Re: [asterisk-users] Refund from SellVoip?
I got money back around 6 months ago . It was a via paypal claim and hey didn't reply till paypal's deadline so i got $30 back . On 17/03/07, Ira [EMAIL PROTECTED] wrote: At 02:32 PM 3/16/2007, you wrote: You were able to cancel service with Sellvoip? That's impressive, that Actually, it's Voxee I tried to cancel and failed. I still use SellVOIP and it mostly works but support is a problem. I'm starting to use using Telasip more though as they work and have a POP only 19ms from here, a big advantage. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip tunnel
try changing bindport of asterisk from 5060 to something else . On 09/03/07, Pezhman Lali [EMAIL PROTECTED] wrote: Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP??
check your sip.conf and make sure it has allow=ulaw and allow=alaw line ( you can even remove gsm to test it it works fine or not ) On 09/02/07, Florea Igor [EMAIL PROTECTED] wrote: ip_pbx2 is not asterisk, it knowk only PCMU,PCMA,g723,g729 On Thursday 08 February 2007 19:00, Vicky wrote: config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc ) On 08/02/07, Florea Igor [EMAIL PROTECTED] wrote: Hi, I'm new to *,so i apologize for stupid questions. I'm having problem with this arhitecture: I'm calling asterisk from behind a NAT(sjphone user) with a low band so I'm using GSM codec. In extensions.conf I have: exten = 337,1,Dial(SIP/99@ip_pbx2) so when i dial 337 from sjphone Asterisk is colling 99 on ip_pbx2. RTP stream between sjphone and Asterisk are ok (GSM). The problem is rtp packets from Asterisk to ip_pbx2 are also GSM although ip_pbx2 sip is telling asterisk It only knows codec 0 Is this a config problem or a bug? Igor, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- There are 10 kinds of people in the world: those who know binary and those who don't. Igor Florea Ing. dezvoltare Phone: +40 21 232 04 24 Fax: +40 21 232 31 56 Local time: GMT+2 www.topex.ro ___ --Bandwidth 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] BindPort
I also encountered the problem of port 5060 being blocked by some user's isp and redirected port 5098 to 5060 but still asterisk wasnt able to detect hangup properly and had load of voice problems ( lot of nat involved and softphones were being used ) so i made asterisk listen on 5098 and redirected port 5060 to 5098 via iptables and it solved all problems ( port block users were able to use 5098 completely while other users had no problems with 5060 too ) . Try this method if u get some voice problems for port blocked users . On 09/02/07, Il Neofita [EMAIL PROTECTED] wrote: The point is to use more than one port, I think the only way is to use the redirect from iptables On 2/6/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Ciao, just change port value in sip.conf . Giorgio Il Neofita wrote: Hi, I was wondering if it is possible to set asterisk in order to listen to different ports for the sip or I need to do this operation with iptables? All of this since some time the port 5060 is blocked. 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 ___ --Bandwidth 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] Linux Kernel Timer Frequency and Asterisk
1000 Hz is recommended if you use lot of meetme channels ( and maybe iax trunking ? ) without a hardware timer . On 08/02/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 7 Feb 2007, Mark Coccimiglio wrote: Ok here is a real geek question, I building my own linux kernel for my asterisk system and came across the kernel setting for the timer frequency. I have one of 3 hardcode choices 100Hz, 250 Hz and 1000Hz. From what I understand the default Freq was changed from 100Hz in kernel 2.4 to 1000Hz (1KHz) in kernel 2.6. Timing is a BIG issue in asterisk with all the TDM and zap channel stuff. My guess is to go with the lower 100 or 250 Hz option but that is only a guess. The 1KHz sounds like it will conflict with the Zap 1khz timer (or am I wrong about that). Does anyone know what the prefered settings are for Trixbox or AsteriskNOW (or the asterisk code forks e.g. OpenPBX)? Please let me know what your experience has been. I always compile custom kernels and have been using 1KHz in all my systems which are for anything vaguely interactive. Most of my asterisk systems are 1GHz processors but I have a small handful which are 533MHz and all are working just fine. You're not using a recently kernel then ;-) 2.6.20 offers 300Hz too (supposedly good for video applications) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP??
config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc ) On 08/02/07, Florea Igor [EMAIL PROTECTED] wrote: Hi, I'm new to *,so i apologize for stupid questions. I'm having problem with this arhitecture: I'm calling asterisk from behind a NAT(sjphone user) with a low band so I'm using GSM codec. In extensions.conf I have: exten = 337,1,Dial(SIP/99@ip_pbx2) so when i dial 337 from sjphone Asterisk is colling 99 on ip_pbx2. RTP stream between sjphone and Asterisk are ok (GSM). The problem is rtp packets from Asterisk to ip_pbx2 are also GSM although ip_pbx2 sip is telling asterisk It only knows codec 0 Is this a config problem or a bug? Igor, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: On what distribution is www.asterisknow.com basedon ?
You can easily recompile asterisk with mysql logging enabled also use all add-ons u can use on debian and any other distro .. On 08/02/07, Chris Earle [EMAIL PROTECTED] wrote: I'm tempted to rebuild my asterisk network with AsteriskNow - my question is, can you ADD anything to it? i.e. cdr_mysql logging? I thought I saw it didn't have that And how does it handle the hardware? I don't use digium cards in all of my servers because of country issues (Junghanns in Germany, Sangoma in UK, etc) If it's expandable through usual package addons etc, then it would seem there is alot of added value because of the increased EASE of administration over your well-maintained debian box Thoughts? -- Chris Maxim Veksler [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hello Asteriskies, Has someone tried www.asterisknow.com ? What is the package manager used? And what is the added value compared to the well maintained debian based asterisk ? Thanks, -- Cheers, Maxim Veksler Free as in Freedom - Do u GNU ? ___ --Bandwidth 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] Problem with Voipjet ...
Voipjet locks $1.2 per running call and unlocks when call ends .. so $12 = 10 simultaneous calls ( if rate is 1.2 cents ) . On 02/02/07, Robert DeVries [EMAIL PROTECTED] wrote: I have found that if you don't have the minimum balance required for the voipjet premium server, you get the circuits busy message, you might want to check your balance. On 1/30/07, Alejandro Lengua [EMAIL PROTECTED] wrote: Hello, we have this problem with Trixbox 1.23 I have created an outgoing route where the 1st line has Voipjet and the 2nd an 3rd have voipcheap accounts. The problem is that at certain moments, when we call all the calls go through the voipcheap SIP accounts SIP, whose quality are not only not good enough but also consume a lot of bandwidth. The error message that returns Voipjet to Asterisk is that all circuits busy. What I asume from this? Thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] IAX Trunk timing
If the other server doesnt have any hardware device that can act as timer. then just compile zaptel and modprobe ztdummy .. This kernel module should act as timing source i think . ( it works with meetme ) . On 16/01/07, Andy Hester [EMAIL PROTECTED] wrote: I have read that an IAX trunk requires a timing device. What wasn't clear to me was whether it is like TDM ie 1 timing device for the trunk, or if each end requires a timing device. I have a zaptel card in one server; do I have to have one in the second server in order to do an IAX trunk? I set up a trunk and so far calls can be made one way, but not the other. It is probably just not configured correctly, but I just wanted to make sure as I can't seem to find any reason at the moment. Thanks, Andy ___ --Bandwidth 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] Stumped with Dial - $50 for answer
its notransfer=yes in iax.conf not transfer=no :) On 16/01/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Monday 15 January 2007 6:21 pm, Anselm Martin Hoffmeister wrote: could you verify or negate that adding the T option makes it work? That or transfer=no in iax.conf for hte user/peer entries involved. I never thought of IAX2 transfers here, for some reason I thought that Asterisk was terminating the call to TDM itself (one of the two ends). I wouldn't try transfer=mediaonly at this point; remove the transfer capability altogether. -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 ___ --Bandwidth 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 dynamic host question
Asterisk can manage dynamic hostnames itseld type dnsmgr refresh in asterisk cli . Also see /etc/asterisk/dnsmgr.conf On 10/01/07, Ale [EMAIL PROTECTED] wrote: Hi all, My asterisk box have some peers with as host the name of a dynamic dns resolver ex: foo.dyndns.org. All works fine, until the host foo.dyndns.org for some reason change his ip, asterisk didn't resolve again the new ip until a sip relolad Actually, i use a cron with a bash script to track the ip and eventually reload the sip.conf. Any tips for Asterisk ? Something like externrefresh for a peer? Thanks, Alessandro ___ --Bandwidth 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] How accurate is show translation?
I tried it on a intel 3 ghz p4 box and a athlon 3000 768 mb ram running vista and host for centos 4 ( vmware ) considering the load on athlon running asterisk ( that too under vista plus vmware ) while intel 3 ghz p4 1 GB ram box was sitting idle with centos , there was hardly a 1 ms difference in show translation on both machines . Besides i just compared my p4's results to ur D930 results and there is no difference ( infact my g729 results are better than ) .. But this doesnt mean both are same dual core cpu's will definitely give much higher number of channel transcoding then lower p4's . Put both the box under some cpu load by other programs and then use show translation recalc 30 and you will see performance difference between them ;) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - -- - - - - - - - gsm - - 22 2 2 1 5 82814 ulaw - 3 -1 2 2 1 5 82814 alaw - 3 1- 2 2 1 5 82814 g726 - 3 22 - 2 15 8 2814 adpcm - 3 22 2 - 1 5 8 2814 slin - 2 11 1 1 - 4 7 2713 lpc10 - 4 33 3 3 2 - 9 2915 g729 - 3 2 2 2 2 1 5 - 2814 speex - 4 3 3 3 3 2 6 9 -15 ilbc - 4 3 3 3 3 2 6 9 29 - On 23/12/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Eric ManxPower Wieling wrote: Leo Ann Boon wrote: Hi all, I'm using 'show translation' to help dimension my system, but I confused by the results I get. My 2 test systems (results below): an AthlonXP 2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar results (D930 is slightly faster). Googling shows that others have similar results running on other CPU speeds 2.0GHz. Does show translation recalc 30 show any different results? Eric, Before I posted, I ran tests with various recalc values between 10 and 200. The results are pretty much the same, give and take 1ms on either side. Forgot to add, I'm running 1.2.11 on both systems. Leo ___ --Bandwidth 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] Need quality toll free 800 number over IAX?
I have 3 toll free did's with nufone since 1 month .. Until now i dont have a problem with them .. their portal was good enough to do proper configuration and call quality wasnt bad ( even though i havent used them in really huge traffic yet ) . On 23/12/06, John Novack [EMAIL PROTECTED] wrote: Doug Crompton wrote: At the very least the BBB (bbb.org) should be notified. They have a web site SORRY, BUT IMHO the BBB is a joke. I wouldn't waste the time to type in one word and if it is really wire/internet fraud then the FBI (www.fbi.gov/majcases/fraud/internetschemes.htm) has a site you can register a complaint with. The FBI and the FTC, Federal Trade Comission are good choices. They don't require you to fly anywhere or do anything in court, and if enough people have simial experiences, they may very well do something The US government moves slow, but when it moves you REALLY don't want to be in the way of the army of bureaucrats. Don't talk yourself into inaction. If in fact you have any evedence at all, then register it. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] New astGUIclient VICIDIAL Release: 2.0.2
this really is a great program as far as i have heard even though i am not able to make it work for me _ On 23/12/06, Matt Florell [EMAIL PROTECTED] wrote: Hello, We've released another update to our astGUIclient suite: 2.0.2 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL call center suite. This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have made many changes on the server side, including changing logging from AGI to FastAGI which can cut system load in half on busy VICIDIAL systems. We have also tested the suite on Asterisk versions through 1.2.14(cannot use 1.2.11 or 1.2.12 because of Asterisk bugs) All client web-apps and administration pages are available in English, Spanish, Greek and German, with translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ --Bandwidth 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 crashed
Post this at bugs.digium.com along with some more info like if it crashes at
use of some specific application or randomly .
On 22/12/06, Edwin Lam [EMAIL PROTECTED] wrote:
our * crashed twice in a month with segmentation fault
a core dump. here's the stack trace:
#0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6
#1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6
#2 0xb7e17090 in strdup () from /lib/tls/libc.so.6
#3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879
#4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at
res_musiconhold.c:180
#5 0x080673ae in ast_deactivate_generator (chan=0x9455ca0) at channel.c
:1382
#6 0x08068d4e in generator_force (data=0x9455ca0) at channel.c:1405
#7 0x08061c50 in ast_read (chan=0x9455ca0) at channel.c:1857
#8 0x08069293 in ast_generic_bridge (c0=0xb659fcd0, c1=0x9455ca0,
config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c, bridge_end=
{tv_sec = 0, tv_usec = 0}) at channel.c:3260
#9 0x080655fd in ast_channel_bridge (c0=0xb659fcd0, c1=0x9455ca0,
config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c)
at channel.c:3524
#10 0xb78fad29 in ast_bridge_call (chan=0xb659fcd0, peer=0x9455ca0,
config=0xb6c4feb0) at res_features.c:1319
#11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0,
peerflags=0xb6c50568) at app_dial.c:1577
#12 0xb7097dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at
app_dial.c:1619
#13 0x0808e445 in pbx_extension_helper (c=0xb659fcd0, con=0xb7ed1900,
context=0xb659fe20 op05_x, exten=0xb659ff14 00116,
priority=1, label=0x0, callerid=0x0, action=0) at pbx.c:553
#14 0x0808efea in __ast_pbx_run (c=0xb659fcd0) at pbx.c:2227
#15 0x0808fcdf in pbx_thread (data=0xb7ed1900) at pbx.c:2514
#16 0xb7f7cb63 in start_thread () from /lib/tls/libpthread.so.0
#17 0xb7e7718a in clone () from /lib/tls/libc.so.6
another one:
#0 0xb6ff38e2 in decodeMP3 () from
/usr/lib/asterisk/modules/format_mp3.so
#1 0xb6ff4be6 in key () from /usr/lib/asterisk/modules/format_mp3.so
#2 0xb6ff4545 in key () from /usr/lib/asterisk/modules/format_mp3.so
#3 0x0806d3a1 in ast_readframe (s=0xb7eb490c) at file.c:570
#4 0xb7b0c134 in moh_files_generator (chan=0xb6b26dc0, data=0xb6b03328,
len=0, samples=160) at res_musiconhold.c:246
#5 0x08068cfe in generator_force (data=0xb6b26dc0) at channel.c:1401
#6 0x08061c50 in ast_read (chan=0xb6b26dc0) at channel.c:1857
#7 0x08069293 in ast_generic_bridge (c0=0xb6b26dc0, c1=0x8699fe8,
config=0xb6677eb0, fo=0xb6677988, rc=0xb667798c, bridge_end=
{tv_sec = 0, tv_usec = 0}) at channel.c:3260
#8 0x080655fd in ast_channel_bridge (c0=0xb6b26dc0, c1=0x8699fe8,
config=0xb6677eb0, fo=0xb6677988, rc=0xb667798c)
at channel.c:3524
#9 0xb78ddd29 in ast_bridge_call (chan=0xb6b26dc0, peer=0x8699fe8,
config=0xb6677eb0) at res_features.c:1319
#10 0xb7033301 in dial_exec_full (chan=0xb6b26dc0, data=0xb6677eb0,
peerflags=0xb6678568) at app_dial.c:1577
#11 0xb7031dc5 in dial_exec (chan=0x48, data=0x48) at app_dial.c:1619
#12 0x0808e445 in pbx_extension_helper (c=0xb6b26dc0, con=0x48,
context=0xb6b26f10 op05_x, exten=0xb6b27004 00116, priority=1,
label=0x0, callerid=0x0, action=0) at pbx.c:553
#13 0x0808efea in __ast_pbx_run (c=0xb6b26dc0) at pbx.c:2227
#14 0x0808fcdf in pbx_thread (data=0x48) at pbx.c:2514
#15 0xb7f5fb63 in start_thread () from /lib/tls/libpthread.so.0
#16 0xb7e5a18a in clone () from /lib/tls/libc.so.6
here's the versions of various components:
asterisk: 1.2.7.1, zaptel: 1.2.5, libpri: 1.2.2
any clues would be appreciated?
--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] Billing solution
I am looking for exactly same kind of billing stuff but i dont think you will get it without letting ur billing program make some changes in asterisk . On 20/12/06, Carlos Rojas [EMAIL PROTECTED] wrote: a2billing Is very good On 12/19/06, Giedrius Augys [EMAIL PROTECTED] wrote: 2006/12/19, C F [EMAIL PROTECTED]: Can anyone recommend a call accounting solution with rating for post paid billing that works well with asterisk using the account code or any other info from the CDR? I don't want the billing software to any phone calls for me, therefore any solution that modifies my extensions.conf is out, nor does it have to allow for customers the ability to log in to check their usage/balances. I have looked at astbill but it looks to be way overcomplicated for what I want, as well as it requires realtime. 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 Mor and Mcc ___ --Bandwidth 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] Linux distro + Asterisk or Trixbox?
Besides that you can use centos-plus repository which has lot of updated stuff not available in RHEL4 like php5 , mysql5 and all . On 18/12/06, Carla Schroder [EMAIL PROTECTED] wrote: On Sunday 17 December 2006 10:47 pm, Andrew Joakimsen wrote: I've used Asterisk on a bunch of RH 7.3 machines which were then replaced by RHEL 4. It is very stable, my biggest compliant is that RHEL(or CentOS, which is a direct rip-off) uses outdated packages (Linux 2.4.x, Apache 1, Mysql 4, php 4, etc) and Linux 2.4.x requires certain USB hardware to use zaptel timing without a hardware card, so we have a bunch of these dual xeon machines with the wrong USB hardware and can only run MeetMe on the one with the t1 cards. CentOS 4 was released May 2005 with a 2.6 kernel, Apache 2, and all other similarly current packages. The current kernel is 2.6.9-something. CentOS is a legal re-distribution of RHEL 4 rebuilt from source RPMs. Just like Pie Box, White Box, Tao, Lineox, and all the other Red Hat clones. -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power 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] iax2 softphone attended transfers
I have configure it by using the *2 atxfer feature of asterisk but its not as good as other attended transfer which sipphones give ( like sjphone where you can switch between two anytime ) . Also tried zoiper but it do not have even blind transfer yet . Any idea when idefisk 2.0 is going to be released :( or any other iax phone . On 16/12/06, Zoa [EMAIL PROTECTED] wrote: Idefisk 2.0 will have it. Zoa Mail list wrote: Is there any good iax2 softphone capable of attended transfer ( like sjphone for sip ) . ? I tried iaxcomm and idefisk both seems unable to handle attended transfers. ___ --Bandwidth 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] Linux distro + Asterisk or Trixbox?
If you are really new to linux then go for trixbox . I started with trixbox and eventually went away from it by removing extra stuff and putting custom compiled asterisk's and removing their rpm's . If you are good at linux then definitely go for debian + asterisk or centos+asterisk and put freepbx on it for some ease of use . For production server fop server and all are not at all required .. Trixbox is good for newbies but it saves lot of time :) . On 17/12/06, Carla Schroder [EMAIL PROTECTED] wrote: On Saturday 16 December 2006 5:14 am, Phil Finkler wrote: Hey all, I've been doing a lot of playing, and a lot of reading, and it seems people are split as to whereas if they're running their favorite Linux distro and asterisk or Trixbox. I'm getting closer to really looking at a production environment and I'm just looking for any opinions. I'm really enjoying learning linux and asterisk, so initial ease of use isn't really a huge benefit to me. In the end stability and upgradeability will be my main concerns. Trixbox is HUGE. If you need all the bells and whistles- a MySQL backend, the CentOS operating system, AMP, SugarCRM, Festival, monitoring consoles, everything pointy-clicky, and on and on and on, then Trixbox is for you. It has some disadvantages. There is not a clear correlation between the graphical admin tools and the underlying text configuration files, so debugging problems is harder, and you have to know two ways of doing things. The documentation sucks rocks- there isn't any to speak of. When [EMAIL PROTECTED] changed the name to Trixbox, they moved to a new web site and didn't bring any of the help docs or forums with them. There is a book you can buy, 'Trixbox Made Easy'. I think it's better to learn plain-vanilla Asterisk first. Then if you move on to some other implementation you'll be better prepared to understand what it's doing. You might give AstLinux a try. It's a complete Linux distribution + Asterisk 1.2.-something, but tiny, about 40 megabytes. No wasted bits. It has a nicely-organized Web GUI for those who like such. You can switch between the Web interface and editing the config files directly without getting in trouble. It runs on single-board computers and ordinary old PCs. It's my current fave, though I'm also running Asterisk 1.4/CentOS on a test box. -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power 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] Iptables rule help
I am sure rtp ports arent blocked .. On 16/12/06, Derek Whitten [EMAIL PROTECTED] wrote: Mail list wrote: Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under extensions is not working .. asterisk doesnt listen to port 5091 .. but if i put in general section of sip.confthen it works but then asterisk wont listen on 5060 . How can i use iptables in this situation ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users RTP ports blocked? (rtp.conf) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Iptables rule help
I have shifted asterisk port to 5091 . Now i am able to register properly using sjphone but still when dialing number it keep on showing calling .. and do not go ahead . I change asterisk's rtp ports too but still i am unable to make call . My other softphone on different internet isp is working properly . :( On 16/12/06, Tim C. Lewis [EMAIL PROTECTED] wrote: well that should map incoming packets to 5091 to 5060, but may not rewrite [new] outbound packets from 5060 to 5091, which your isp may be blocking. an iptables SNAT or MASQUERADE might help you there. i'm not positive on if this would be needed or not. more importantly, however, if your isp is blocking all outgoing traffic to 5060, it won't get to your softphone anyway, unless you also configure that end to also not use 5060. and if you're reconfiguring ports on the softphone end anyway, why not just put 5091 in there, 5091 in sip.conf's bindport, and not mess with iptables at all? another option might be that your isp is blocking rtp as well. can you see what the asterisk console is doing when you attempt such calls? and/or tcpdump? -tcl. On Sat, 16 Dec 2006, Mail list wrote: Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under extensions is not working .. asterisk doesnt listen to port 5091 .. but if i put in general section of sip.confthen it works but then asterisk wont listen on 5060 . How can i use iptables in this situation ? ___ --Bandwidth 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] Iptables rule help
Actually port block is on softphone side and not on asterisk server's internet connection .I put this in iptables of asterisk server iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j REDIRECT --to-port 127.0.0.1:5060 server is listening on port 5060 Now strange part is everything s working .. but asterisk is not detecting hangup . I make call on softphone .. call goes everything works fine but when i hang softphone .. i can see on asterisk that call is still going on .,... and this is not a problem of softphone i am sure of that :( On 16/12/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Samstag, den 16.12.2006, 02:32 +0530 schrieb Vicky: I have shifted asterisk port to 5091 . Now i am able to register properly using sjphone but still when dialing number it keep on showing calling .. and do not go ahead . I change asterisk's rtp ports too but still i am unable to make call . My other softphone on different internet isp is working properly . :( Can your softphone reach a voicemail() extension, or echotest? Will that work with audio in both directions? And then, I am not sure wether I understand your setup correctly. Are you trying Asterisk [portblocked line] ISP. Internet. Softphone ? In that case, local tests like those mentioned above will help to rule out wether problems are on the internet part or possibly on a PRI or whatever connected to your asterisk, over which you would like to dial out. BR Anselm ___ --Bandwidth 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] IAX2 to SIP protocol translation overhead?
One main disadvantage would be the media stream will pass through asterisk ( no reinvites like sip-sip ) but its not a problem if client pc'a and your asterisk server are on same network .Sip-iax conversion takes less cpu but it will be more if codec transcoding is involved . On 12/12/06, David Thomas [EMAIL PROTECTED] wrote: Just wondering if there is much CPU overhead in the translation from IAX2 to SIP, and how taxing this function is as compared to transcoding. We're trying to build an efficient system and would like to avoid taxing the CPU as much as possible. Our upstream service provider is 100% SIP, however we'd like to use IAX2 in our network as well, if it does not cause too much overhead. Not sure if it matters, but we will be running aprox 100 simultaneous calls. Thanks, David ___ --Bandwidth 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] illegal VoIP in India
Yeh problem is they are directly buying from providers in US/UK without paying 12 % tax on voip .. i guess people who buy itsp license can resell this minutes by paying tax to government in between . On 08/12/06, ram [EMAIL PROTECTED] wrote: I'm not sure, but does this only apply to VoIP service providers? What about self run asterisk servers? Tom Hi if the self running Asterisks people connected to Indian ISP not a problem i belive. if they are directly connecting to USA provider, Avoiding India ITSP that could be a problem i think. ram ___ --Bandwidth 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] Basic question regarding re-INVITE
canreinvite = yes in sip,conf ( trunk section ) ?? No t,t in dial command . No call recording in between , same codec should be supported by both trunk as well as extension . If trunk is iax2 and extension is sip then also asterisk will sit in media path . On 08/12/06, Alex Guan [EMAIL PROTECTED] wrote: All, This basic question might have been asked thousands of timesbut anyways: when can Asterisk send out an re-INVITE to the line/trunk side? It seems that the canreinvite does NOT matter for calls toward the trunk. E.g. When I put a phone on hold, the re-INVITE is sent from phone to the Asterisk, but then that's it. The Asterisk never sends it out. It seems to work for extension to extention, but not extension to line. What am I missing? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wierd callerid problem
Yeh asterisk seems to use extension number for calls between extensions on same server and sends callerid only for outside numbers ( via sip trunks ) . On 08/12/06, Greg Kennedy [EMAIL PROTECTED] wrote: I have a site running asterisk 1.2.8 with a hand full of polycoms and grandstream 2Kxp's. When a call is missed and you look at the missed call logs on either, its has the persons exten, not the incoming caller id. Any ideas? \\\|/// \\ ~ ~ // ( @ @ ) --oOOo-(_)-oOOo— ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [asterisk-biz] Server for 100 concurrent calls
that site also has g729 codecs for asterisk but is it legal to use them ?? ( digium charges $10 each g729 channel ) On 08/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: g723 codec isn't problem, you can obtain for all asterisk versions from: http://kvin.lv/pub/Linux/Asterisk/ PJ Jean-Michel Hiver wrote: [EMAIL PROTECTED] a écrit : Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Since Digium doesn't provide g723 codecs (as far as I'm aware), and there's yet no transcoding card for Asterisk (one is supposed to be out at some point, but when... god knows), for the moment you should look into something else than Asterisk. Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] No ID from the calling party in SIP Header
callerid=John Doe 1234 On 05/12/06, Sven Beisiegel [EMAIL PROTECTED] wrote: Hi... I just started working with Asterisk and found something that looks like an error, but i want to be sure, so that's why I'm asking you. When i make a call from A to B (both SIP clients), I don't see the name of the called party in the phone that initiated the call, just the dialed number. I made an ethereal trace and found out, that there is no name during the initiation in the SIP Header? But there is a Remote-Party-ID in the SIP Packet that goes from the Server to the called party...There is nothing like P-Asserted-Id in the SIP Packet that goes to the calling party. My question... Is this an error or did i forget to activate something? The configuration of the sip.conf is: [general] language=de port=5060 disallow=all allow=alaw allow=ulaw allow=GSM nat=no canreinvite=no tos=lowdelay context=default [9001] type=friend username=9001 secret=password host=dynamic callerid=Beckenbauer, Franz 9001 context=default mailbox=9001 callgroup=1 pickupgroup=1 sendrpid=yes [9002] type=friend username=9002 secret=password host=dynamic callerid=Walter, Fritz 9002 context=default mailbox=9002 callgroup=1 pickupgroup=1 sendrpid=yes cheers, Sven ___ --Bandwidth 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 voice recording through TE110p
Asterisk can record all outgoing calls ( see voip-info.org for asterisk cmd monitor and mixmonitor ) hardware requirements depends on volume of calls to be recorded . Faster sata raid or scsi drives recommended for high number of alternate calls . On 09/12/06, Raja Chidambaram [EMAIL PROTECTED] wrote: Hi all, We are in the process of setting up a E1 (TE110p)connection based asterisk server in which we want to record all the voice conversations.Isthis facility supported on asterisk if so how to configure.What are hardware dependencies invloued in setting up this facility. Thanks in advance. with regards raja -- Everyone is raving about the all-new Yahoo! Mail beta.http://us.rd.yahoo.com/evt=45083/*http://advision.webevents.yahoo.com/mailbeta ___ --Bandwidth 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] Codec Selection in asterisk
I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as ilbc . Now when i put allow=gsmilbc in my voip carrier's extension then it uses gsm ( first preference ) to send calls but half of my softphones use ilbc so asterisk does codec transcoding in between using lot of cpu .. how ever my carrier does support ilbc tooo but when i put allow=ilbcgsm then it uses ilbc again and does codec transcoding from gsm to ilbc for rest of softphones . How can i make asterisk to be smart in choosing codec .. and use ilbc to voip carrier if softphone is using ilbc or use gsm when softphone is using gsm ( but still should do call recording in between ) .. I am using freepbx for most of configuration btw... Any suggestions ? ___ --Bandwidth 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] Codec Selection in asterisk
I am still on asterisk 1.2 branch svn ( afraid of word beta on server :( ) . I will try out that patch. On 08/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: you can try this patch, 0004825: [patch][post 1.4] New codec negotiation algorithm http://bugs.digium.com/view.php?id=4825 I'm think, this is one of the most wanted feature, but unfortunately will not be in asterisk 1.4 and we must wait for 1.6 to be officially supported feature :'( PJ Vicky wrote: I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as ilbc . Now when i put allow=gsmilbc in my voip carrier's extension then it uses gsm ( first preference ) to send calls but half of my softphones use ilbc so asterisk does codec transcoding in between using lot of cpu .. how ever my carrier does support ilbc tooo but when i put allow=ilbcgsm then it uses ilbc again and does codec transcoding from gsm to ilbc for rest of softphones . How can i make asterisk to be smart in choosing codec .. and use ilbc to voip carrier if softphone is using ilbc or use gsm when softphone is using gsm ( but still should do call recording in between ) .. I am using freepbx for most of configuration btw... Any suggestions ? ___ --Bandwidth 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] centos 4.4 + asterisk
I am not sure but i think that fix is for compiling zaptel not asterisk . Asterisk runs on centos with 0 problems :) On 05/12/06, varun [EMAIL PROTECTED] wrote: Thanks Karl. On Tue, 2006-12-05 at 08:20 -0500, [EMAIL PROTECTED] wrote: I have CentOS 4.4 on several boxes with Asterisk 1.2 and they run great. Have not tested conferencing yet though. Karl Hello, Are there any issues with Centos 4.4 and asterisk. Thanks in advance Varun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec transcoding and call recording
heres my scenariosoftphone-Asterisk( outgoing call recording )Call Provider I am recording all outgoing calls on asterisk so its obvious that there is no native bridging . Suppose if i am using gsm from softphone--asterisk and then what codec should i prefer for asterisk- provider ?? Bandwith is not at all a problem between asterisk-provider connection . So if i am using gsm from softphone to asterisk ... then what would be better choice ? sending gsm to provider or ulaw/alaw ? I suppose asterisk already touches audio stream since it is recording to wav file but what i want to know is do asterisk sends same gsm stream ahead to provider ( and reduces cpu usage ) or it still have to transcode for recording ? because if its transcoding anyway then sending ulaw/alaw stream ahead would be better right ? or am i missing something :-/ Also what would be better if i use g729 from softphone--asterisk( capable of g729 transcoding ) should i send g729 ahead or ulaw/alaw? ___ --Bandwidth 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] Moderate setup
I am planning to put up a asterisk server with around 50-60 phones over a lan . I am planning on keeping a decent server ( for outbound pstn ) and all phones connected via linksys pap2 ( all 60 phones as pap2 registering to asterisk) . Does this kind of setup give problem ? ___ --Bandwidth 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] RTP Media Path
Asterisk wont sit in media path if both callee and caller agrees on common codec, both have canreinvite=yes in sip.conf, no t,T are used in dialplan ( please correct me if i am wrong ) , no call recording is enabled . I think asterisk does native bridging even if one is behind nat ( i tested with atleast one party behind nat not sure if it works when both are behind nat ) and devices should support reinvites .. On 03/12/06, Dovid B [EMAIL PROTECTED] wrote: I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider Asterisk - ATA and vice versa (ATA - Asterisk SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP provider and vice versa. (This is of course if there is no NAT involved). Now say I had such a set up will the server be able to handle more calls than average if the only responsibility if the server is to authenticated and pass along the calls ? (There will be an AGI running in the begining to determine what route to used based on how many minutes each route has used). Now if the ATA's are behind VOIP and asterisk is on a public IP then does asterisk have to sit in the media path ? Also can some one explain exaclty when the RTP session is started and stopped. Also another set up we are woroking on is SIP Provider (Incoming DID) Asterisk (for authentication based on PIN) - Back to SIP Provider. The asterisk server will be on a public IP. Can I have asterisk stay out fo the media path (here I asume yes. Just wana be 100% sure). Thanks a lot. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call recording filename
I am using asterisk along with freepbx . When recording is enabled for a extension the call record file made in /var/spool/asterisk/monitor contains information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can be a big mess if there are more than 1000-2000 files in that folder and very hard to locate a call recording based on call time and extension number who dialled. I need to put something like outgoing number dialled within call file name instead of uniqueid .. After watching in console i opened up /var/lib/asterisk/agi-bin/recordingcheck and saw that it is setting callfilename variable with extension number,time,unique id , etc. so i edited and instead of $uniqueid i put $DIALEDPEERNUMBER ( saw in http://www.voip-info.org/wiki/index.php?page=Asterisk+variables ) but its just not giving dialed number and hence callfilename doesnt contain outgoing number . Any suggestions how can i get outgoing call number in recording file ? ___ --Bandwidth 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] Looking for toll-free US did
I am looking for a toll-free US 1800 DID which can be setup quickly . I have seen nufone there quality is very good but they charge for 30 seconds minimum ( others do 6/6 i guess ) . east coast gateway server preffered . . Plz lemme know if you have some suggestions i want it to be setup very quickly :) . Thx . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip reinvite
If canreinvite=yes is specified in sip.conf for 2 sip extensions and call recording is disabled in asterisk, both legs have same codec . Doesit always does native bridging . I am using freepbx . How can i know if a call is going through asterisk or they are bridged directly to each other ? Does sip reinvite gives problems in billing ? Is there any cli command to know that ? ___ --Bandwidth 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 incoming call behaviour
I am using asterisk to receive call from a DID provider . In configured everything in freepbx properly and its working . I forwarded incoming calls from did to a certain extension . Now i tried calling from another sip provider to this box , when i call from other provider to my DID number then call reaches asterisk and is sent to configured extension .. however if the extension hangs up without picking then also i am being billed at sip provider ( outgoing one ) . In simple words when people call me then they ( other people ) are billed even if configured extension isnt picked up and hangs the phone. Normally when you call a person and they hang up then you arent charged . Is this asterisk behaviour or is it freepbx dialplan the culprit here ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] qualify=yes
I doubt that . I think qualify=500 means asterisk checks every 500 ms if the other extension is available or not . Because when qualify=( value in ms ) is set and you do a sip show peers in console asterisk whos how much latency is there between extension and asterisk . If i set qualify = no then it shows UNKNOWN . If i set qualify=10 then it doesnt mean asterisk shows extension lagged if latency is less than 10 ms ... It just checks every 10 ms for extension . I am not very sure though :) On 22/11/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Enrico Pasqualotto wrote: Enrico Pasqualotto wrote: hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? You have to set qualify=second instead of qualify=yes|no. This is WRONG. qualify=500 means consider this device lagged if responses take longer than 500ms I don't know if you can set the frequency of qualify packets. If you can, I assume the option would be listed in sip.conf.sample. ___ --Bandwidth 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] Recordings.
Yeh even a simple UDMA 5 enabled hard drive can handle 30 calls recording easily . Sata hard drives are even better . On 22/11/06, Marcus Franke [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 [EMAIL PROTECTED] wrote: Does anyone have experience with recording multiple calls simultaneously on a single system with or without performance trouble? What kind of system do I need? Well, isnt this just a simple calculation? Do a record of one of your lines for about a minute. Look at the size of the created file and divide the kb by 60 and multiply by 20 and you have an first overview about how much data will get written down to harddisk per second. But I think you should be fairly well if you use state-of-the-art server disks. They should be fast enough for this. Marcus -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZFjMqwWWw48OFWoRAvhtAKC58l2WXpK/RmzWB2FtRDbHFxsJWQCgp3OI o9zojHnurfaMtAOjLytHFUs= =upqP -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 ___ --Bandwidth 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] Recordings.
Hey i said that as per his requirement as an example :) . His requirement is just around 20 calls . For a moderate server i think sata raid should be fine ..Heres some result posted by someone for recording calls on ram disk . http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/118497 On 22/11/06, Marcus Franke [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Vicky wrote: Yeh even a simple UDMA 5 enabled hard drive can handle 30 calls recording easily . Sata hard drives are even better . Hehe, UDMA sounds like EIDE drives.. nice to see they are fast enough, but I do not recommend those as server hardware. ;-) But, if John is going to buy a extra new server, he could use two drives in a mirror setup extra for recordings of these files. As it is not only the frequency of reading/writing these files but other accesses of the media like starting programs or reading/writing of logfiles that slowes down the access to the recorded audio files. Marcus -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZGUUqwWWw48OFWoRAvidAJwPSpTSuY6nwxKTDKI8fZDmshmbUgCgtWAp 27akzsEDv03q5CmlGMObo50= =2jAI -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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on CDR Database
I am not sure if i understood what you mean but yes asterisk cdr's can be used for billing with some modifications of your own. Asterisk can make cdr in csv,mysql,postgresql with complete call info which can be used for billing system's . On 19/11/06, Noc Phibee [EMAIL PROTECTED] wrote: Hi I have a small question on CDR Database: It's used by billing software no ? he have only one structure of data or they have multi structure with more information logged ? sample: cdr simple and cdr_extended thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] AdvancedVoIP Billing ?
I am also searching one for post-paid billing .. but most like astpp wants to eat whole system themselves managing extensions and all . I need a type of solution that can just bill people based on mysql cdr using accountcode and amagflags .. I am thinking to make some myself now but it will take me time to learn php so i am still searching :( On 18/11/06, Noc Phibee [EMAIL PROTECTED] wrote: Hi thanks for your answer, no i don't have see this software because i don't see screenshot or demo ;) Hermann Wecke a écrit : Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html ___ --Bandwidth 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] AdvancedVoIP Billing ?
I will definitely give it a try again to astpp then . I actually saw its online demo and was bit confused . I thought its managing extensions and all and i will have to start from scratch so i didnt gave it a try .It is a great software but only thing holding me back was thought that i will have to start from scratch :P . On 18/11/06, Darren Wiebe [EMAIL PROTECTED] wrote: I'll jump in here. As the author of ASTPP, I have gone to considerable effort to make it so that ASTPP does NOT need to eat a whole system. All that you really need on the asterisk box to get ASTPP working in terms of asterisk requirements, is to make sure that the cdrs in the database have an accountcode set. You do not need to use it to manage your dids and extensions, etc. Darren Wiebe [EMAIL PROTECTED] Vicky wrote: I am also searching one for post-paid billing .. but most like astpp wants to eat whole system themselves managing extensions and all . I need a type of solution that can just bill people based on mysql cdr using accountcode and amagflags .. I am thinking to make some myself now but it will take me time to learn php so i am still searching :( On 18/11/06, *Noc Phibee* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi thanks for your answer, no i don't have see this software because i don't see screenshot or demo ;) Hermann Wecke a écrit : Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html http://www.aradial.com/voip-billing-radius.html ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] chanspy crash the asterisk 1.4
Please go to bugs.digium.com and file this bug they will difinitely get it working . On 16/11/06, Thirumal Saminathan [EMAIL PROTECTED] wrote: hi, exten =6000,1,dial(SIP/6000,15,tr) exten =6002,1,dial(SIP/6002,15,tr) exten =6004,1,dial(SIP/6004,15,tr) exten =6006,1,dial(SIP/6006,15,tr) exten =6008,1,chanspy(SIP/6006 | wbq) when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one. when coversation between the 6002 to 6006. in my Console mode i got the following comment *CLI -- Executing [EMAIL PROTECTED]:1] Dial(SIP/6002-08590248, SIP/6006|15|tr) in new stack -- Called 6006 -- SIP/6006-08594188 is ringing -- SIP/6006-08594188 answered SIP/6002-08590248 -- Executing [EMAIL PROTECTED]:1] *ChanSpy*(SIP/6004-08589040, SIP/6006|wq) in new stack == Spying on channel SIP/6006-08594188 [Nov 11 15:19:37] NOTICE[8974]: app_chanspy.c:202 start_spying: Attaching SIP/6004-08589040 to SIP/6006-08594188 Segmentation fault (core dumped) linux:~ # please reply reg . this issue. Regards, Thiru ___ --Bandwidth 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] Attempting native bridge of
g729 is not a free codec . YOu have to buy it from digium at rateof $10 per channel license . If you are just using asterisk and havent bought g729 license then asterisk will just do bridging of g729 and wont edit/transcode stream . On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote: I have the following scenario: g729gsm UAS --- * --- UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ¿Am I wrong? The UAC and UAS are registering with * properly: --- sip.conf [testgsm] type=friend host=dynamic username=testgsm context=astertest canreinvite=no disallow=all allow=gsm [testg729] type=friend host=dynamic username=testg729 context=astertest canreinvite=no disallow=all allow=g729 - - dspam*CLI sip show peers Name/username HostDyn Nat ACL Port Status testgsm/testgsm172.16.51.244D 1 Unmonitored testg729/testg729 172.16.51.244D 2 Unmonitored -- Executing Answer(SIP/testgsm-081784b0, ) in new stack -- Executing Wait(SIP/testgsm-081784b0, 1) in new stack -- Executing Dial(SIP/testgsm-081784b0, SIP/testg729) in new stack -- Called testg729 -- SIP/testg729-0817dd90 is ringing -- SIP/testg729-0817dd90 answered SIP/testgsm-081784b0 -- Attempting native bridge of SIP/testgsm-081784b0 and SIP/testg729-0817dd90 - After the call is established the UAC is sending some RTP captured in a pcap file in gsm: -- tcpdump -T rtp udp --- 15:58:31.868404 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.868676 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.895551 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.895775 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.936468 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.936477 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.936711 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.936908 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 - Is there something wrong within the SDP? or Am I doing something wrong? Any comments would be appreciated.. thanks!! P.S. I am using Asterisk 1.2.12.1 if that matters. -- Greetings... Víctor Toofic --- 2006-11-15 16:15:12 UDP message sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:34836 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 138 v=0 o=user1 53655765 2353687637 IN IP4 172.16.51.244 s=- c=IN IP4 172.16.51.244 t=0 0 m=audio 10001 RTP/AVP 0 a=rtpmap:18 GSM/8000 --- 2006-11-15 16:15:12 UDP message received [404] bytes : SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0;received= 172.16.51.244 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- 2006-11-15 16:15:12 UDP message received [609] bytes : SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0;received= 172.16.51.244 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060;tag=as14685910 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 161 v=0 o=root 3567 3567 IN IP4 172.16.51.215 s=session c=IN IP4 172.16.51.215 t=0 0 m=audio 17050 RTP/AVP 18 a=rtpmap:18 GSM/8000 a=silenceSupp:off - - - - --- 2006-11-15 16:15:12 UDP message sent: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-5 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060;tag=as14685910 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Contact: sip:[EMAIL PROTECTED]:34836 Max-Forwards: 70 Subject: Performance Test
[asterisk-users] Asterisk call recording
I have call recording enabled for some extensions and they make lot of calls . I see there are many files in /var/spool/asterisk/monitor but i need to know which file belongs to which call .. In old version of asterisk this was available in lastapp field of mysql table . Now lastapp shows resetcdr .. ARI ( asterisk recording interface ) that comes with freepbx has this ability ( it shows call number/time/recorded file ) but it stops working if there are more than 2000-3000 files in monitor folder . How can i make asterisk makes index of call recording files ? If its making then how can i retrieve it ? Any ideas :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempting native bridge of
Thats really strange .. if you have made canreinvite=no then it should not even attampt native bridging and should transcode codecs ..something's fishy here .. Also try to put canreinvite=no in testulaw exntension too . On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote: El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba: g729 is not a free codec . YOu have to buy it from digium at rateof $10 per channel license . If you are just using asterisk and havent bought g729 license then asterisk will just do bridging of g729 and wont edit/transcode stream . On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote: I have the following scenario: g729gsm UAS --- * --- UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ¿Am I wrong? I get the same message even if I'm not using g729: --Attempting native bridge of SIP/testgsm-081784b0 and SIP/testulaw-0817da80 ulawgsm UAS --- * --- UAC ___ --Bandwidth 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] safe_asterisks pawning multiple asterisk process???
its normal .if there are many calls going . You should worry if your load or memory usage is very high .On 16/11/06, Andre Courchesne - Consultant
[EMAIL PROTECTED] wrote:
We have 1 server that after a few hours operating has multiple processof asterisk running. Here is the pstree output:# pstreeinit-+-atftpd |-auditd---{auditd} |-bash---safe_opserver---op_server.pl
|-crond |-cwASTcall.pl |-dbus-daemon |-events/0 |-hald-+-hald-addon-acpi |`-2*[hald-addon-stor] |-httpd---3*[httpd] |-khelper |-klogd |-ksoftirqd/0
|-kswapd0 |-kthread-+-aio/0 | |-ata/0 | |-hda_codec/0 | |-kacpid | |-kauditd | |-kblockd/0 | |-khubd | |-kseriod
| |-2*[pdflush] | |-reiserfs/0 | |-rpciod/0 | |-scsi_eh_0 | |-scsi_eh_1 | `-scsi_eh_2 |-2*[mingetty] |-mysqld_safe---mysqld---16*[{mysqld}]
|-ntpd |-safe_asterisk---asterisk-+-45*[asterisk] |`-22*[{asterisk}] |-sshd---sshd---bash---pstree |-syslogd |-udevd |-usb-storage `-wan_ecd---wan_ecd
And ps aux | grep asterisk:# ps aux | grep asteriskasterisk20470.00.1 92001516 ?SNov14 0:00/usr/sbin/httpdasterisk20840.00.2115442388 ?SNov14 0:00
/usr/sbin/httpdasterisk20850.00.2115442384 ?SNov14 0:00/usr/sbin/httpdroot21960.00.0 2172 456 ?SNov14 0:00 /bin/sh/usr/sbin/safe_asterisk -U asterisk -G asterisk
asterisk22151.6 10.0 122496 90984 ?Sl Nov1438:07/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk22500.00.0 2176 376 ?SNov14 0:00 -bash-c cd /var/www/AMP/panel /var/www/AMP/panel/safe_opserver
asterisk22510.00.0 2128 868 ?SNov14 0:00/bin/bash /var/www/AMP/panel/safe_opserverasterisk22533.20.8 89887336 ?RNov1473:54/usr/bin/perl -w ./op_server.pl
asterisk 121050.00.8314407804 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 130800.00.8320967616 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 183520.00.9360808684 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 193520.00.9365288764 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 254020.00.9391968972 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 264970.01.0404489372 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 309010.01.0420649308 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk31600.00.6439685624 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 124440.00.5496365148 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 225380.00.5545325148 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 260320.00.5569485148 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 275170.00.5570565148 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 318060.01.0589569800 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk36550.01.0600889932 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk46360.01.160956 10316 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk74940.01.262200 10952 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk92760.01.364856 12040 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 115920.01.465404 12720 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 127570.01.466808 13504 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 131670.01.466576 13296 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 133630.01.465936 13156 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 162510.01.668812 14664 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 169420.01.668600 14676 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 248180.01.672740 15308 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 297140.01.775332 15824 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk51140.01.678932 15144 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk57160.01.778560 15408 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk96050.01.781680 16228 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 102350.01.881020 16864 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 174290.01.984996 17896 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 180900.02.085480 18176 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 205420.02.086980 18732 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 243700.04.088652 36340 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 302470.06.092268 54432 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 309040.06.192492 55920 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 313630.06.292500 56396 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk
Re: [asterisk-users] PortSip and Astericks new install
try : [John] type=friend secret=test host=dynamic disallow=all allow =gsmilbculawalaw Also try other sip phone slike sjphone just to make sure there is no prob . On 16/11/06, Charlie Grosvenor [EMAIL PROTECTED] wrote: I have just installed Asterisk and installed the sample configuration files. Asterisks appears to be working and I have added a SIP client: [John] type=friend secret=test host=dynamic allow=all I have been trying to dial the demo number 500 when using PortSip, Asterisks answers the phone but PortSip gives me the error: Call failed: codec not accepted 488. I have tried changing the enabled codecs in PortSip but this makes no difference. I have also tried various other SoftPhone but none of them seem to work. Anybody know what I have missed / doing wrong? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is asterisk able to integrate with MS SQL
Yes asterisk can do that . If you mena for call records then see http://www.voip-info.org/wiki-Asterisk+cdr+mysqlAlso see http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQLOn 14/11/06, Sharon Lim [EMAIL PROTECTED] wrote: Hi there, I am looking around, is there anyone did any integration asterisk talk to / connect to MS SQL? Thanks-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___--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] Re: Is asterisk able to integrate with MS SQL
oops sorry i thought its my sql didnt notice it's MS SQL :D On 14/11/06, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED],Sharon Lim [EMAIL PROTECTED] wrote: Hi there, I am looking around, is there anyone did any integration asterisk talk to / connect to MS SQL?Look for the package FreeTDS and install it. Then build Asterisk and it will include the TDS driver that can log CDRs to MS SQL.Alternatively, install ODBC drivers for MS SQL and then use Asterisk's ODBCfunctions.CheersTony--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED] - http://tony.mountifield.org ___--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] Can I disable send e-mail feature in the voicemail application?
just dont enter any email address while creating extension / mailbox ;)On 14/11/06, Ma Zhiyong [EMAIL PROTECTED] wrote: HI, allCan I disable send e-mail feature in the voicemail application?___--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 forward behind a nat
One more thing i would like to point out is that softphones like sjphone use some freeware stun server to detect nat on network (as a client ) . Asterisk(asclient) cannot use external stun server to detect nat type automatically so i think thats why it isnt able to make calls while softphone works . Port forwarding should help here .Also edit sip_nat.conf after port forwarding but it will be a burden to setup if asteriskisondynamicip. On 14/11/06, nik600 [EMAIL PROTECTED] wrote: On 11/13/06, Vicky [EMAIL PROTECTED] wrote: IF your asterisk server is behind NAT and no port forwarding is done then how can that static ip user/device reach it . You will have to keep asterisk server in static ip or do port forwarding to accept connections from outside .i've understand but my Asterisk server doesn't have to accept calls,it only have to make a call OR maybe i didnt understand senario properly here . Is it like your Server with SIP application (public_address) responds to sip calls made by any program ( like sjphone pc-pc sip ) . exaclty, infact if i use sjphone from an adsl connection (using amodem) with sjphone i can make a call to the server with SIPapplication thats case then asterisk should be able to call it like any other program or maybe theres nat scenario playing bad here :-/ . Can you port forward from firewall to asterisk server??I can port forward from firewall to server, but i'd like to maintainthe server not accessible from internet. I'll check some NAT configuration, many thanks to your reply.___--Bandwidth 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] operator console
Could this be considered spam ? I believe this is second threas realted to that pbx .On 13/11/06, Jordi Nelissen [EMAIL PROTECTED] wrote: Check out the ESCAUX net.PBX operator console. In use in variouscompanies with 200+ extensions. Powerfull and convenient. http://www.escaux.com//index.php?option=com_contenttask=viewid=61Itemid=350Best Regards,Jordi--www.escaux.comBusiness IP Telephony Forrest Beck wrote: Talk to the folks at Asteria.The have a product called Reign.It looks just like your old interface, runs off .NET as a client on the machine. http://www.asteriasgi.com/pbx/reign On 11/7/06, Stephen Wingfield [EMAIL PROTECTED] wrote: Andres, The Bicom Systems Operator Panel is probably what you are looking for. OPCOM http://www.bicomsystems.com/docs/opcom/1.0/html/ This is included with every copy of PBXware and is fully supported. If you care to register you may order a trial of PBXware with our SOHO. Regards Steve steve 'at' bicomsystems 'dot' com - Original Message - From: Andres Paglayan To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 5:27 PM Subject: [asterisk-users] operator console Hi, My users are currently using an operator console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass, and extension,and voila. I'll be switching the company's phone system to Asterisk. I know * is way much more flexible and rich featured than the box we currently have, ...but I'll need to give the users a good mean to see what's going on, who is busy, easy transfer with click here and there, easy conference, easy queue handler, easy way to see/use many lines at the same time is there any best console they can use? I don't mind using a commercial product, if the only part we have to pay for is the gui, besides, we will buying the enterprise * version Thanks a bunch, Andres ___ --Bandwidth 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--Jordi NelissenE S C A U XBusiness IP Telephony www.escaux.com--Email from people at escaux.com does not usually represent officialpolicy of ESCAUX. See http://www.escaux.com/disclaimer for details.___--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] IAX2 one way audio
I am not sure if it will help but try to put notansfer=yes in ur iax2 extension (just experiment a bit ;) ).On 12 Nov 2006 17:48:13 -0500, joe a. ( [EMAIL PROTECTED]) [EMAIL PROTECTED] wrote: Experiencing one way audio using IAX2.I did see some other posts on this, and see there may be some internal issues with asterisk and one way audio.Can this be a widespread problem?So many seem to be using IAX, I find it puzzling. Some information points to this being a problem on asymmetrical connections.This is a decidedly asymmetrical connection, with 1.5 Mbs download and 256 kbs, upload. A satellite link, to boot.So, maybe this is a meltdown right from the start?Event the vendor of the IAX service was not too keen. Oddly, my first few connections worked fine (unexpectedly good audio, both ways).Being all happy and stuff, made a call to a client, to show off. Yep.could not hear me.Since then all calls have connected quickly, but are receive only.I've tried rebooting the asterisk box, changing jitter related stuff, no joy. It is behind a firewall, but I can see no packets dropped, related to the IP's involved.Anyway, if there are experiences to relate, please do.joe a.___ --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] Re: Moh stops immediately
What is the length of music on old mp3 file ? Maybe file is very short .On 13/11/06, zen Perry [EMAIL PROTECTED] wrote: Mac OS X, Asterisk 1.4 beta--- Martin Joseph [EMAIL PROTECTED] wrote: On 2006-11-12 23:08:05 -0800, zen Perry [EMAIL PROTECTED] said: I'm trying to set up the Music on Hold feature. However, when I place a call the moh starts and stops immediately and as a result I dont hear the audio. -- Started music on hold, class 'default', on channel 'SIP/XXX' -- Stopped music on hold on SIP/XXX NOTICE[380]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! My extensions.conf reads: exten = 2000,1,Answer exten = 2000,2,MusicOnHold(default) I've also tried: exten = 2000,1,Answer exten = 2000,2,MusicOnHold(default) exten = 2000,3,WaitMusicOnHold(20) exten = 2000,4,Hangup Which OS? Which asterisk? Mine does this also. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Cheap talk?Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com___--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] Slow to get dialtone when going off hook - bigproblem for me :(
You can also use waitexten = X,1,Wait(3)(for3secs) On 13/11/06, Jim Archer [EMAIL PROTECTED] wrote: --On Sunday, November 12, 2006 11:53 PM -0500 John Novack[EMAIL PROTECTED] wrote: Dovid B wrote: snip How hard would it be to have asterisk detect a dial tone ? I really can't say. I am not a C programmer, so I wouldn't even know where to start. Given that cheap dial up modems have, for the past ??20?? years, have been able to do just that, I would think it should have been an early consideration For those 1% of users, the last time I tried, the insertion of a whad no effect for pulse dialing either.Well thanks to everyone who responded, and thanks to multiple w's I am back in operation.I went off hook a bunch of times and the worst case seemedto be 3 seconds to get a dial tone (which is pretty bad).It's hard togoogle one letter, but I eventually found that each w is .5 seconds, so 7 w's were inserted to be safe.I also called Cox and griped but I doubtthat will do me any good.I am a C programmer, but I don't know anything about the inards ofAsterisk.However, I would expect that dial tone detection would be a function of the hardware, not the Asterisk software.The cheap modems dothis on board and export a simple command set.But I also don't knowanything about Digium's hardware either.Thanks again!I really appreciate the help! Jim___--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] Dial : Executing context/priority after bridge?
Put canreinvite=no in asterisk sip user extension.Someprovidersdonotsupportreinvitesandhenceyougetsilenceiguess. On 13/11/06, Yuri Veremeyenko [EMAIL PROTECTED] wrote: Hi, I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a user via SIP and playing a reminder file when the user picks the phone. I use Gizmo service for SIP and I'm able to call through it. However, when asterisk dials a number, Gizmo first answers then tries bridging 2 channels. Right after answer Asterisk starts playing the reminder.It obviously results in hearing silence when the call is bridged (you've picked the phone). I use Asterisk cmd Dial like this : exten = s,1,Dial(SIP/NUMBER,30,rA(announce)) which should play file announce to the called party once they answer. I also tried exten = s,1,Dial(SIP/NUMBER,30,rG(default^play^1)) which separates caller and callee,for the same purpose. Here's the asterisk console: -- Executing SetCallerID(SIP/sipphone-cbfb, NAME NUMBER) in new stack -- Executing NoOp(SIP/sipphone-cbfb, Dialing 011 to deliver file /usr/vt/result/200611135/test) in new stack-- Executing SetVar(SIP/sipphone-cbfb, __MSG=/usr/vt/result/200611135/98_011380673805838) in new stack -- Executing Dial(SIP/sipphone-cbfb, SIP/[EMAIL PROTECTED]|45|rA(/usr/vt/result/200611135/test)) in new stack-- Called [EMAIL PROTECTED]-- SIP/sipphone-ebf3 answered SIP/sipphone-cbfb -- Playing '/usr/vt/result/200611135/98_011380673805838' (language 'en')-- Attempting native bridge of SIP/sipphone-cbfb and SIP/sipphone-ebf3 It looks like Asterisk is starting to execute context/play announcement after the dialed channel answers. The problem is that Gizmo SIP first answers and then tries bridging (that's where the actual call is taking place), so my announcement is played before the call and when I pick up I just hear the silence. Is there a workaround or a way to make Asterisk play the message when the call is bridged? I use Asterisk CVS-HEAD built on 28 Oct 2006.Any advice is highly appreciated.YuriPS. I tried this on my local server with a local SIP account, and the bridge step was absent. So it worked. Is it then a Gizmo issue or a standard SIP way? ___--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] How to get CDR to show answered calls only
Its pretty easy . If you have mysql records enabled via a patch just do sql queryuse asteriskcdrdb;select * from `cdr` where billsec 0 ( if answered then billsec always greater than 0 or you cna also use disposition = 'ANSWERED' ) On 13/11/06, Olivier [EMAIL PROTECTED] wrote: Why is it awful ?Regards___--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] operator console
oops sorry i didnt saw quoted text of other user and it showed as first post in gmail draft so i thought u made a topic for that pbx ( so considered spam :P ) . Sorry again :)On 13/11/06, Jordi Nelissen [EMAIL PROTECTED] wrote: Vicky,my other post related to a Web GUI for asterisk. This post is related toan Operator Console. I am simply answering the user's question, so Idon't see why you would consider this to be spam, and I never read you can not send two mails to the list on the same day. ___ --Bandwidth 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] Mysql 6 second rounding
This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know some function to round this to 6 seconds ( or any custom valud like 30 seconds ) ..Suppose if billsec field is 3 seconds then it should round to 6 seconds , if its 13 second then it should round up to 18 seconds ( for 6sec pulse counting ) . What would be mysql function to do this ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with internet down
Why not directly use ip address in host= lineinextensions instead of dynamic address like sip.voipprovider.com .. temporary fix but it may work . On 13/11/06, Steve Langstaff [EMAIL PROTECTED] wrote: A search of google should turn up some recommendations about running alocal cacheing DNS proxy, or similar.I've never done it myself (the cacheing proxy, not the searching ongoogle) so I don't know the specifics. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Andre Luiz Martins Sent: 13 November 2006 15:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with internet down I believe that the problem really is fault of DNS lookups, but as I should proceed for resolve that??___--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 forward behind a nat
IF your asterisk server is behind NAT and no port forwarding is done then how can that static ip user/device reach it .Youwillhavetokeepasteriskserverinstaticipordoportforwardingtoacceptconnectionsfromoutside. ORmaybeididntunderstandsenarioproperlyhere.Isitlikeyour Server with SIP application (public_address) responds to sip calls made by any program ( like sjphone pc-pc sip ) . If thats case then asterisk should be able to call it like any other program or maybe theres nat scenario playing bad here :-/ . Can you port forwardfromfirewalltoasteriskserver ?? On 13/11/06, nik600 [EMAIL PROTECTED] wrote: On 11/12/06, nik600 [EMAIL PROTECTED] wrote: On 11/12/06, Vicky [EMAIL PROTECTED] wrote: Yep make the server with dynamic ip register to server with static ip ( sip or iax both will do but in sip keep nat=yes while making extension ) the problem is that the server with dynamic ip can't register on the other server! This is the situation: Server with SIP application (public_address) | | - - - Internet | | Firewall (NAT) | | Server Asterisk (private ip: 192.168.100.249/public ip:public_address_2) | Analogic Board | Telecom I want to make a call from Server Asterisk to the server with SIP Application. The SIP Application can't register to Server Asterisk (because the application can't do it, i know, it isn't a good thingbut this is the application) When The SIP Application receives a SIP call it responds (because a dummy SIP user is autoregistered on hisself) So i only have to make a call to SIP/[EMAIL PROTECTED] I've also tried to setup an asterisk server on my laptop, and make a call to SIP/[EMAIL PROTECTED] from the public_address network. It works! I only have to setup the Asterisk server in production to make a SIP call throw the NAT but without any SIP user registered on it. Can i do that? Many thanks to all maybe you need some other information?___--Bandwidth 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] DSl and more then 1 call
Does it happen when you make more than one call from you main voip server alone ? Or it happens when there are more than 1 call on your branch server ? Pin the problem is in which server first , If main server can handle 2-3 calls with no lag then its probably problem in branch server . On 13/11/06, Kelly Opal [EMAIL PROTECTED] wrote: Hi I have 2 asterisk servers running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch site and all calls go to server 1. If I make 1 call on server 2 everything is fine. If I make a 2nd call so there a two calls going at the same time the ping times go up to 2500 and above and the call quality is horrible. If I add a third call the system becomes unusable. But if you hang up all calls except 1 (it doesn't matter which one) it works fine again. Any help you could provide would be greatly appreciated.Kelly___ --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] Mysql 6 second rounding
Thx and what would the sql query be ?.Iplantoputadditionalfieldas6second.Howcanimakebillsecofvaluesofwholetablegetroundedandfilledinfield6second Sorry i am a noob with mysql :D On 14/11/06, James Coberly [EMAIL PROTECTED] wrote: sum(duration+(6-mod(duration,6) for summary of seconds divisible by 6, /60 for minutes On Tue, 2006-11-14 at 00:07 +0530, Vicky wrote: This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know some function to round this to 6 seconds ( or any custom valud like 30 seconds ) ..Suppose if billsec field is 3 seconds then it should round to 6 seconds , if its 13 second then it should round up to 18 seconds ( for 6sec pulse counting ) . What would be mysql function to do this ? ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--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] Mysql 6 second rounding
Hey thanx for that Marnus . Thatsworking just exactly how i wanted :).Damoniactuallycameupwithsamerow/60+0.5thenrounduptrickwheniwasdoingsomethingsameinexcelsheets:)anditsusefulforbillingin1minuteroundup(60secpulse)butifailedtogetitworkingfor6secondpulse. Marnus'ssqlqueryisperfect..nowsupposechargeis1.5cent/minthenicanused6secondroundup'dvalueandmultipleby0.15togetcallbillincentand it can also be used for 30 second pulse or any other valuewithsmallmodification ..Thx. On 14/11/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: Supposing you have an extra column called 6second: UPDATE cdr SET 6second=billsec+(6-mod(billsec,6) where 6second=0 if you want a decimal minutes column called billmin UPDATE cdr SET billmin=round((billsec/60)+0.5),1) where billmin=0 Vicky wrote: Thx and what would the sql query be ?.Iplantoputadditionalfieldas6second. Howcanimakebillsecofvaluesofwholetablegetroundedandfilledinfield6second Sorry i am a noob with mysql :D On 14/11/06, James Coberly [EMAIL PROTECTED] wrote: sum(duration+(6-mod(duration,6) for summary of seconds divisible by 6, /60 for minutes___--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 Ports (1000 to 2000 works)
actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur rtp.conf what range its using and open that in firewall . Default with asterisk is 1-2 unless changed . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: I was reading the posts and someone said about the default 1000 to 2000I see in the .conf the default is 1 to 2I found a service that gives inbound DID's in the firewall 5060 and1 - 2 is setup no workie on the DIDBut when I set 5060 , 1 - 2 and (Unblocked) 1000 - 2000Now the DID works fine.So you me what the standard is--Best regards,Al BochterBochter Services http://www.BochterServices.com/?t=EmailAre you outside of the US?Do you need to call US Toll Free Numbers?We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED](Cellular) 1-712-432-5401(Voip PBX) Free World DialUp: 780-217 EXT: 250WebSite: http://www.freeworlddialup.com/BUY and sell Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=emailFor new and used security items http://www.bochterservices.com/?j=storet=email_securityGOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Ports (1000 to 2000 works)
FRom voip-info.org# SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT # IAX2- the IAX protocol iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT # IAX - most have switched to IAX v2, or ought to iptables -A INPUT -p udp -m udp --dport 5036 -j ACCEPT # RTP - the media stream iptables -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT # MGCP - if you use media gateway control protocol in your configuration iptables -A INPUT -p udp -m udp --dport 2727 -j ACCEPTOpen all above ports and you should be good to go . Maybe you are recieving calls over iax and u havent opened iax2 port 4569 .. Anyway my server has all above ports opened and i have zero problems :) . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use: 1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask before So is there any standard portsBest regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Vicky wrote: actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur rtp.conf what range its using and open that in firewall . Default with asterisk is 1-2 unless changed . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: I was reading the posts and someone said about the default 1000 to 2000 I see in the .conf the default is 1 to 2 I found a service that gives inbound DID's in the firewall 5060 and 1 - 2 is setup no workie on the DID But when I set 5060 , 1 - 2 and (Unblocked) 1000 - 2000 Now the DID works fine. So you me what the standard is -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICEShttp://www.bochterservices.com/?j=platingt=email ___ --Bandwidth 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 Inbound (clean). Database: 0648-1, 11/13/2006 - 11/13/2006 4:03:01 PM ___--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 Ports (1000 to 2000 works)
Thereis definitely wrong in your setup . I have ipkall setup on my asterisk and dont have ports 1000-2000 open ( only 1-2,5060,4569 open ) . and incoming calls word fine for me . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: No 1000 to 2000 is not a typo.Well let me put some light on this..If you goto http://www.ipkall.com/and your firewall is set to 1 to 2 you WILL NOT get SIP calls from http://www.ipkall.com/ DID'sAs soon as you OPEN ports 1000 to 2000 to the PBX Server the calls fromhttp://www.ipkall.com/ will work fine. You DON'T have to make any changes to /etc/asterisk/rtp.confThis is what I ran into todaySo I guess you are right... It's a free for all on ports. Makes thingsharder to do.I think we need to get a better standard just to make this easier. // There's no standard - there are several different conventions adopted// by different vendors, though.Best regards,Al BochterBochter Services http://www.BochterServices.com/?t=EmailAre you outside of the US?Do you need to call US Toll Free Numbers?We can help you save money on calling US toll free numbers.Email for information: [EMAIL PROTECTED](Cellular) 1-712-432-5401(Voip PBX) Free World DialUp: 780-217 EXT: 250WebSite: http://www.freeworlddialup.com/BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=emailFor new and used security items http://www.bochterservices.com/?j=storet=email_securityGOLD PLATING SERVICEShttp://www.bochterservices.com/?j=platingt=email Peter Bowyer wrote: On 13/11/06, Al Bochter [EMAIL PROTECTED] wrote: Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use:1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask beforeSo is there any standard ports Both sides have to be willing to negotiate a port. Maybe your handytone has its own restrictions on RTP ports? As you now know, Asterisk doesn't care as long as you specify a range in rtp.conf. 1000-2000 must be a typo as ports 1024 are reserved and privileged. There's no standard - there are several different conventions adopted by different vendors, though. http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help. Peter___ --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 forward behind a nat
Yep make the server with dynamic ip register to server with static ip ( sip or iax both will do but in sip keep nat=yes while making extension ) On 12/11/06,
Rosli Sukri [EMAIL PROTECTED] wrote:
u need another box say box a with real/addressable ip address. create an iax entry in box a and have the private ip (box b) box register to box a. then you can do a Dial(IAX2/boxb/${EXTEN}) that will ring the extension connected to
your 192.168.100.249 boxhope that helps;)
On 11/12/06, nik600 [EMAIL PROTECTED]
wrote:
Hii have to forward a call from my asterisk server on another server but my server is behind nat.How can i setup my extension.conf?Actually i have set up it as follows:exten = 046566,1,Dial(
SIP/[EMAIL PROTECTED])my server has a private ip 192.168.100.249 and doesn't have a public ipIf i try to call
SIP/[EMAIL PROTECTED] from an adsl connection (with amodem, without nat) the call is routed succesfuly.If i try to forward the call from my server i cant route the call... (i send many INVITE but without any answer)
How can i fix it?many thanks in advance___--Bandwidth and Colocation provided by
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Re: [asterisk-users] dynamically modifying the dialplan?
I think its same as DND (do not disturb ) . It can be activated by *78 and deactivated by *79 . I use freepbx for configuration so i am not sure if its there in default asterisk setup . I snipped some part of my configuration from freepbx's config files
[app-dnd-on]exten = *78,1,Answerexten = *78,n,Wait(1)exten = *78,n,Macro(user-callerid,)exten = *78,n,Set(DB(DND/${CALLERID(number)})=YES)exten = *78,n,Playback(do-not-disturbactivated)
exten = *78,n,Macro(hangupcall,)[app-dnd-off]exten = *79,1,Answerexten = *79,n,Wait(1)exten = *79,n,Macro(user-callerid,)exten = *79,n,dbDel(DND/${CALLERID(number)})exten = *79,n,Playback(do-not-disturbde-activated)
exten = *79,n,Macro(hangupcall,)On 12/11/06, Norbert Zawodsky [EMAIL PROTECTED]
wrote:
Hi Brian,many thanks to you for your answers in the past! The always gave me thelittle bit of mising information...My Asterisk box is running fine now so I want to try the next step...And now to all of you
What I want to implement is to use 1 button of my snom-360 phone forfollowing purpose:If I leave my desk I press this button. A light should show up as anindicator/reminder. From this moment all calls to my extension should
immediately be transferred to my voicemail box.When I return I press the button again, the light goes off and all callsto my extension should ring my phone again.Now, can I achieve this with a static dialplan in
extensions.conf or doI have to use all that Realtime + DB magic?Many thanks,Norbert___--Bandwidth and Colocation provided by
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[asterisk-users] Asterisk billing
I am having asterisk working with cdr mysql patch and freepbx for configurations . It stores all records in mysql tables and i can do further post paid billing myself . I am looking for a simple system that can show a user live call logs via web interfaceasperaccountcodeonsipextensions(muchlikeasterisk-stats)butitshouldnotshowalllogsatsametime..ineeditasperaccountcode(likeextension777-999hasaccountcodeaccount1soitwouldshowonlycallsofthisextensionfrommysqltable( on some php page ). Ihaveseenastppandotherbillingsystemsbutidontneedthatmuchfunctionalityorcomplexityrightnow.Can someone suggest the easiest way to do this ?? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxee lag problems ?
I doubt how many days more voxee will survive . Its been a month nw and tehir support doesnt want to repair this . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re:[asterisk-users] register suddenly fails
For the time being try putting 212.41.253.181in hostname= line in ur sipconfig and it should work . Also check if you /etc/resolv.conf has correctdns list ( i guess it does bcoz OS canresolve) . Also check /etc/asterisk/dnsmgr.conf . Here's example :[general]enable=yes ; enable creation of managed DNS lookupsrefreshinterval=1200 ; refresh managed DNS lookups every n second - Forwarded message -- From: Norbert Zawodsky norbert at zawodsky.at To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users at lists.digium.com Date: Thu, 09 Nov 2006 20:28:59 +0100 Subject: [asterisk-users] register suddenly fails Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider inode fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout: -- Registration for '018904676 at voip.inode.at' timed out, trying again (Attempt #1) Nov 9 20:01:07 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.at Nov 9 20:01:07 WARNING[952]: chan_sip.c:5505 transmit_register: Probably a DNS error for registration to 018904676 at voip.inode.at, trying REGISTER again (after 20 seconds) Nov 9 20:01:27 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout: -- Registration for '018904676 at voip.inode.at' timed out, trying again (Attempt #2) Nov 9 20:01:28 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.at Nov 9 20:01:28 WARNING[952]: chan_sip.c:5505 transmit_register: Probably a DNS error for registration to 018904676 at voip.inode.at, trying REGISTER again (after 20 seconds DNS lookup works: root at asterina:~# ping voip.inode.at PING voip.inode.at (212.41.253.181) 56(84) bytes of data. 64 bytes from 212-41-253-181.inhouse-line.inode.at ( 212.41.253.181): icmp_seq=1 ttl=60 time=15.3 ms 64 bytes from 212-41-253-181.inhouse-line.inode.at (212.41.253.181 ): icmp_seq=2 ttl=60 time=15.9 ms --- voip.inode.at ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1001ms rtt min/avg/max/mdev = 15.375/15.669/15.963/0.294 ms Since I am sure that I didn't change anything within the last week, I called inode support. But they said, that they didn't change anything either. Next I tried was a 'SIP RELOAD' which produced following output: asterina*CLI sip reload Nov 9 20:02:48 WARNING[952]: acl.c:244 ast_get_ip_or_srv: Unable to lookup 'h. } ' Nov 9 20:02:48 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.at Nov 9 20:02:48 WARNING[952]: chan_sip.c:5505 transmit_register: Probably a DNS error for registration to 018904676 at voip.inode.at, trying REGISTER again (after 20 seconds) Nov 9 20:03:08 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout: -- Registration for '018904676 at voip.inode.at ' timed out, trying again (Attempt #1) Nov 9 20:03:08 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.at Nov 9 20:03:08 WARNING[952]: chan_sip.c:5505 transmit_register: Probably a DNS error for registration to 018904676 at voip.inode.at, trying REGISTER again (after 20 seconds) asterina*CLI Now, what makes me wonder ist the first line after the reload which says Unable to lookup 'h. }. Anybody of you got any idea ?? Norbert ___ --Bandwidth 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] Voxee lag problems ?
Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak time even when no high latency ).Noproblemswithanyotherprovider . Anyone else having same problem ? ___ --Bandwidth 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] Voxee lag problems ?
Yeh voxee is really of no use now . You can try voipjet though .. even though they dont have good support but i hardly had any big problems with them . Also there is icall but i wont recommend that right now but their rates are pretty good . - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 09, 2006 9:11 PM Subject: Re: [asterisk-users] Voxee lag problems ?Hi Vicky, I used to use their termination services, but I had the same problems ... It is impossible to work with that latency ... A lot of gaps and dead spots ... If you know ... I'm looking for a good termination provider that I can use the combination IAX/iLBC ... If you know some .. can you please tell me ? 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] register suddenly fails
For the time being try putting 212.41.253.181in hostname= line in ur sip config and it should work . Also check if you /etc/resolv.conf has correct dns list ( i guess it does bcoz OS can resolve).Alsocheck/etc/asterisk/dnsmgr.conf. Here's xample :[general]enable=yes ; enable creation of managed DNS lookupsrefreshinterval=1200; refresh managed DNS lookups every n seconds - Forwarded message --From: Norbert Zawodsky [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate: Thu, 09 Nov 2006 20:28:59 +0100Subject: [asterisk-users] register suddenly failsHi everybody,I've got a very strange problem:As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls.Today I noticed that outbound calls to provider inode fail (andinbound from this provider too). On the CLI I get every 20 secondsfollowing messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout:--Registration for '[EMAIL PROTECTED]' timed out, trying again(Attempt #1)Nov 9 20:01:07 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.atNov 9 20:01:07 WARNING[952]: chan_sip.c:5505 transmit_register:Probably a DNS error for registration to [EMAIL PROTECTED] , tryingREGISTER again (after 20 seconds)Nov 9 20:01:27 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout:--Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #2)Nov 9 20:01:28 WARNING[952]: chan_sip.c:1998 create_addr: No such host:voip.inode.atNov 9 20:01:28 WARNING[952]: chan_sip.c:5505 transmit_register:Probably a DNS error for registration to [EMAIL PROTECTED], tryingREGISTER again (after 20 secondsDNS lookup works:[EMAIL PROTECTED]:~# ping voip.inode.atPING voip.inode.at (212.41.253.181) 56(84) bytes of data.64 bytes from 212-41-253-181.inhouse-line.inode.at (212.41.253.181):icmp_seq=1 ttl=60 time=15.3 ms64 bytes from 212-41-253-181.inhouse-line.inode.at ( 212.41.253.181):icmp_seq=2 ttl=60 time=15.9 ms--- voip.inode.at ping statistics ---2 packets transmitted, 2 received, 0% packet loss, time 1001msrtt min/avg/max/mdev = 15.375/15.669/15.963/0.294 msSince I am sure that I didn't change anything within the last week, Icalled inode support. But they said, that they didn't change anythingeither.Next I tried was a 'SIP RELOAD' which produced following output: asterina*CLI sip reloadNov 9 20:02:48 WARNING[952]: acl.c:244 ast_get_ip_or_srv: Unable tolookup 'h. }'Nov 9 20:02:48 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.atNov 9 20:02:48 WARNING[952]: chan_sip.c:5505 transmit_register:Probably a DNS error for registration to [EMAIL PROTECTED], tryingREGISTER again (after 20 seconds) Nov 9 20:03:08 NOTICE[952]: chan_sip.c:5422 sip_reg_timeout:--Registration for '[EMAIL PROTECTED]' timed out, trying again(Attempt #1)Nov 9 20:03:08 WARNING[952]: chan_sip.c:1998 create_addr: No such host: voip.inode.atNov 9 20:03:08 WARNING[952]: chan_sip.c:5505 transmit_register:Probably a DNS error for registration to [EMAIL PROTECTED] , tryingREGISTER again (after 20 seconds)asterina*CLINow, what makes me wonder ist the first line after the reload which saysUnable to lookup 'h. }.Anybody of you got any idea ?? Norbert ___ --Bandwidth 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] unable to execute call file
hi all, i am trying to execute a call file in asterisk by placing it in the outgoing directory.In my system i m running rtpproxy and openser also.Asterisk is communicating with openser because i am able to make incoming calls to asterisk.But when i try to put call file in the outgoing directory we are getting the following errors. Attempting call on SIP/[EMAIL PROTECTED] for application Playback(demo-congrats) (Retry 1) -- Got SIP response 482 Loop Detected back from 192.168.0.111 Jan 2 16:58:51 NOTICE[4685]: pbx_spool.c:266 attempt_thread: Call failed to go through, reason 8 Can anyone help me regarding this please?? with regards vicky ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] unable to execute call file
hi, thanks for reply. even after specifying the port, we are getting the same error. with regards vicky On Mon, 02 Jan 2006 Karsten Wemheuer wrote : Hello, as You are running two processes handling SIP (asterisk and openser), I think the Call-File addresses the wrong instance. If Your callfile contains a line like Channel: SIP/accountname try something like Channel: SIP/[EMAIL PROTECTED]:port where ipaddress and port addressing the responable instance. HTH, Karsten ___ --Bandwidth 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] need help in building dynamic conference
hi all, can any one helpme in how to invite a user(exisiting person) to an already started conference, by using meetme app. in asterisk. hope every got what i mean. with regards asteriskuser ___ --Bandwidth 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] need help in building dynamic conference
hi all, can any one helpme, how to invite a user to an already started conference by using meetme app. in asterisk. with regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice alternatives
Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAC Access Bank Manual
Hi, The asterisk configuration and the channel bank configuration are both set to esf and b8zs. Howerver I am still getting the framing Error Red and blinking. zttool shows there are no alarms. According to the manual, Framing Error (Red and Blinking )means Network T1 is out of frame (received signal cannot be framed to ESF or D5 as configured by T1 Option switch 4) I tried with both DIP switch on and off, but no help. Any ideas ? Is my card or channel bank bad ? On Thursday 17 March 2005 19:12, Jerry wrote: Carrier Access generally have all of their manuals available for download. You just have to request a free login. they also provide excellent dialin support - also free. If your framing LED is blinking I would double check that both ends of your span are set for ESF. zttool is the tool for working on the cards. On Mar 17, 2005, at 4:40 AM, Vicky Shrestha wrote: Hi, Does anyone have Carrier Access Corporation (CAC) Access Bank I Manual ? Could you please email it to me off list ? We have a FXS channel bank and the framing Error Led is blinking and I have no clue on what could be the problem . Is there command line utilities available in Linux to Troubleshoot T1 connection using Zaptel drivers ? /etc/zaptel.conf = span=1,1,0,esf,b8zs #span=1,1,0,esf,ami #span=1,1,0,d4,b8zs #span=1,1,0,d4,ami #em=1-24 fxols=1-24 loadzone=us defaultzone=us == /etc/asterisk/zapata.conf = [channels] language=us context=default signalling=fxo_ls ;usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes ;threewaycalling=yes transfer=yes ;cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel = 1-24 === dmesg output = Zapata Telephony Interface Registered on major 196 Found TE410P at base address dfcdff80, remapped to d0e23f80 TE410P version c01a009b, burst ON FALC version: 0005, Board ID: 00 Reg 0: 0x0e3c6800 Reg 1: 0x0e3c6000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1000 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 0 (United States / North America) TE410P: Span 1 configured for ESF/B8ZS SPAN 1: Primary Sync Source == -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAC Access Bank Manual
Hi, Does anyone have Carrier Access Corporation (CAC) Access Bank I Manual ? Could you please email it to me off list ? We have a FXS channel bank and the framing Error Led is blinking and I have no clue on what could be the problem . Is there command line utilities available in Linux to Troubleshoot T1 connection using Zaptel drivers ? /etc/zaptel.conf = span=1,1,0,esf,b8zs #span=1,1,0,esf,ami #span=1,1,0,d4,b8zs #span=1,1,0,d4,ami #em=1-24 fxols=1-24 loadzone=us defaultzone=us == /etc/asterisk/zapata.conf = [channels] language=us context=default signalling=fxo_ls ;usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes ;threewaycalling=yes transfer=yes ;cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel = 1-24 === dmesg output = Zapata Telephony Interface Registered on major 196 Found TE410P at base address dfcdff80, remapped to d0e23f80 TE410P version c01a009b, burst ON FALC version: 0005, Board ID: 00 Reg 0: 0x0e3c6800 Reg 1: 0x0e3c6000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1000 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 0 (United States / North America) TE410P: Span 1 configured for ESF/B8ZS SPAN 1: Primary Sync Source == -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(
Hi,
I tried that but same error
Specially I didn't find people posting about Bad Request or Unknown Dialog
-- Got SIP response 400 Bad request back from 147.135.8.128
-- SIP/5092321848-ccd4 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/502-f815, 5) in new stack
-- Got SIP response 400 Bad request back from 147.135.8.128
-- Got SIP response 481 Unknown Dialog back from 147.135.8.128
On Friday 11 March 2005 02:30, Dan Weber wrote:
Here is my sip.conf:
===
register =
[EMAIL PROTECTED]::[EMAIL PROTECTED]/broadvoi
ce
[]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=
secret=
username=
insecure=very
context=default
authname=
dtmfmode=inband
dtmf=inband
canreinvite=no
Contact information must not change between register and call. Whats
happening here is that when you register its [EMAIL PROTECTED], however,
when you call, its [EMAIL PROTECTED] Change the extension of the
register to match your phone number.
Dan
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Content preview: Here is my sip.conf:
=== register =
[EMAIL PROTECTED]::[EMAIL PROTECTED]/broadvoice
[] type=peer user=phone host=sip.broadvoice.com
fromdomain=sip.broadvoice.com fromuser= secret=
username= insecure=very context=default
authname= dtmfmode=inband dtmf=inband canreinvite=no
[...]
Content analysis details: (0.1 points, 5.0 required)
pts rule name description
--
-- 0.1 FORGED_RCVD_HELO
Received: contains a forged HELO
--
With regards,
Vicky Shrestha
System Director
WorldLink Communications
Jawalakhel , Kathmandu, Nepal
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Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(
It doesn't try to authenticate the incoming call. On Friday 11 March 2005 03:56, Randy Johnson wrote: What does insecure=very do? Dan Weber wrote: Here is my sip.conf: === register = [EMAIL PROTECTED]::[EMAIL PROTECTED]/broadvo ice [] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser= secret= username= insecure=very context=default authname= dtmfmode=inband dtmf=inband canreinvite=no Contact information must not change between register and call. Whats happening here is that when you register its [EMAIL PROTECTED], however, when you call, its [EMAIL PROTECTED] Change the extension of the register to match your phone number. Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and Broadvoice Outgoing Again :(
Thanks, I have that already in my /etc/hosts But it's still not working :( On Saturday 12 March 2005 03:48, Rich Adamson wrote: For everyone that's trying to get BV to work, you'all might want to edit your /etc/hosts file and insert something like: 147.135.8.128 sip.broadvoice.com This was a requirement from way back and I've since discontinuted BV for a different provider, but seems as though of all the suggestions posted in recent weeks, few mention the above. After editing /etc/hosts, there is no need to reboot, etc. The contents are read dynamically. Then make sure that your contexts and extensions.conf use sip.broadvoice.com in them. They did have four different servers at one time (with four different IP's), but if you stick with one (like the above) and play with the other parameters to get it to work, then you can change servers at a later time. As one more comment, any changes that you make to sip.conf or extensions.conf associated with trying to make BV work, don't forget to stop and restart asterisk. Don't rely on a reload as it does not reread all parameter changes. I can't make outgoing calls via Broadvoice. I have tried each and every configuration that was posted to list previously. I am able to receive incoming calls fine. I get the following in asterisk console: = asterisk*CLI show version Asterisk CVS-HEAD-03/10/05-22:51:28 built by [EMAIL PROTECTED] on a i686 running Linux asterisk*CLI -- Executing Dial(SIP/502-c147, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 400 Bad request back from 147.135.8.128 -- SIP/-19dd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Congestion(SIP/502-c147, 5) in new stack == Spawn extension (vicky, 0018086749157, 2) exited non-zero on 'SIP/502-c147' -- Got SIP response 400 Bad request back from 147.135.8.128 -- Executing Dial(SIP/502-8efd, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 400 Bad request back from 147.135.8.128 -- SIP/-4bf5 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Congestion(SIP/502-8efd, 5) in new stack == Spawn extension (vicky, 008086749157, 2) exited non-zero on 'SIP/502-8efd' -- Got SIP response 400 Bad request back from 147.135.8.128 -- Got SIP response 481 Unknown Dialog back from 147.135.8.128 Here is my sip.conf: === register = [EMAIL PROTECTED]::[EMAIL PROTECTED]/broadvoi ce [] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser= secret= username= insecure=very context=default authname= dtmfmode=inband dtmf=inband canreinvite=no -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and Broadvoice Outgoing Again :(
Hi, I can't make outgoing calls via Broadvoice. I have tried each and every configuration that was posted to list previously. I am able to receive incoming calls fine. I get the following in asterisk console: = asterisk*CLI show version Asterisk CVS-HEAD-03/10/05-22:51:28 built by [EMAIL PROTECTED] on a i686 running Linux asterisk*CLI -- Executing Dial(SIP/502-c147, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 400 Bad request back from 147.135.8.128 -- SIP/-19dd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Congestion(SIP/502-c147, 5) in new stack == Spawn extension (vicky, 0018086749157, 2) exited non-zero on 'SIP/502-c147' -- Got SIP response 400 Bad request back from 147.135.8.128 -- Executing Dial(SIP/502-8efd, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 400 Bad request back from 147.135.8.128 -- SIP/-4bf5 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Congestion(SIP/502-8efd, 5) in new stack == Spawn extension (vicky, 008086749157, 2) exited non-zero on 'SIP/502-8efd' -- Got SIP response 400 Bad request back from 147.135.8.128 -- Got SIP response 481 Unknown Dialog back from 147.135.8.128 Here is my sip.conf: === register = [EMAIL PROTECTED]::[EMAIL PROTECTED]/broadvoice [] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser= secret= username= insecure=very context=default authname= dtmfmode=inband dtmf=inband canreinvite=no -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
