbox) Everything else: http://www.dell.com/downloads/global/products/pedge/en/PE6850_specs.pdf - Linux kernel version 2.6.12-1.1376_FC3smp (Fedora Core 3). - running Xwindows? No. - Asterisk version ABE-A.2-beta (Asterisk Business Edition A.2 beta). - kind of calls you are recording (Zap, SIP, IAX, Meetme, ...) Calls originate on the PSTN and are handled by a Cisco AS5400 Universal Gateway that is a SIP peer of Asterisk. The AS5400 converts the calls from TDM channels to VoIP (SIP) channels before sending them to Asterisk. The Asterisk dialplan then routes them to one of our agents, who are using SNOM 320 VoIP (SIP) phones. Essentially all of our calls are SIP-to-SIP, with absolutely no protocol bridging or transcoding occurring on the Asterisk server. The Asterisk server handles the following major tasks: - Routing calls through the dialplan to (dynamic) agents in the appropriate queues. - Adding/removing agents to/from queues via AddQueueMember and RemoveQueueMember (NO static agents!). - Recording calls via the Monitor application directly to RAM disk. Calls are moved to a remote machine for mixing. - ChanSpy-based quality assurance of calls. Neither ChanSpy nor the quality of the calls themselves is affected by the problem. - how many recordings at once Anywhere from 5 to 30 concurrent recordings. This is not our planned peak, but it's where we've experienced the problem so far. We have not yet determined if the number of concurrent recordings is an issue, but we are
Re: [Asterisk-Users] format_mp3 uninstalling mpg123
On Tue, Dec 13, 2005 at 05:12:44PM -0200, Dov Bigio wrote: For that, the wiki says Be sure to remove mpg123 from your system (this may attribute to 'Request to schedule in the past!?!?!' messages). Now you are set! How do I uninstall mpg123? How did you install mpg123? If you installed it with the package management system, then use the package management system on your OS to remove it. If you installed it manually, you'll need to remove it manually. To actually allow format_mp3 to work you also need to change musiconhold.conf from mode=quietmp3 to mode=files. Hope that helps --- Gil Kloepfer [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000: Dual Registrations?
Kerry Garrison wrote: We just posted an updated guide to the SPA-3000 a few days ago. The example uses AMP but all the settings are there: It was exactly that example that I was using to start with. Using the setup just as below, I get the following error: chan_sip.c:10823 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.113' - Username/auth name mismatch If I comment out the stanza named spa3000 in sip.conf (below), and set the Register setting for the PSTN screen to No things work fine. But when calls come in on the PSTN line, Asterisk uses the *peer* setting from-pstn for the connection. So to reiterate things are working, but I'm not doing the two registrations like I thought should be the way it would be done. Thanks. B. *** Firmware 3.1.7 Here are the configs I'm using. My Asterisk server is at 192.168.1.1, and the SPA is at 192.168.1.113: On the SPA-3000: Line 1 Page, Proxy and Registration Settings: Proxy: 192.168.1.1 Register: Yes Use Outbound Proxy: No Use OB Proxy in Dialing: Yes Make Call w/o Reg: No Ans Call w/o Reg: No (Others settings left at factory defaults) Subscriber Registration: User ID: spa3k Password: Display Name: Asterisk Auth ID: spa3k Use Auth ID: No SIP Settings: Port 5060 PSTN Line Page, Proxy and Registration Settings: Proxy: 192.168.1.1 Register: Yes Use Outbound Proxy: No Use OB Proxy in Dialing: No Make Call w/o Reg: Yes Ans Call w/o Reg: Yes (Other settings at fact defaults) Subscriber Registration: User ID: spa3000 Password: Display Name: blank Auth ID: blank Use Auth ID: No SIP Settings: Port 5061 * Asterisk Configs, in sip.conf: [spa3000] type=user username=spa3000 secret= host=dynamic context=testcontext port=5061 canreinvite=no insecure=very disallow=all allow=ulaw [pstn-spa] type=peer username=spa3000 secret= host=192.168.1.113 context=testcontext port=5061 canreinvite=no insecure=very disallow=all allow=g726,ulaw dtmfmode=info [spa3k] type=friend username=spa3k secret= host=dynamic context=testcontext mailbox=101 canreinvite=no callerid=My Name123-456-7890 callgroup=1 pickupgroup=1 dtmfmode=rfc2833 disallow=all allow=g726,ulaw ___ --Bandwidth 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 on having asterisk put calls into a meetme.
If I'm in a meetme conference, what would I need to do to have some call files make calls and connect them into the meetme conference with me? ___ --Bandwidth 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] Partial PRI pass thru?
I'd recommend the Digium dual port cards - generation 2 card are excellent and the support we receive superb. And it supports Digiums support and development of Asterisk - Sangomas contribution is token if any.Rob On 12/13/05, Christian Victor [EMAIL PROTECTED] wrote: Matt Burleigh schrieb: Thanks for the responses. I guess the next step is to get a Digium TE210P. Are there any other 2 port PRI cards anyone would recommend for *?Yes - there are 2xPRI cards based on the CologneChip HFC-E1 chip and the A102u cards from the Canada based manufacturer SANGOMA:I personally prefer the Sangoma cards because of the good support (wichwe needed only very little compared to our problems with other brands)and the stability we experienced in our setups. On top of this they are field upgradeable and work in 3,3v and 5v PCI slots and low profile cases.You can find more information on www.sangoma.comChris___ --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] format_mp3 uninstalling mpg123
On Tue, Dec 13, 2005 at 05:12:44PM -0200, Dov Bigio wrote: For that, the wiki says Be sure to remove mpg123 from your system (this may attribute to 'Request to schedule in the past!?!?!' messages). Now you are set! How do I uninstall mpg123? How did you install mpg123? If you installed it with the package management system, then use the package management system on your OS to remove it. If you installed it manually, you'll need to remove it manually. Actually I did it manually (tar -xvzf)... but I am not sure which files I have to delete manually.. is there an explanation somehere? I couldn't find it on Google... To actually allow format_mp3 to work you also need to change musiconhold.conf from mode=quietmp3 to mode=files. This is new for me... I didn't find any information on this mode parameter... Should it be put under [classes] or [moh_files] in musiconhold.conf??? Hope that helps Thank you very much! Dov --- Gil Kloepfer [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
