[asterisk-users] RE : Re: differential billing
Stop advertising. Le 26 sept. 2010 09:46, Faisal Hanif fai...@vopium.com a écrit : Hi Abdul-Basit, If you need only different intervals of billing you can easily do it using any AGI as we are doing it in Perl AGIs using post call billing. But if you need realtime billing then the most stable and flexible option is to use FastAGI+ AMI. I have tested it in JAVA and it worked for me up to a load 100 calls. It may work more but I haven't tested it. Asterisk and Billing-Server was running on separate machines. For further help you can call me (as you know my number :P). Regards, Faisal Hanif -- _ -- Bandwidth and Colocation Pr... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi not available in Asterisk
Have you installed dahdi ? And do not mix 1.4 with 1.6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mail-2-Fax and Fax-2-Mail solution for Asterisk with T38
It does not support T.38 is that correct ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?
Have you tryed to generate .call files at once ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ACD ASR
2009/10/14 B.Masoud @ SH i...@saudihome.com Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source destination? Thanks. You can calculate by yourself with cdr's, its only statistics. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
Allmost your solutions require second server or some hardware, why do you use shorewall ? Its a iptables rule generator with a friendly config files. Mine was up and running in 30 min or reading some docs. And you can trace all traffic live. Good day. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcement: Howler-optimised G.729A Solution for Asterisk
2009/6/24 Senad Jordanovic se...@bicom.us Jay Fenton wrote: [ Optimised G.729A 'Howlet' for Asterisk FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. Why would someone buy it instead of Digium g729 codec? Concurrence is good. And the floating model across many server is interesting idea. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modifying CID for different trunks
2009/6/17 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi,
I have 2 trunks connected to my asterisk installation.
One is a inbound connection between ericsson pbx and the other is thru a
voip service.
I am using 4 digit numbers both in ericsson and asterisk..
And also i have full real prefix for that numbers..
As all 290 are real numbers and if smbody dials 290 from outside
starts to ring without a problem.
Now.. My problem is, from asterisk side, both ericsson and trunk are
outside trunks..
So.. if i dont enter any Caller ID for an extension, 4 digit caller
id(internal CID) is sent to both trunks.. This is waht i want for ericsson
trunk but not the other one..
So, how can i modify CID for a single trunk, so if i dial ericsson i can
use 4 digit one, and if i dial other trunk i can add smthing like 290 to
all outgoing CID data..
Hello, you can try to do a Set(${Callerid(num)}=290XXX) before call if you
are using asterisk 1.2 the $var is different, search wiki. Tell us if it
works for you. Also your provider need to acept your number, so be sure he
is.
Have a good day.
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Re: [asterisk-users] modifying CID for different trunks
Make sure you are actually setting it as:
Set(CALLERID(num)=290)
The previous poster has the formatting incorrect. If your callerID is a 4
digit
number, and you want to modify it to have the prefix on it before you send
it
back out, you can do:
Set(CALLERID(num)=290${CALLERID(num)})
Alternatively you could make the 290 a variable, then set it prior to
calling
the Set() application if you needed the prefix to be set based upon some
logic.
Thank you for correcting me. I forgot that actually Set() does no need that
${} brackets to set that $var.
And about prefix, it's can be done with many ways my direct setting number
was just an exemple.
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Re: [asterisk-users] OT - Aastra phones provisioning
2009/6/11 Olivier oza-4...@myamail.com Hi, I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP root directory. For instance, TFTP root directory is /srv/tftp. When config files are stored in /srv/tftp, a new Aastra can find its config files. When config files are stored in /srv/tftp/aastra, a new Aastra can't find its config files. I tried to using DHCP root-path option to tell Aastra phones to search the right subdirectory, but it doesn't seem to work. Any advice on this ? Regards How about /srv/tftp/aastraphones ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer call from analog telephone
Remember that the time between the two digits is VERY short. You must press those two digits in quick succession or else the requested feature code will not activate. - Or set featuredigittimeout longer. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] connection fail between Service provider's proxy server and my asterisk server
2009/5/29 김무성 ki...@infosec.co.kr
I wanna connect proxy server.
my IP Phone - my asterisk - service provider's proxy server - extern
PSTN phone
but asterisk server can't register to proxy server.
I think that configuration is right.
When asterisk send to register request, proxy server don't response.
I did capture packet. but no response.
MY setting
sip.conf
[kms]
username=kms
type=friend
secret=
host=dynamic
nat=yes
qualify=yes
callerid=0134
register = 0700134:passw...@proxy.sp.co.kr:5060/0134
[my-out]
type=peer
host=SP's proxy IP
username=0700134
secret=password
fromuser=0700134
fromdomain=proxy.SP.co.kr
extensions.conf
[default]
exten = _X.,1,Dial(SIP/${ext...@my-out)
If lines provided is not a form of trunk, can't my asterisk server connect
to proxy?
I could connect my IPPhone to proxy directly.
but asterisk not.
We need the sip trace for the call.
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Re: [asterisk-users] VoIP over satellite internet
2009/5/9 Don E. Wisdom d...@engineeringinc.com I work on the salmon river in Idaho as a computer/radio tech. All of the satellite isp's do not have the upstream capability. Skype barely works. (you have to try upwards of 20 times for it to work) If I have to make phone calls when I am there I always use the SSB Radiophone or satellite phone because it is far far far more reliable and doesn't irritate the living hell out of the person your calling. I have tried 2-3 different VoIP providers all have the exact same result. The other side only hears a few pieces of word or nothing at all and hangs up. I have tried this on Starband (360 480 modems) wild blue Starband also has outages during the day where you cant see their satellite. Most of the satellite ISP's also have rolling bandwith caps. (Starbands is 1gig down 300megs up in a 7 day period for the plans I deal with) Overall I think its a bad idea. It most likely will not work. --Don Hello, i did once install in south Africa, and the only problem i had is the delay, however the client has the dedicated 2 mbit uplink. But when i talked over it the delay was really noticable. On 5/8/09 10:56 PM, Frank Bulk frnk...@iname.com wrote: If people don't mind taking turns talking, it will work. It's just going to be like talking on a CB. Reminds me of talking to my grandparents in the Europe as a child in the early 80's. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Fort Sent: Friday, May 08, 2009 10:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VoIP over satellite internet Could those on the list who have used or tried to use VoIP over a satellite internet connection comment on how well it works or if it even works at all in a reliable way. What is the effect of latency on the VoIP path and how much is generally tolerable? routing via satellite adds about a quarter second of latency to the path. Is that too much? Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over satellite internet
Forgot to add, it is no so bad, i mean if you are in situation where local telco male you pay hell of a price. Or if you are in location not covered by any telco, i would go by sattelite option. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Professional Setup..
Not a taboo at all, you are providing your knowledge to setup the call center for example, and i your support in future. It is commen practice. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk blade server
2009/5/9 Dean Collins d...@cognation.net Perfect office rackmount asterisk server? http://www.tgdaily.com/html_tmp/content-view-42372-135.html Lacking dual hdd for raid 1. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Professional Setup..
2009/5/9 Steve Edwards asterisk@sedwards.com On Fri, 8 May 2009, Dave Walker wrote: I have a question for those who have done a few professional installs of Asterisk. Is it taboo to use something like AsteriskNow/FreePBX/Trixbox to get a base installation of Asterisk installed and functional for a small office? If not then do you always compile from scratch or use CentOS and the yum repositories? I used Asterisk At Home (predecessor of Trixbox) for my first couple of installs. I didn't need most of the cruft included and never took the time to understand all the funny little things that were done behind my back to make life easy for someone with little to no Linux skills. Fortunately, I was able to replace those systems before they were hacked by default passwords and buggy code. Now, I install a minimal CentOS (de-selecting every single package) and yum in just what I need. Then I install Zaptel, Libpri, and Asterisk from source. (I'm a 1.2 Luddite.) Parts left out don't get broke. Thanks in advance, Building everything from scrach each time is good when you have time, but when you want a nice web interface for cdr stats, end point management and good auto provisioning, it is not an option. Security issues are standing on good knowledge of the system and experience, to do not encounter default passwords and buggy code issues. Have fun. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma a104d and channel banks
2009/5/7 Jim Dickenson dicken...@cfmc.com I have * 1.6.0.9 with dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.2, libpri 1.4.10 and wanpipe-3.4.1 running on CentOS 5.3 64bit. I have 2 ports of the a104d configured for use with PRI lines and 2 ports configured for use with Adtran Total Access 850 channel banks. The channel banks have 6 four port FXS cards in them. The PRI lines work as expected. I can call a phone connected to the channel bank and this works as expected. If I pick up the phone connected to the channel bank and dial a digit I immediately get a fast busy tone. The exceptions is that if I dial *7anything then I can dial these three keys. Nothing of much interest shows on the * console other then when I hang up the phone this event is reported. If while talking on the phone when someone calls me, I press a key on the phone connected to the channel bank, the person called hears a faint pop and then the pressed dtmf tone. Any ideas as to what might cause this behavior? Thanks for any ideas! -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ Look if your channel bank dont have any fancy dialplan configured. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma a104d and channel banks
2009/5/7 Jim Dickenson dicken...@cfmc.com *From: *Grygoriy Dobrovolskyy megaho...@gmail.com *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Date: *Thu, 7 May 2009 12:20:07 +0200 *To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject: *Re: [asterisk-users] Sangoma a104d and channel banks 2009/5/7 Jim Dickenson dicken...@cfmc.com I have * 1.6.0.9 with dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.2, libpri 1.4.10 and wanpipe-3.4.1 running on CentOS 5.3 64bit. I have 2 ports of the a104d configured for use with PRI lines and 2 ports configured for use with Adtran Total Access 850 channel banks. The channel banks have 6 four port FXS cards in them. The PRI lines work as expected. I can call a phone connected to the channel bank and this works as expected. If I pick up the phone connected to the channel bank and dial a digit I immediately get a fast busy tone. The exceptions is that if I dial *7anything then I can dial these three keys. Nothing of much interest shows on the * console other then when I hang up the phone this event is reported. If while talking on the phone when someone calls me, I press a key on the phone connected to the channel bank, the person called hears a faint pop and then the pressed dtmf tone. Any ideas as to what might cause this behavior? Thanks for any ideas! -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ Look if your channel bank dont have any fancy dialplan configured. -- The context the channel bank phones are in includes one context that only has extensions 111, 222, 333, 444 and 555. I am not taling about the dialplan inside asterisk, i dont know if you channel bank is managable or not, like for example aastra phones has their dialplant configured inside teh phone. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Asterisk 1.6 web GUI
2009/4/20 Gary Li garyli0...@gmail.com Hi, I had some experience on Asterisk 1.0.7 and 1.2.0. Now, I want to do something on the New Release of Asterisk 1.6.xx. I want to know wheather there are already web GUI for use now in the release. Or still nedd integrate some other third part GUI? Any advice will be appreciated. Thanks ahead, *Best Regards,* *Gary*** Gui with version 1.6 has Elastix and AsteriskNow. Not sure about druid destribution. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Here is Step by Step Example of Asterisk PBX System Install and configuration
On the last page http://qvlweb.blogspot.com/2009/04/asterisk-pbx-system-install-04-pbx-test.html there is a small screen, number 3 from bottom, looks like you are editing exgensions.conf not extensions.conf. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY
2009/4/1 Michael mich...@networkstuff.co.nz haw haw haw... April Fools Day is over in this part of the world. Hey dont kill the magic ! :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote host can't match request CANCEL to call
2009/4/1 Shaun Wingrin voi...@gmail.com Hi, Why does this warning occur and what are the implications of it? I'm concerned about calls never getting hung up.! chan_sip.c:12890 handle_response: Remote host can't match request CANCEL to call '2f197e56611061a678c13b881b269...@411.2.139.106'. Giving up. Tx Hello It's other end who is not aware if the call leg for that cancel, it is happening when some provider missconfigured the load balancing stuff for example, or call leg allready was destroyed for any reason. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay's SIP for Skype
2009/3/27 Marco Sambo derwid...@gmail.com I have to try Skip2PBX, integrated into my Asterisk machine, but it seem more invasive than Gizmo5 opensky. Doesn't it? Marco Skip2pbx is based on freebsd so i dont think thank you can install it on the same pc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Know who's logged in
2009/3/27 Mr. James W. Laferriere bab...@baby-dragons.com Hello Mark Miquel , On Thu, 26 Mar 2009, Mark Michelson wrote: Miguel Molina wrote: Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: # asterisk -rx show channel SIP/303-b2f1c368 -- General -- Name: SIP/303-b2f1c368 Type: SIP UniqueID: 1238094839.425549 Caller ID: 303 Caller ID Name: Ext. 303 DNID Digits: 7700 State: Up (6) Rings: 0 NativeFormats: 0x2 (gsm) WriteFormat: 0x2 (gsm) ReadFormat: 0x2 (gsm) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 111 Frames in: 6199 Frames out: 4847 Time to Hangup: 0 Elapsed Time: 3h29m16s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: XXX Extension: X Priority: XX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=X SIPURI=sip:3...@x CDR Variables: level 1: clid=Ext. 303 303 level 1: src=303 level 1: dst=XX level 1: dcontext=XXX level 1: channel=SIP/303-b2f1c368 level 1: lastapp=AgentLogin level 1: start=2009-03-26 14:13:59 level 1: answer=2009-03-26 14:13:59 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1238094839.425549 Is there an option for Agentlogin() to set a channel variable on the login channel that contains the code of the agent that successfully logged in? If not, would this be hard to accomplish by tweaking the chan_agent.c code to do that? It would be a really nice feature. I'm using asterisk 1.4.22. Thanks for any clue on this, There is a CLI command agent show which will list all agents. This output will show the agent's number, name, whether he/she is logged in, and moh class. Similarly, there is a command agent show online which will only list logged-in agents. Mark Michelson There does not seem to be a 'agent' command in 1.4.2x . asterisk-2*CLI core show version Asterisk 1.4.21.2 built by root @ asterisk-2 on a i686 running Linux on 2009-01-07 05:57:09 UTC asterisk-2*CLI agent No such command 'agent' (type 'help agent' for other possible commands) And he mentions 1.4.22 . Now unless I've misconfigured my compile of 1.4 then ... Hopefully there is a differant command ? Tia , JimL -- I would like to find a way to do it in asteris 1.2 'show agents' do not show me all agents, i have 30 agents connected to a queue and show agents show me 6 and they are offline. So is there any way to know how many agents are logged in ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sisky to connect Skype to Asterisk
2009/3/26 Alejandro Cabrera Obed aco1...@gmail.com Dear all, I've read some news about Sisky (http://www.yeastar.com/Products/SiSkyEE.asp), a service to interconnect Skype clients with SIP clients. Does anybody test Sisky and can tell me about his experience ??? (Sisky runs on Windows because Skype and its API are more stable on this OS). Regards, Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have tested, the quality suffers from normal skype call, and far behind a good voip quality. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay's SIP for Skype
skip2pbx is the best i tryed, but nasty price ;) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk with 250 lines
2009/3/24 Christian Victor christ...@victormedia.de Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as this will be some kind of a hosted pbx. Given he finds a provider wich has this much SIP capacity and IP bandwith and no codec conversion is needed - do you think this is possible with pure asterisk on a decent system? Is there anything I shoudl watch out for? Your help is much appreciated! Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If the switch is fine why not ? But i wander why kind if switch is that 240-480 fxo ? ;) Sounds like a big overkill. And i dont see a problem with asterisk, if not too much transcoding involved and with the right hardware. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Global videoconferencing solution.
Hello everybody, i am searching a solution for a videoconferencing, Any solution (Free/commercial). Asterisk is a great software, but recently we have more and more demands about videoconferencing of 3 or more peoples, Existing solutions are heavy and costly, around 2500€ for 1 client. This is insane. Is there any solutions out there for non millionaires ? Or even Free ? I remember a company who sold his software called cu see mee There were some conference rooms, used webcams 12 ppl max as remember. It could be perfect. Thank you for giving me advices. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with large user base?
2009/3/17 zoach...@securax.org zoach...@securax.org Vincent Li wrote: Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. Asterisk can scale to 10.000 users. Its probably about the maximum you could do on a quite powerful server if you don't need TDM hardware, but better would be to use a cluster, the database used would then eventually become the limit to the scaling. I have no experience with SipX so i can't say if it will scale better without clustering. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running in my office for small user base, I don't have experience with large scale Asterisk implementation. I know little about sipX. Does anyone in the community has any input about this? Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello i suggest opensips/kamalio for register server role and asterisk for a voicemail server and to pstn/pri/whatever gateway. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to get 60+ analogue extensions.
2009/3/16 Alex Balashov abalas...@evaristesys.com I don't know how good Asterisk's GR.303 support, but you could use DLCs as well. However, that's a lot of complexity and (seemingly) immature functionality liability to achieve the same end you'd get with a channel bank. The only benefit is that DLCs are specifically for oversubscription, whereas on PRIs you'd be doing one timeslot per one POTS line on the trunk side. On Mon, 16 Mar 2009 18:48:10 -0400, C F shma...@gmail.com wrote: Channel Banks would be the way I would do it. On Sun, Mar 15, 2009 at 3:12 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. Thanks very much Cheers Duncan ___ xorcom 2x32 fxs and done ;) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK ISDN-30 and ANI
2009/3/13 Andrew Thomas a...@datavox.co.uk I think I understand what you mean now. The biggest difference between CLI and ANI is that ANI can't be blocked/withheld (like you can with CLI by using 141). It also uses different signalling. This is mainly used by law enforcement agencies to trace calls etc. So, you want the number - regardless of what the user dials. I presume you are some sort of 'carrier' then. You'll be lucky to get the information otherwise as it throws up all sorts of privacy laws (ie. you have to have a damn good reason for wanting it). BT are the main people to ask I suppose (unless your calls go through another main carrier). I'm not even sure if ANI signalling is implemented in Asterisk - one for the config file writers ;). Cheers I am sure of one thing that i can do a sip trunk with ANI in our billing system, not sure how it works, but the option is there ;) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Printing faxes
2009/3/12 Tristan tris...@telemaque.fr Hi, Send it to cups via the FaxDispatch script ;) Regards, Tristan voip crazy a écrit : Hello list, I have an asterisk / hylafax / iaxmodem configured in one machine. All is working nicely. Now I need the fax to be print when arriving. ¿Anybody have this feature implementing in their systems? ¿How is the best way to get that? Any clue will be welcomed. Thanks. VoipCrazy ___ Oh sorry i meant Faxdispatch not faxrcvd ;) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Printing faxes
2009/3/12 voip crazy voipcr...@gmail.com Hello list, I have an asterisk / hylafax / iaxmodem configured in one machine. All is working nicely. Now I need the fax to be print when arriving. ¿Anybody have this feature implementing in their systems? ¿How is the best way to get that? Any clue will be welcomed. Thanks. VoipCrazy ___ I am printing at the end of faxrcvd script, just configure printer you like read the script, it is pretty simple, and add some command for lpd for that $File. Hope i helped. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to do Load-Balancing for Asterisk with OpenSIPS
2009/3/10 Ali Jawad alijaw...@gmail.com Great Job Bogdan On Tue, Mar 10, 2009 at 12:52 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi, When trying to cluster Asterisk boxes to gain scalability and more performance, there is now a new simple and efficient solution for doing it. OpenSIPS/OpenSER 1.5 can now implement traffic routing based on load. Shortly, when OpenSIPS routes calls to a set of destinations, it is able to keep the load status (as number of ongoing calls) of each destination and to choose to route to the less loaded destination (at that moment). OpenSIPS is aware of the capacity of each destination - it is preconfigured with the maximum load accepted by the destinations. This is an idea Load-Balancer to front your Asterisk cluster. A nice tutorial about how to do LB for your Asterisk is available: http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing Regards, Bogdan The best server ever, Great Job! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
2009/3/8 Marco marcota...@libero.it Hi List, I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years on my lab test setup and I appreciate it. Moreover the global quantity of fax handled by this setup is not very high. I'll be involved in a more complex system for a customer and I would like to ask to All of you if you have experiences and/or statistical results on faxing success and failure rate. The system I have to deploy will operate in the following context: - It will be interfaced to an E1 PRI - It will be able to send and receive faxes (by e-mail and/or virtual printers) - It will be able to send faxes from a normal fax machine. The system will be placed on the same building, i.e. only private ethernet trunks. I'm thinking to this type of solution: - Patton external unit for E1 - Asterisk 1.4 + IAXModem + Hylafax - An external ATA for the fax machine but I'm open to any other possible solution (I'm thinking to have a demodulation on Patton and talk T38 with Asterisk 1.6). The fax volume will be high because actually the customer has a ZFax software system with 12 fax-modem installed (that will be replaced by the system). I know that this was already asked in this list in the past, but I would like to know if someone has experience on this and could share their opinion, tricks and/or statistical results on failure/success rate when faxing. I think that this could be useful to other people have to realize a system like that one depicted. Thank you in advance. Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com I had a very good success rate with multitech modems connected to fxs card and calling out with T0 (bri) I had 3 modems in total, - 1 receiveing 2 sending, they were heavily loaded and i had very low failure rate 20 000 faxes and only 3 failed (!) from 3 only one due to incompatibility (brother model know issue) and 2 bad line. I dont inclide here wrong numbers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Meetme Question
2009/3/8 Sven Geggus use...@fuchsschwanzdomain.de Gavin Henry gavin.he...@gmail.com wrote: Just transfer them to your meetme extension after you've called them. Hm, how would I do this? Until now call switching usually ended for me when the call has been established. I'm using a SIP phone connected to an asterisk box which is connected to the world via various ways (ISDN, SIP, IAX2). So what would I do on the my SIP phone after the call has been established and what needs to be changed in the dialplan to actually reconnect the current call to the MeetMe Conference then? Sven You need to transfer option enabled in dial() (tT) CLI core show application Dial And you need to press a transfer button ;) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about MeetMe performance.
2009/3/6 BERGANZ François franc...@acropolistelecom.net hello, I will do a server to do a lots of conferences (MeetMe). I want to know that if I dont use a digum card, the limit of simultaneous calls is harder without a card than with a card ?if, yes, how harder is the limit? thank you Maybe upper not harder ? And what card ? (model) If you add just a source of timing your quality would be better not the limit, there is a transcoding card sold by digium, and i dont know the possibility of htat card, ask digium. Anyway the meetme is all about transcoding. Less transcoding = more conferences. Do your test's! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] after install the zaptel but the rtp failed
type in cli Core show application meetme and read how to use it MeetMe([confno][,[options][,pin]]): Enters the user into a specified MeetMe conference. exten = 4105,n,meetme(99008664105|Ap) So what conf number do you join here ? 99008664105 do you have a conf with that number ? I have compare my two different manchines,(one work OK,and another is failed): when use zap show channels to see the channels status: Chan Extension Context Language MOH Interpret pseudodefaultdefault then i dial the 4105 and channels show Chan Extension Context Language MOH Interpret pseudodefaultdefault pseudodefaultdefault then i hangup,but the channels still have two pseudo: Chan Extension Context Language MOH Interpret pseudodefaultdefault pseudodefaultdefault then i try again,the Meetme didn't ctreat room anymore. and i found a strange thing : after i install the zaptel ,my asterisk didn't play any voice. i use the Playback(Nomoney): Executing [4...@4105:1] Answer(SIP/22238-08211340, ) in new stack -- Executing [4...@4105:2] Playback(SIP/22238-08211340, NoMoney) in new stack -- SIP/22238-08211340 Playing 'NoMoney' (language 'en') It show well but no voice!! Is it wrong in my system? thanks 2009-03-05 -- 邱磊 -- *发件人：* Grygoriy Dobrovolskyy *发送时间：* 2009-03-04 16:30:06 *收件人：* Asterisk Users Mailing List - Non-Commercial Discussion *抄送：* *主题：* Re: [asterisk-users] after install the zaptel but the rtp failed 2009/3/4 邱磊 qiulei...@163.com hi Grygoriy : appreciate your reply , that's my cli command: CLI zap show status Description Alarms IRQbpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 Is't all right? forward your echo . thanks Yes normally you should have meetme working. Paste your extensions.conf here (only the context with the conference) Also the config of the sip peer who is trying to join the conference and more cli output during that join. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] after install the zaptel but the rtp failed
2009/3/4 邱磊 qiulei...@163.com hi Grygoriy : appreciate your reply , that's my cli command: CLI zap show status Description Alarms IRQbpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 Is't all right? forward your echo . thanks Yes normally you should have meetme working. Paste your extensions.conf here (only the context with the conference) Also the config of the sip peer who is trying to join the conference and more cli output during that join. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] after install the zaptel but the rtp failed
2009/3/3 邱磊 qiulei...@163.com hi everyone: now ,i have a strange situation: I want to make a meetme conference and install the zaptel1.4* in my asterisk. every things seem well but it did't work normally. I use the Playback app for test .It didn't reply any voice.I tried in another asterisk server the playback app work well. i don't know why ,any some guys can give me some help? PS: my meetme app also didn't work normally: the asterisk log show: Executing [4...@4105:2] MeetMe(SIP/22479-08203390, 4105|Ap) in new stack without create 4105 room!! but i have config the meetme.conf [rooms] conf =4105; can some guys give me help!! thanks a lot 2009-03-03 -- Check if your ztdummy is loaded zap show status in cli, or that you have a proper source of timing. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Switch Options for a service provider
2009/2/27 Alistair Cunningham acunning...@integrics.com Ignacio, Our Enswitch product matches all these requirements; indeed it goes well beyond them: - We scale far beyond 3000-4000 concurrent calls. We'd consider such a system medium sized. At this size the system is fully failover/redundant, and we have solved the telephony problems of queues, conferences, transfers, etc, with calls on multiple machines. - Full integrated prepaid/postpaid billing and invoicing. - Full real-time reports via web, SOAP API, and direct MySQL. - Full reseller platform with resellers able to set their own pricing, and able to rebrand the product as their own. - Full LCR and carrier failover. - You have root access to the machines and MySQL database. - Full hosted PBX and ITSP features for your customers which they can administer themselves using the web and/or SOAP API. It's in production today from a few hundred users on a single machine to over 150,000 users on large clusters. For more details, please see the links at the bottom of: http://integrics.com/products/enswitch/ In particular, I suggest reading the feature list link. Please do contact me off list for pricing and to arrange a demo installation. Alistair Cunningham +1 888 468 3111 +44 20 799 39 799 http://integrics.com/ Ignacio Ortega A. wrote: Hi, I have a growing voip business Im i looking a solution that can handle at least 3000-4000 concurrent calls with great performance. Also with a billing platform, reports, reseller platform, LCR, call routing,real time reports, SQL dababase access real time Load Reports. I would consider kolmisoft as for payable solution, they are just more advanced then this system. We are using it here. And it can scale from 1 to n system. Also a opensips +cdrtool + mediaproxy are totally free but need a deep understanding of what you are doing ;) But there is hope! i have managed to build a reliable opensips proxy for 20k call's a day from scratch. I think with couple experienced programmers you can do a lot more. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No rtp activity
2009/3/1 michel freiha mich...@gmail.com Dear David, I'm using G729 pass though mode...No transcoding is used here Regarding concurrent calls, I have 3 asterisk servers working in load balancing mode...The issue that the same problem appear on 3 asterisk...each asterisk handle around 150 calls... I'll use tcpdump next time I face such issue Regards On Sat, Feb 28, 2009 at 7:21 PM, michel freiha mich...@gmail.com wrote: Hi all I'm using asterisk for making PSTN calls from extensions registered on OpenSIPS...In peak hours ,number of calls Increase dramatically to a non logic number..When checking the calls using asterisk CLI I saw a lot of calls in ringing status and after 300s(rtphold timeout), asterisk release all calls...I checked the log file and found.. [Feb 28 11:34:14] NOTICE[19197] chan_sip.c: Disconnecting call 'SIP/netcafe2-b7da99b8' for lack of RTP activity in 301 seconds After that the log show: [Feb 28 11:41:12] WARNING[19197] chan_sip.c: Remote host can't match request CANCEL to call '669b27bb46ca01dc42b526adf...@asterisk_ip_address'. Giving up. Did someone faced this issue before? Thanks for help Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Check your opensips config, besides network check, you need to be sure that your sip messages are not redirected to wrong servers, sometimes, if it the case some call legs are messed up, and wrong cancels go to wrong servers... I saw that with bad NAT config on opensips it was passible. Good Day ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] building a phone
2009/2/27 Wilton Helm wh...@compuserve.com I assume that the relevant application requires some non-trivial CPU power. I would exclude e.g. a 486-based systems. I'm not sure that's the case. The industry has gone in the direction of throwing lots of silicon at a problem, often as an excuse for poorly written code, sometimes in an interpreted language. There are a number of high integration CPUs out there that I suspect could do this sort of thing. I develop device controllers for a variety of industry needs. They tend to have Ethernet, RS-232, sometimes 1 Mb/s synchronous communication. G711, quarter VGA color LCD with touchscreen and control loops running at about a 1 ms rate. The entire code takes less than 256K in C. My choice of processor is the DStni Ex (made by Lantronix and sold by Grid Connect) which is a high integration, high speed 186 core with two 10/100 Ethernet Ports and 256K of RAM on it in addition to the usual assortment of other stuff. The above required platform adds three support chips (one being the LCD controller). The CPU can run over 100 MHz. Memory accesses take one clock and typical instructions take two or three. Cost is in the $10 to $20 range for the chip and power consumption is around 1 W (the LCD backlight takes more than that!) I'm sure there are several other comparable platforms out there, such as by Digi International. The Geode is a good candidate as are some VIA chips, if one wants to use protected mode x86. The biggest thing for this is don't even consider Intel. For most of their life they have not provided cutting edge solutions for embedded use. Most of their stuff consumes too much power. And most importantly, they are targeting the very volatile, short lived PC market. By the time you get an embedded design up and running and reach market penetration, you won't be able to buy the chip any more. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I wonder what kind of hardware snom use, they got linux, they got openvpn. I would be nice to have that, and yes i want a gui, maybe not embedded to reduce load, but something like an external config generator software would be nice. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Internet connectivity
2009/2/25 Klaus Darilion klaus.mailingli...@pernau.at Hi! I have a setup with Asterisk in front of a PBX connected with ISDN to the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing ENUM for outgoing calls and allows incoming calls per SIP. Recently the IP connectivity for this location was down the whole telephony was down too - not even incoming calls did work. This is really strange as incoming calls from PSTN are routed directly to the PBX without any IP needed, ISDN to ISDN. Once the IP connectivity was reestablished everything worked fine again. So I wonder what could be the reason that Asterisk blocked all the telephony. thanks klaus Asterisk is using dns resolution, when he is unable to reach dns server * freezes, it's a know issue, this can be avoided if you point asterisk to your local dns service (inside private network) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI pdf book
Big companies, especially those with major computing systems use paid software because they want a vendor they can hold responsible for it. As for OSS and FOSS, it is majorly used by the sort of businesses and individuals who call me (and other IT pros) up and talk the talk, but they don't have a 2 dimes to rub together. This problem is only going to get worse as the so-called 'recession' bites... fellow I.T. professionals - get used to your clients trying to weasel free service out of you. Everything I am hearing from fellow I.T. people is that there is no shortage of 'work' but a lot of clients are resisting paying. Well it is possible to be responsible for the opensource software also. When you have a support package you dont really care if it is a 'open' or 'closed' Look at fonality, ok their soft is not 100% open source but if you take the community edition you still able to subscribe to support, and they WILL take the responsibility to repair you system in case of disaster. Many others in this case. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please help test the gender detection moduleat 575-613-4392
2009/2/18 Asterisk Asterisk nt_aster...@yahoo.com Thanks for the feedback. I did some research and it looks like you were calling over international lines. It also appears that there was high than average static on the line, which is not normal for my system. It's true that I threw my recordings together quickly and the beep was supposed to be funny - it was actually me saying beep. However, the static and noise you received was probably not from my system. Nonetheless, I am working on improving the results of detection and will have a new release today or tomorrow. I'll post it up on the test systems for people to test and build additional data for refinement. Most importantly, I'll be adding a background noise filter and fine tuning the male/female results. After I get the gender detection done, I'll also be adding age range detection. Justin It was so fun :) I have noticed a bad quality audio, not choppy, but hears like very compressed. Calling from France ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Network architecture
I think in this case when 5k call are involved i think all the difficulty of the project is to split the load on different parts of the system. In my case i would do it like that: Phones ---Opensips (Double server with heartbeat and in different places) | | ..asterisk 1-n (mainly for voicemail) Opensips should easily handle you registrations and calls, you just need the LCR module for outgoing and dbalias for incoming. If you need the 100% exact and accurate CDR you should consider mediaproxy, the advantage of mediaproxy is that it is capable if detecting if there is no more udp traffic and send BYE packet, the disadvantage is considerable, the traffic is going through your system, many mediaproxy servers are advised in this case, also they need to be installed in strategic points (close to clients). I would let asterisk to do it's part of voicemail server only. Opensips is a great tool, look at their site, also look at cdrtool and mediaproxy. I hope my post helped. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Network architecture
2009/2/17 Danny Nicholas da...@debsinc.com Just a laypersons opinion – I'm sure others here have better answers or justifications. 1. no (at least not realistically, mathematically there are some) 2. perhaps – bandwidth would be your primary concern since 5K calls would take 150 M of bandwidth 3. IMO it would be better to divide the load, but this depends on the hardware you are using. I would recommend opensips with cdrtool and mediaproxy all load balanced with heartbeat or dns. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Gmail is broken (was: Re: WiFi SIP phone w/VPN?)
2009/2/13 Philipp Kempgen philipp.kemp...@amooma.de Benny Amorsen schrieb: Top posting is annoying. Gmail is broken; maybe I should just killfile @gmail.com. Emails sent through Gmail's *web interface* are broken. :-) Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry about off-topic, but can you advise the mail client who is able to organise the web mailing list topic as web interface does ? (i mean by blocks/topics) I wold be glad to use something else with the same usability, but dont see any alternative. Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing with asterisk
2009/2/16 Fabio Mosti fmo...@gmail.com Hi All, I need to setup asterisk to receive fax. I'm try Spandsp (opensource) and Attrafax (commercial) both on asterisk 1.4.23) but the results are disappointing. with spandsp many times the fax arrives cut. with Attrafax i have some problem. Anyone have any idea or solution (Opensource or commercial) to suggest me ? Best Regards Try hylafax with IAXmodem. The best results i had it the multitech modems directly connected to FXS PCI card, you have a nice web interface if you wish also (avantfax) You can find some nice install scripts at the elastix forums. Have a nice day ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed
2009/2/13 Tzafrir Cohen tzafrir.co...@xorcom.com On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote: I've been involved with getting better data for running Asterisk on the Amazon EC2 cloud computing system. Here are some calculations I've made on costs based on current published prices on Amazon's system. Feel free to tell me that I'm wrong with these calculations - but be specific if you find any problems, as I suspect others may glom onto these figures as gospel and I'd hate to have the wrong data in there. http://www.loligo.com/asterisk/misc/amazon-ec2.xls The net of my calculations is that a small instance of 20 users in a standard office environment would cost about $75 per month, which when compared to running a server in-house works out to be (raw cost, not including admin time and not discounting out-of-office bandwidth) only $38.56 more. Very interesting. For 20$ or slightly more you can rent a Xen or OpenVZ virtual host which will probably do as well. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users And in France it is possible to have a dedicated server with 100 mbit /160 gb hdd 1.6 Ghz for 19€ and unlimited bandwith, and it is real unlimited. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing with asterisk
2009/2/16 Michael mich...@networkstuff.co.nz Anyone have any idea or solution (Opensource or commercial) to suggest me ? Best Regards Try hylafax with IAXmodem. The best results i had it the multitech modems directly connected to FXS PCI card, you have a nice web interface if you wish also (avantfax) You can find some nice install scripts at the elastix forums. Best results are with Hylafax and Multitech serial modems connected directly to the PSTN. Well you dont need asterisk then. U think it is nice to have some cdr's for the incoming faxes. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing with asterisk
2009/2/16 Grygoriy Dobrovolskyy megaho...@gmail.com 2009/2/16 Michael mich...@networkstuff.co.nz Anyone have any idea or solution (Opensource or commercial) to suggest me ? Best Regards Try hylafax with IAXmodem. The best results i had it the multitech modems directly connected to FXS PCI card, you have a nice web interface if you wish also (avantfax) You can find some nice install scripts at the elastix forums. Best results are with Hylafax and Multitech serial modems connected directly to the PSTN. Well you dont need asterisk then. U think it is nice to have some cdr's for the incoming faxes. misstype I think ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed
2009/2/16 SIP s...@arcdiv.com Grygoriy Dobrovolskyy wrote: 2009/2/13 Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote: I've been involved with getting better data for running Asterisk on the Amazon EC2 cloud computing system. Here are some calculations I've made on costs based on current published prices on Amazon's system. Feel free to tell me that I'm wrong with these calculations - but be specific if you find any problems, as I suspect others may glom onto these figures as gospel and I'd hate to have the wrong data in there. http://www.loligo.com/asterisk/misc/amazon-ec2.xls The net of my calculations is that a small instance of 20 users in a standard office environment would cost about $75 per month, which when compared to running a server in-house works out to be (raw cost, not including admin time and not discounting out-of-office bandwidth) only $38.56 more. Very interesting. For 20$ or slightly more you can rent a Xen or OpenVZ virtual host which will probably do as well. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com mailto:jabber%3atzafrir.co...@xorcom.comjabber%253atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir http://iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users And in France it is possible to have a dedicated server with 100 mbit /160 gb hdd 1.6 Ghz for 19€ and unlimited bandwith, and it is real unlimited. Seriously? Where? Sign me up! N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It is not a biz list but i am not owning the company, it is ovh.com click to kimsufior go directly to http://www.kimsufi.com/ oh it is 19.99 but not a 160 gb but a 250 gb and still unlimited. have fun. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WiFi SIP phone w/VPN?
The desktop versions of snom support Openvpn, i am not sure about M3 (dect). Take a tour to their site. 2009/2/12 Frank Bulk - iName.com frnk...@iname.com Not in the form factor that you would expect. Can I ask why? Most modern VoFi phones support WPA2. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Wednesday, February 11, 2009 5:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] WiFi SIP phone w/VPN? Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN abilities? Failing that, a WiFi phone that runs Linux? I already know one phone that does meet my requirements -- the iPhone. The new software comes with a Cisco VPN client, and a SIP client can be had from third-party vendors for jailbroken phones. And, while I'm not averse to the idea, a) it ain't cheap, and b) it's a bit hack. I've googled my heart out, but haven't found anything else that (I'm sure) meets all three requirements. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security issue
Hello, if you dont know iptables that much, and would like to see more user friendly configuration method, i suggest you to use Shorewall, which is very flexible, has some clear logs, and generates same iptable rules behind. 2009/2/8 David fire ddf...@gmail.com denay permit are in sip.conf and iax.conf David 2009/2/7 oumar ndiaye ondi...@antg.com David, Thanks in advance. Where do I change the user/peers definition? Is it in the firewall of the OS? In that case that won't work because the server host other services such as ssh http that are open to any IP as long as the user has the correct credentials. Doesn't asterisk itself has built in security filters? If the only choice is to do in the OS's firewall, then I will need to include the port numbers of SIP, IAX in my firewall rules. In this case, which ports should I block to keep unwanted SIP/IAX connections from specific IP's. Thanks. On Sat, Feb 7, 2009 at 9:29 AM, David fire ddf...@gmail.com wrote: you have many options but you should use it together. firewall in the user/peers definitions add host=ip and/or deny=0.0.0.0/0.0.0.0 permit=ip/mask change the ip of your server. use something like ossec to avoid force brute. David 2009/2/6 oumar ndiaye ond4...@gmail.com Is there a way to restrict connection to my asterisk server to users based on their IP addresses, and not just password. I have some hackers who connect to my server to make illegitimate solicitation calls to people. I had to shutdown the server for now until I find a solution. ANY HELP? Thanks. ond ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Oumar Ndiaye CTO ANTG Telecom www.antg.com ondi...@antg.com ondi...@alum.mit.edu ond4...@gmail.com Tel: +1-919-291-8742 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GTalk Channel
How many ports have you forwarded for the * ? (in rtp.conf) If a limited amount (50-100), try to forward more. 2009/1/29 GNUbie gnu...@gmail.com Hello all, In addition to my previous e-mail, below is a more verbosed messages I got on my Asterisk shell when calling from another GTalk User ID to the Asterisk-1.4.21.2 box: pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=49 from=cal...@gmail.com/Talk.v1041B79926Bsession type=initiate id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;description xml:lang=en xmlns=http://www.google.com/session/phone;payload-type id=103 name=ISAC clockrate=16000/payload-type id=97 name=IPCMWB clockrate=16000 bitrate=8/payload-type id=99 name=speex clockrate=16000 bitrate=22000/payload-type id=4 name=G723 clockrate=8000 bitrate=6300/payload-type id=98 name=speex clockrate=8000 bitrate=11000/payload-type id=100 name=EG711U clockrate=8000 bitrate=64000/payload-type id=101 name=EG711A clockrate=8000 bitrate=64000/payload-type id=0 name=PCMU clockrate=8000 bitrate=64000/payload-type id=8 name=PCMA clockrate=8000 bitrate=64000/payload-type id=13 name=CN clockrate=8000/payload-type id=102 name=iLBC clockrate= JABBER: gtalk INCOMING: 8000 bitrate=13300/payload-type id=106 name=telephone-event clockrate=8000//descriptiontransport xmlns=http://www.google.com/transport/p2p//session/iq [Jan 29 11:18:24] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions? [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: Unable to allocate gtalk structure! pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=51 from=cal...@gmail.com/Talk.v1041B79926Bsession type=transport-info id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;transport xmlns=http://www.google.com/transport/p2p;candidate name=rtp address=10.20.1.151 port=1587 preference=1 username=RrBBqm7MeJW2zTgi protocol=udp generation=0 password=OjLNI9dyFLqqBi/Y type=local network=0//transport/session/iq pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=52 from=cal...@gmail.com/Talk.v1041B79926Bsession type=transport-info id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;transport xmlns=http://www.google.com/transport/p2p;candidate name=rtp address=219.74.65.168 port=1588 preference=0.9 username=sHhE4y2GwRBmLQUB protocol=udp generation=0 password=BYAvdVRiU94RVOJW type=stun network=0//transport/session/iq pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=54 from=cal...@gmail.com/Talk.v1041B79926Bsession type=terminate id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session//iq [Jan 29 11:18:40] NOTICE[1303]: chan_gtalk.c:783 gtalk_hangup_farend: Whoa, didn't find call! JABBER: gtalk OUTGOING: iq type='result' from='ast...@gmail.com/asteriskE2D976CC' to='cal...@gmail.com/Talk.v1041B79926B' id='54'/ JABBER: gtalk INCOMING: pbx*CLI JABBER: gtalk INCOMING: presence from=cal...@gmail.com/Talk.v1041B79926B to=ast...@gmail.compriority24/priorityc node=http://www.google.com/xmpp/client/caps; ver=1.0.0.104 ext=share-v1 voice-v1 xmlns=http://jabber.org/protocol/caps/x stamp=20090129T03:17:52 xmlns=jabber:x:delay/status/x xmlns=vcard-temp:x:updatephoto8939f8f8ed0a9cd794e9e3c7065c2cc80fa9dbf0/photo/x/presence pbx*CLI JABBER: gtalk INCOMING: presence from=cal...@gmail.com/Talk.v1041B79926B type=unavailable to=ast...@gmail.com/ pbx*CLI Thank you in advance. Regards, Marvin On Thu, Jan 29, 2009 at 10:47 AM, GNUbie gnu...@gmail.com wrote: Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions? [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: Unable to allocate gtalk structure! [Jan 29 10:38:06] NOTICE[1303]: chan_gtalk.c:783 gtalk_hangup_farend: Whoa, didn't find call! Any idea? Thank you in advance. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] GTalk Channel
And what ports have you set in rtp.conf ? i suppose they are the same ? Try to search if there are no spercial jabber ports to open. 2009/1/29 GNUbie gnu...@gmail.com Hello Grygoriy, I am forwarding UDP ports from 1 to 10100. That only means that I am forwarding 101 ports. Please take note also that when I tried calling the GTalk ID, the Asterisk box was idle or there was no any other on-going calls. Regards, GNUbie On Thu, Jan 29, 2009 at 4:15 PM, Grygoriy Dobrovolskyy megaho...@gmail.com wrote: How many ports have you forwarded for the * ? (in rtp.conf) If a limited amount (50-100), try to forward more. 2009/1/29 GNUbie gnu...@gmail.com Hello all, In addition to my previous e-mail, below is a more verbosed messages I got on my Asterisk shell when calling from another GTalk User ID to the Asterisk-1.4.21.2 box: pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=49 from=cal...@gmail.com/Talk.v1041B79926Bsession type=initiate id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;description xml:lang=en xmlns=http://www.google.com/session/phone;payload-type id=103 name=ISAC clockrate=16000/payload-type id=97 name=IPCMWB clockrate=16000 bitrate=8/payload-type id=99 name=speex clockrate=16000 bitrate=22000/payload-type id=4 name=G723 clockrate=8000 bitrate=6300/payload-type id=98 name=speex clockrate=8000 bitrate=11000/payload-type id=100 name=EG711U clockrate=8000 bitrate=64000/payload-type id=101 name=EG711A clockrate=8000 bitrate=64000/payload-type id=0 name=PCMU clockrate=8000 bitrate=64000/payload-type id=8 name=PCMA clockrate=8000 bitrate=64000/payload-type id=13 name=CN clockrate=8000/payload-type id=102 name=iLBC clockrate= JABBER: gtalk INCOMING: 8000 bitrate=13300/payload-type id=106 name=telephone-event clockrate=8000//descriptiontransport xmlns=http://www.google.com/transport/p2p//session/iq [Jan 29 11:18:24] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions? [Jan 29 11:18:24] WARNING[1303]: chan_gtalk.c:1175 gtalk_newcall: Unable to allocate gtalk structure! pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=51 from=cal...@gmail.com/Talk.v1041B79926Bsession type=transport-info id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;transport xmlns=http://www.google.com/transport/p2p;candidate name=rtp address=10.20.1.151 port=1587 preference=1 username=RrBBqm7MeJW2zTgi protocol=udp generation=0 password=OjLNI9dyFLqqBi/Y type=local network=0//transport/session/iq pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=52 from=cal...@gmail.com/Talk.v1041B79926Bsession type=transport-info id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session;transport xmlns=http://www.google.com/transport/p2p;candidate name=rtp address=219.74.65.168 port=1588 preference=0.9 username=sHhE4y2GwRBmLQUB protocol=udp generation=0 password=BYAvdVRiU94RVOJW type=stun network=0//transport/session/iq pbx*CLI JABBER: gtalk INCOMING: iq to=ast...@gmail.com/asteriskE2D976CC type=set id=54 from=cal...@gmail.com/Talk.v1041B79926Bsession type=terminate id=3756468934 initiator=cal...@gmail.com/Talk.v1041B79926B xmlns=http://www.google.com/session//iq [Jan 29 11:18:40] NOTICE[1303]: chan_gtalk.c:783 gtalk_hangup_farend: Whoa, didn't find call! JABBER: gtalk OUTGOING: iq type='result' from='ast...@gmail.com/asteriskE2D976CC' to='cal...@gmail.com/Talk.v1041B79926B' id='54'/ JABBER: gtalk INCOMING: pbx*CLI JABBER: gtalk INCOMING: presence from=cal...@gmail.com/Talk.v1041B79926B to=ast...@gmail.compriority24/priorityc node=http://www.google.com/xmpp/client/caps; ver=1.0.0.104 ext=share-v1 voice-v1 xmlns=http://jabber.org/protocol/caps/x stamp=20090129T03:17:52 xmlns=jabber:x:delay/status/x xmlns=vcard-temp:x:updatephoto8939f8f8ed0a9cd794e9e3c7065c2cc80fa9dbf0/photo/x/presence pbx*CLI JABBER: gtalk INCOMING: presence from=cal...@gmail.com/Talk.v1041B79926B type=unavailable to=ast...@gmail.com/ pbx*CLI Thank you in advance. Regards, Marvin On Thu, Jan 29, 2009 at 10:47 AM, GNUbie gnu...@gmail.com wrote: Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]: chan_gtalk.c:971 gtalk_alloc: Out of RTP sessions
Re: [asterisk-users] Don't get asterisk to run behind NAT router
You enabled port forwarding, but have you actually forwarded any ports ? Defaults are tcp 5060 udp 1-2 2009/1/29 Tamer Higazi th9...@googlemail.com Hi people! I am not getting smart getting asterisk 1.6 behind a NAT to run. 1. I enabled IP forwarding on debian linux 2. told asterisk in general that he is behind NAT and mentioned him his external static IP Adress as well his domain in the outside world. If a client who is connected with a DSL modem calls me, a grandstream module in the LAN behind the router, in the same network asterisk is running at, takes the call. but we can't hear / talk with each other. Ay ideas to get this thing solved?! My general section in sip.conf: [general] port=5060 bindaddr=0.0.0.0 localnet=192.168.1.0/255.255.255.0 externip=85.183.112.3 externhost=voipfax.higazi-it.comhttp://192.168.1.0/255.255.255.0externip=85.183.112.3externhost=voipfax.higazi-it.com allowtransfer=yes qualify=yes nat=yes [2006] type=friend secret=frank host=dynamic context=nurintern nat=no [2007] type=friend secret=jochen host=192.168.1.2 context=nurintern nat=yes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
Paste your register lines (hide pass) 2009/1/29 Imanol Pardavila imanol.pardav...@ibercom.com I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using a sip account (Asterisk 1 acting as a conventional sip user). Thanks Regards Danny Nicholas escribió: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
You are repeating yourself, paste here sip.conf of each server with register lines AND peer configurations. 2009/1/29 Imanol Pardavila imanol.pardav...@ibercom.com Hi, The SIP messages flow is this: ### AAA.BBB.CCC.DDD: Asterisk 1 IP address EEE.FFF.GGG.HHH: Asterisk 2 IP address ### REGISTER sip:ast2.domain.comSIP/2.0 Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;rport From: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com ;tag=as715628d7 To: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:s...@aaa.bbb.ccc.ddd:19646 Event: registration Content-Length: 0 Using latest REGISTER request as basis request Sending to AAA.BBB.CCC.DDD : 19646 (NAT) Transmitting (NAT) to AAA.BBB.CCC.DDD:19646: SIP/2.0 100 Trying Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;received=AAA.BBB.CCC.DDD;rport=19646 From: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com ;tag=as715628d7 To: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:0...@eee.fff.ggg.hhh Content-Length: 0 Transmitting (NAT) to AAA.BBB.CCC.DDD:19646: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AAA.BBB.CCC.DDD:19646;branch=z9hG4bK0939cc12;received=AAA.BBB.CCC.DDD;rport=19646 From: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com ;tag=as715628d7 To: sip:0...@ast2.domain.com sip%3a0...@ast2.domain.com;tag=as5ccb43ac Call-ID: 13106f3936f01d1c103d5f5230278...@aaa.bbb.ccc.ddd CSeq: 133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7294c1d1 ontent-Length: 0D Asterisk 1 sends an REGISTER without credentials, and Asterisk 2 replies with a 401 message (with Digest algorithm, realm and nonce). I want to configure the Asterisk 1 in order to send REGISTER with credentials. Thanks Regards Grygoriy Dobrovolskyy escribió: Paste your register lines (hide pass) 2009/1/29 Imanol Pardavila imanol.pardav...@ibercom.com mailto:imanol.pardav...@ibercom.com I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using a sip account (Asterisk 1 acting as a conventional sip user). Thanks Regards Danny Nicholas escribió: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] asterisk help
Disable the firewall which is enabled by default in centos Run system-config-securitylevel Set both Security Level and SELinux to Disabled and hit OK: http://images.howtoforge.com/images/perfect_server_centos_5.2/big/24.png 2009/1/25 David fire ddf...@gmail.com paste all your sip.conf or attach it. David 2009/1/24 Vinicius Neves vinicius.ne...@live.com hello! i'm new to asterisk. i'm using CentOS 5.2 + ASterisk 1.6 when i finish installing asterisk, i configure sip.conf like: [4455] type=friend username=4455 secret=1234 host=dynamic context=internal [4466] type=friend username=4466 secret=1234 host=dynamic context=internal and extensions.conf like: [internal] exten = 4455,1,Dial(SIP/4455) exten = 4466,1,Dial(SIP/4466) ok. i start asterisk with: #asterisk -cvvv and open a softphone try to connect and nothing! [image: Sad] i try nmap @ port 5060 but it's closed! [image: Sad] what i can do? thx -- Get news, entertainment and everything you care about at Live.com. Check it out! http://www.live.com/getstarted.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Detect
try lspci 2009/1/26 Tzafrir Cohen tzafrir.co...@xorcom.com On Mon, Jan 26, 2009 at 01:45:56PM +0100, Philipp Kempgen wrote: Tzafrir Cohen schrieb: On Mon, Jan 26, 2009 at 05:24:03PM +0530, David @ULC wrote: Which command to run which will auto detect all hardwares present in the system ? Hardware of what type? The OP clearly said *all* hardware. :-) Most of your hardware is something a standard linux distro would handle. * chan_alsa and chan_oss will then use whatever you defined through ALSA/OSS (and usually the distro does that for you) * Likewise chan_consoles sees whatever pulseaudio sees, I believe. * chan_vpb usually detects hardware automatically (in libvpb) * chan_phone: no idea * chan_misdn: should have something similar. I'm not familiar with it. * chan_usbradio: I'm likewise completely unfamiliar with that. So I guess I was close enough. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk freezes with Fixup failed on channel SIP/...MASQ
Copy paste from freeswitch.org Asterisk uses a modular design where a central core loads shared objects to extend the functionality with bits of code known as modules. Modules are used to implement specific protocols such as SIP, add applications such as custom IVRs and tie in other external interfaces such as the Manager Interface. The core of Asterisk is a threading model but a very conservative one. Only origination channels and channels executing an application have threads. The B leg of any call operate only within the same thread as the A leg and when something happens like a call transfer the channel must first be transferred to a threaded mode which often times includes a practice called channel masquerade, a process where all the internals of a channel are torn from one dynamic memory object and placed into another. A practice that was once described in the code comments as being nasty. The same went for the opposite operation the thread was discarded by cloning the channel and letting the original hang-up which also required hacking the cdr structure to avoid seeing it as a new call. One will often see 3 or 4 channels up for a single call during a call transfer because of this. /* XXX This is a seriously wacked out operation. We're essentially putting the guts of the clone channel into the original channel. Start by killing off the original channel's backend. I'm not sure we're going to keep this function, because while the features are nice, the cost is very high in terms of pure nastiness. XXX */ This became the de facto way to pull a channel out of the grips of another thread and the source of many headaches for application developers. This uncertain threading scheme was one of the motivating factors for a rewrite. Asterisk uses linked-lists to manage its open channels. A linked-list is a series of dynamic memory chained together by using a structure that has a pointer to its own type as one of the members allowing you to endlessly chain objects and keep track of them. They are indeed a useful programming practice but when used in a threaded application become very difficult to manage. One must use mutexes, a kind of traffic light for threads to make sure only 1 thread ever has write access to the list or you risk one thread tearing a link out of a list while another is traversing it. This also leads to horrible situations where one thread may be destroying or masquerading a channel while another is accessing it which will result in a Segmentation Fault which is a fatal error in the program and causes it to instantly halt which, of course means in most cases all your calls will be lost. We've all seen the infamous Avoiding initial deadlock message which essentially is an attempt to lock a channel 10 times and if still won't lock, just go ahead and forget about the lock. 2009/1/24 Udo Schacht-Wiegand aster...@wiegand.name On a production system, running 1.4.17 (compiled from bristuff-0.4.0-test6-xr1) we had this strange issue two times in the last weeks: [2009-01-13 13:58:30] WARNING[1213] channel.c: Fixup failed on channel SIP/2332-081d0108MASQ, strange things may happen. [2009-01-13 13:58:30] WARNING[1213] channel.c: Hangup failed! Strange things may happen! [2009-01-13 13:58:30] WARNING[1213] channel.c: Failed to perform masquerade [2009-01-13 13:58:30] WARNING[1213] channel.c: Channel 'SIP/2332-081d0108' may not have been hung up properly and: [2009-01-23 14:27:17] WARNING[21528] channel.c: Fixup failed on channel SIP/2332-083c3778MASQ, strange things may happen. [2009-01-23 14:27:17] WARNING[21528] channel.c: Hangup failed! Strange things may happen! [2009-01-23 14:27:17] WARNING[21528] channel.c: Failed to perform masquerade [2009-01-23 14:27:17] WARNING[21528] channel.c: Channel 'SIP/2332-083c3778' may not have been hung up properly Both times all SIP channels got stuck and the CLI became inresponsive. Calls continued for a while, but new SIP calls could not be established. On the second time this happended, all SIP phones could not subscribe to the Asterisk any longer and a few minutes later the log filled with: [2009-01-23 14:43:21] ERROR[22319] chan_sip.c: Call to peer '2333' rejected due to usage limit of 10 On the CLI one could see, that there were 100s of (rejected) calls to this SIP phones. The phones that show up in the ERROR messages are in a group call made by a Dial(Local/...Local.../Local/...) construct. But other SIP phones were affected as well. It seemed like the whole chan_sip module became stuck. I also could not unload chan_sip.so, but can't remeber the exact error message it gave. The only thing that was left was to restart Asterisk. Can someone give me some clue what the 'Fixup failed ...' and 'masquerade' warnings actually mean? Any help appreciated. Udo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To
Re: [asterisk-users] Root Password not taking
Or boot in single user type passwd and done. 2009/1/22 Jim Dickenson dicken...@cfmc.com What I have done in the past to set the password for root is to boot in rescue mode and edit /etc/shadow setting the password to some know value from another system. -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ -- *From: *David @ULC ucoms2...@gmail.com *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Date: *Thu, 22 Jan 2009 21:22:08 +0530 *To: *asterisk-users@lists.digium.com *Subject: *[asterisk-users] Root Password not taking In one of my center , its not taking root password. Anyways to recover it ? In other terms , I lost the control of server. Any solution or re-installation is the only way left ? I am using CentOS. -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hangup a call manually...
try to know the whole string ? core show channels 2009/1/16 Carlos Chavez cur...@telecomabmex.com I have this call: SIP/protel-525512047 default 90445528885371 1 Ringing AppDial (Outgoing Line) 90445528885371 264:24:2 (None) I cannot use the soft hangup commando from the CLI because I do not know the whole SIP channel string. What other command can I use to terminate this call or to find the complete channel string to put into soft hangup? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error messgae
Here you go http://tinyurl.com/a7tkkz 2009/1/12 chinmay chakraborty chinmay.chakrabo...@gmail.com Hello, I am having problems getting one xlite clients to communicate through asterisk. I am getting an error message: chan_sip.c:15593 handle_request_register: Registration from 'chinmay chakrabortysip:1...@10.44.32.193 sip%3a1...@10.44.32.193' failed for '10.44.32.193' - No matching peer found sip show peers Name/username HostDyn Nat ACL Port Status 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] so plz tell me how to set up xlite with asterisk thanks chinmay c -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to monitor asterisk with SNMP?
Can you show me your script please ? For which version is it ? 2009/1/10 Markus A. Wipfler mar...@infocom.co.ug Another way to monitor this via cacti (for example if you don't have snmp support for asterisk or need to customize what you are graphing) is to create a new data input method in cacti and then use a script to get you the required data. I use a simple perl script that gets my all active zap, iax, sip channels, how many concurrent calls from network A to B, and more... http://www.cacti.net/downloads/docs/html/making_scripts_work_with_cacti.html I would also suggest to run the cacti poller every 1 minute rather than the default 5. -- Markus On Jan 10, 2009, at 11:24 PM, Matt Gibson wrote: http://www.voipphreak.ca/2007/04/16/monitoring-asterisk-14-with-snmp-and-cacti-for-pretty-graphs/ Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com *From:* asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com ] *On Behalf Of *Robert Augustyn *Sent:* Saturday, January 10, 2009 2:45 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] How to monitor asterisk with SNMP? Hi, We have zabbix running and would love to be able to monitor our asterisk box with it. I believe that some sort of SNMP is build in 1.4+ correct? Where do I find more info or a how to on what is supported and how to use it? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to monitor asterisk with SNMP?
I wonder if the same is possible with centreon ?
Someone is using centreon here ?
2009/1/11 Markus A. Wipfler mar...@infocom.co.ug
On Jan 11, 2009, at 2:43 PM, Grygoriy Dobrovolskyy wrote:
Can you show me your script please ?
if for example you had 4 trunks then the below should give you the active
channels for each trunk plus the total, in a cacti understandable output
format.
#!/usr/bin/perl
($trunk1, $trunk2, $trunk3, $trunk4, $total) = (0,0,0,0,0);
@channels = split(/\n/, qx(/usr/bin/sudo /usr/sbin/asterisk -rx 'core show
channels concise'));
foreach $line (@channels) {
$trunk1++ if ($line =~ m/trunk1/);
$trunk2++ if ($line =~ m/trunk2/);
$trunk3++ if ($line =~ m/trunk3/);
$trunk4++ if ($line =~ m/trunk4/)
}
$total = $trunk1 + $trunk2 + $trunk3 + $trunk4;
print trunk1:$trunk1 trunk2:$trunk2 trunk3:$trunk3 trunk4:$trunk4
total:$total;
exit 0
For which version is it ?
am running this with cacti version 0.8.7b
Hope this helps. The cool thing with this is that you can really customize
what you are graphing in cacti.
--
Markus
2009/1/10 Markus A. Wipfler mar...@infocom.co.ug
Another way to monitor this via cacti (for example if you don't have snmp
support for asterisk or need to customize what you are graphing) is to
create a new data input method in cacti and then use a script to get you the
required data. I use a simple perl script that gets my all active zap, iax,
sip channels, how many concurrent calls from network A to B, and more...
http://www.cacti.net/downloads/docs/html/making_scripts_work_with_cacti.html
I would also suggest to run the cacti poller every 1 minute rather than
the default 5.
--
Markus
On Jan 10, 2009, at 11:24 PM, Matt Gibson wrote:
http://www.voipphreak.ca/2007/04/16/monitoring-asterisk-14-with-snmp-and-cacti-for-pretty-graphs/
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
*From:* asterisk-users-boun...@lists.digium.com [
mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com
] *On Behalf Of *Robert Augustyn
*Sent:* Saturday, January 10, 2009 2:45 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] How to monitor asterisk with SNMP?
Hi,
We have zabbix running and would love to be able to monitor our asterisk
box with it.
I believe that some sort of SNMP is build in 1.4+ correct?
Where do I find more info or a how to on what is supported and how to use
it?
Thank you.
___
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Re: [asterisk-users] RTCP SR transmission error, rtcp halted
You should turn rtcp off in the phones settings. 2009/1/12 Rajkumar S rajkum...@gmail.com Hi, While looking for the cause of disturbance in call I found this error coming in console RTCP SR transmission error, rtcp halted Google search only shows some bug reports relating to MOH and Hold. What could cause this message? Could this be a symptom causing call disturbance? Where should I start digging to find out the reason for this error? I am using Asterisk 1.4.19 with zaptel 1.4.9.2 Thanks and regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lock SIP Account after too many failed logins
2009/1/9 Steve Howes st...@geekinter.net On 9 Jan 2009, at 16:36, Klaus Darilion wrote: Hi! I want to detect brute-force password hacking attacks - thus if there are too many failed login attempts for a SIP account I want to lock this account. Does somebody have any ideas how this could be implemented? Bad plan? Could quite easily turn into a DoS. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have the same problem, just look here: Jan 9 15:14:37 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:37 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:37 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:37 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:37 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:37 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:38 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:38 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:38 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:38 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:38 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:38 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:39 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:39 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch Jan 9 15:14:39 NOTICE[338] chan_sip.c: Registration from '3CXPhonesip:SIP/00085d101...@83.167.156.171:5060' failed for '91.171.139.135' - Username/auth name mismatch It's not a bad idea maybe to create something like maxloginattemts=x ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many quad T1 cards
700-800 is the maximum limit without transcoding on very optimized setup. I would call it suicide without a failover solution. Why dont you consider the dns srv for load balancing among 2 servers ? 2009/1/8 Scott Plante spla...@insightsys.com Jerry, back in August you were thinking about putting 4 T1 cards in a single box--did you end up doing that and how did it work out? We're looking at 700-800 lines for an app and are trying to figure out how many machines we'll need. Has anyone else done more than 2 quad T1 cards? -- Scott Plante, CTO Insight Systems, Inc. (+1) 404 873 0058 x104 spla...@insightsys.com http://zyross.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel variable to identify the calling SIP peer
core show function SIPPEER 2009/1/6 Klaus Darilion klaus.mailingli...@pernau.at since 1.4 you can also use setvar=foo=bar in sip.conf when configuring the peer. Then the channel variable foo is automatically set to bar for calls initiated by this peer. regards klaus Philipp Kempgen wrote: Grey Man schrieb: On Wed, Nov 26, 2008 at 11:47 AM, Richard Brady rnbr...@gmail.com wrote: Hi folks I'm not sure what I am missing but I cannot find a predefined channel variable to identify the SIP peer/user which has initiated a call and established the channel. The one option is to extract it from the CHANNEL variable, but that is fraught with difficulties. Is there another variable I don't know about or another way to do this? In 1.2 and 1.4 I don't believe there is any other way. Parsing the username from the channel name is what we ended up having to do! Since 1.6 there is CHANNEL(peername). Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asus Eee PC 900] as replacement for legacy BRI phone
Xorcom had something, usb bri, but it is pricey. If you dont need to change provider and planning to stay with bri, why dont you buy another bri phone ? 2009/1/7 Matthias Apitz g...@unixarea.de Hello, I own one of these netbooks Asus Eee PC 900, mine is running FreeBSD 7.0, and a Linux based cellphone, the OpenMoko Freerunner. Since some time I'm thinking in a replacement of my 'normal' BRI phone at home and the two items mentioned above let me think that the replacement should be UNIX based as well. The legacy BRI phone, around ten years old, even has more or less the same size as the Eee PC 900 :-) Well, what I want to do is install Asterisk in a new diskless Eee PC 900 which comes with a 20 GByte SSD, big enough for that purpose; more info here: http://www.asus.com/products.aspx?l1=24l2=164model=2744modelmenu=1 http://www.unixarea.de/installEeePC.txt as the phone I want to use the normal desktop (KDE 3.5.8) and some SIP client, for example Ekiga, or even a more simple one. Don't know how to replace the voice box the legacy BRI phone has; but this could be managed as well by the provider services; Is there any USB based BRI card which works with Asterisk in FreeBSD (or if not at least in Linux)? Any other hints or comments about my approach? Thanks in advance matthias -- Matthias Apitz Manager Technical Support - OCLC GmbH Gruenwalder Weg 28g - 82041 Oberhaching - Germany t +49-89-61308 351 - f +49-89-61308 399 - m +49-170-4527211 e matthias.ap...@oclc.org - w http://www.oclc.org/ http://www.UnixArea.de/ b http://gurucubano.blogspot.com/ SPAMer of the year: Subject: Alle Software ist Deutsche Sprachen From: -40 % die Neujahrsaktion gabriellekel...@grungecafe.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI got freezed!!
2009/1/7 Max Alex max.aster...@gmail.com Hi, Thanks for your reply Can you suggest me how can we avoid it by doing any configuration changes in asterisk. So the freeze issue may not be occurred again! Please provide me some help!!! Thanks in advance! Thanks, Max Alex Voip Developer On Wed, Jan 7, 2009 at 12:58 PM, Grey Man greymanv...@gmail.com wrote: Doesn't matter if you have set it up or not Asterisk needs DNS. I haven't checked the code but I think it even does reverse lookups on IP addresses. If you haven't got a reliable DNS server available for Asterisk I suspect you're always going to get issues. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can setup a local dns server. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any SIP client for BlackBerry?
2009/1/7 TianLun Song stl...@gmail.com From the product description, i dont think Gizmo5 allows me to register the client with my asterisk. If i am wrong, please let me know On Wed, Jan 7, 2009 at 4:43 PM, Rodolfo Alcazar Portillo rodolfo.alca...@padep.org.bo wrote: Missed the thread, sorry. Gizmo5.com has some blackberry SIP clients. Could be what you want. Greets! Am Mittwoch, den 07.01.2009, 16:07 -0500 schrieb Eric Moniz: TianLun, I should have know you would have wanted a Blackberry SIP client to connect to an Asterisk box. Sorry my bad! I knew there was a reason why I didn't choose Truphone as my SIP client. I have an iPhone and I am currently using Fring which is local client that connects to my Asterisk box nicely, but at this time Fring has no support for the Blackberry OS. This is why I directed you to Truphone. I did search the forums for a Truphone to asterisk hack, but found nothing substantial. Keep an eye on fring.com maybe they will come through. Sorry best of luck! E. On Tue, Jan 6, 2009 at 2:38 PM, TianLun Song stl...@gmail.com wrote: Thank you, This one looks much better. Is it able to register with Asterisk instead of sign up a plan with Truphone? thank you On Tue, Jan 6, 2009 at 2:02 PM, Eric Moniz emoni...@gmail.com wrote: Take a look at TRUPHONE @ truphone.com Eric On Tue, Jan 6, 2009 at 1:33 PM, TianLun Song stl...@gmail.com wrote: Hi You all, Does anyone know any SIP client for BlackBerry? thank you -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: rodolfo.alca...@padep.org.bo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You are doomed with blackberry for now ;( Carriers want clients spend their money thay are pushing their uma. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very
try do add fromdomain=acme.com/sip.acme.com fromhost=acme.com/sip.acme.com 2009/1/6 Frank Bulk frnk...@iname.com I tried that before, but I just tried it again. Unfortunately, the same thing: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 [ACME] host=172.16.10.40 username=username secret=password type=friend Frank *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Allan Dib *Sent:* Monday, January 05, 2009 9:41 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Try it by IP address instead of hostname as reverse DNS may not be resolving. e.g. host=123.123.123.123 On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk frnk...@iname.com wrote: This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which SIP/2.0 401 Unauthorized is issued after the un-authenticated INVITE and SIP/2.0 403 Forbidden after the authenticated INVITE. When I add insecure=very, this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: sip:5551236...@172.16.10.40sip%3a5551236...@172.16.10.40 It isn't very clear (to me) from the success how the insecure=very helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Frank Bulk - iName.com wrote: The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add insecure=very to my Outgoing settings, but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with SIP/2.0 403 Forbidden I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the Incoming Settings panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com sip%3a%2b15552027...@sip.acme.comSIP/2.0 From: sip:5552022...@172.16.10.40 sip%3a5552022...@172.16.10.40 ;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b ba4 To: sip:+15552027...@sip.acme.com sip%3a%2b15552027...@sip.acme.com Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: sip:5552022...@172.16.10.40sip%3a5552022...@172.16.10.40 ;user=phone Privacy: none Remote-Party-ID: sip:5552022...@172.16.10.40sip%3a5552022...@172.16.10.40;user=phone; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: sip:5552022...@172.16.10.40 sip%3a5552022...@172.16.10.40 Authorization: Digest username=username,realm=asterisk,nonce=118af2b0,uri=sip:+15552027020 @ sip.acme.com,response=111e63ec2a1f3ebabefe4f7dae4087a1,algorithm=MD5 Content-Type: application/SDP
Re: [asterisk-users] R2D2 VOIP Kubuntu 8.4 Ekiga, Ekiga.net voice conference
Sometimes it's a problem of the timing, do you have this problem with normal call's ? 2009/1/6 john_re john...@fastmail.us I'm having a problem getting a good clear output sidnal from Ekiga to a VOIP conference call using the Ekiga.net free conference call system. I'm told that each time I speak, my voice is clear intelligible for about .5 - 2 seconds, but then it starts to be garbled, sounding like the sounds R2D2 makes. I've used 2 or three mic/headsets - two plug into my audio I/O sockets on my laptop, one is a USB headset (but I'm not sure I tested the usb headset properly, though a friend with the exact same usb headset, also on KUbuntu 8.4, like myself, doesn't have the problem.) My voice came through clearly in 1 on 1 conversations to a specific person. I'm told the problem lessens when I turn down the volume of my microphone gain, but I can't recall if that always worked, or just sometimes. One key observation: It worked fine when I was alone, but all the times I've had problems there have been others at the same table as myself who were listening to the conference on laptop speakers - I suspect the problem might be feedback from their speakers to my microphone - if so, perhaps I can solve this by ensuring noone else nearby is in the conference outputting through laptop speakers. -- So, I just want to know if what I've described is a known issue, or if this R2D2 sounding problem has never been noticed before. /or if there is a know solution to this type of problem. Thanks :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Deadlock ? I hope i am wrong
I have thousands if this messages in the logs: Dec 4 10:53:43 NOTICE[26310]: app_queue.c:1980 wait_for_answer: No one is answering queue 'COMMERCIAL-WT' (2/0/0) Dec 4 10:53:43 WARNING[5602]: channel.c:889 channel_find_locked: Warning: Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL Dec 4 10:53:43 WARNING[5602]: channel.c:889 channel_find_locked: Warning: Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL Dec 4 10:53:44 WARNING[5602]: channel.c:889 channel_find_locked: Warning: Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL Dec 4 10:53:44 WARNING[5602]: channel.c:889 channel_find_locked: Warning: Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL Can someone tell me to what it is related ? asterisk 1.4 freepbx Thank you Grygoriy Dobrovolskyy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities
It is very simple take openser(opensips/openser/kamalio) the openser community is great, the project have been here and tested for a years in production, used by the biggest companyes (millions!) of users, it's a carrier grade soft ;) in combination of cdrtool + opensips + mediaproxy you can get 100% billing accuracy. 2008/11/28 Yehavi Bourvine [EMAIL PROTECTED] I did a test yesterday and did 1,000 registrations to Asterisk using SIPP. I did the register test since I am using the realtime DB and asterisk does periodic quesries to it for each registered user. Although Asterisk continued to function as usuall, it was in a steady loop querying the DB for the 1,000 users. OK, you convinced me to look at some front end to it. There are mainly three front ends mentioed here: OpenSer, SipExpress and FreeSwitch. Is there some comparison available which will save me from testing all three of them? Is there one which is more used than the others? (so it has more public QA :-) Thanks! __Yehavi: 2008/11/24 Steve Totaro [EMAIL PROTECTED] Fronting with OpenSER or FS, you should have no problems providing you plan to use SIP extensions. What is critical are the max simultaneous trunks you are going to use. I would go TDM although universities have good bandwidth, and SUPERIOR bandwidth between others. I would think a TDM DS3 or two just to be safe. It should be pretty trivial besides gotchas, like cat3 to the rooms, although channel banks may be an even better solution if phones are already in place. Then you just use SIP when needed or wanted, and Asterisk is simple, although more costly. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Fri, Nov 21, 2008 at 6:24 PM, Wilton Helm [EMAIL PROTECTED] wrote: Yet another option is a commercial system with in-house staff. I used to maintain a NEC (NEAX 2400) for many years. I went to factory training and had total responsibility for it. Some manufacturers discourage or prevent this, but others are open to it. There are also 3rd party organizations (such as Source) that can supply parts and even expertise for those going that direction. Whether the result would be higher availability than Asterisk, I don't know. Given I'm both a telco guy and a computer guru (CS degree) I'd probably go the Asterisk route myself, because its open and I would have more control. Wilton and bug fixes than any commercial product sold in the intra-industrial channel ... and they won't charge you a $30,000 license fee for the upgrade. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
2008/11/21 Yehavi Bourvine [EMAIL PROTECTED] Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello very interesting project you have, however asterisk is not a registry server, i suggest that you use opensips/opense/kamalio for your registrar, from where you dispatch to you asterisk servers, inside a good environment with a controlled network and nice tagged voip flow you could acheve a good results. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SVN - DIGIUM
server problem's 2008/11/21 Luis Morales [EMAIL PROTECTED] Does any know what happens with svn repository on svn.digium.com ? -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
2008/11/20 Nitzan Kon [EMAIL PROTECTED] Hello! We're looking for a solution to reliably load balance our Asterisk boxes. So far we've been using a hodge-podge of directing different services to different boxes/IPs, but eventually I'd like to consolidate things so we can present a single IP address to the outside world. My question is - how do we go about doing that? I've read a lot of things like load-balancing via DUNDi or OpenSER, but it seems to me like these approaches just add to the list of possible failures. In other words I'd like to avoid software solutions. Is it possible to just put Asterisk behind a load balancer? I imagine most of them are optimized for web traffic rather than UDP voice packets. Does that matter? Would any load balancer do - or only specific models will work? my guess is any model will work, but some of them may not be able to handle the load. Any recommended models? I know there are some fancy LBs out there that can actually load balance based on the SIP session rather than something like IP, but I'm afraid to even look at the price tag. I'm more than fine with balancing by user IP address instead - if that works. :) Would appreciate any comments or ideas. Thanks! -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the failover + watchdog to ensure if opensips/kamalio/openser crashes a nice failover reboot, it is working stable here (dispatching to 10 servers + owners DID dispatch to their respective servers) join #opensips on freenode if you need more info. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
2. Overkill to install and maintain (if we can get a simpler solution) I am not agreed on point 2: If I understood how to install opensips + heartbeat WITHOUT knowing any program (opensips ? heartbear ?) or programming language(hell yes!) in a week ( just knew what's invite and bye ;) a more aware IT professional could do it in 2 days ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend Wireless IP Phone
Use snom M3 Siemens got some problems. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tribox
2008/10/6 Tarek Sawah [EMAIL PROTECTED] i haven't facedthse tpe of problems you mentioned with mysql.. but there is one thing that you need to edit the sip.conf iax.conf or you can use the sample ones in the samples folder.. other than that.. i've been with trixbox for over three years now.. it has problems with it comes to Queues and call center services.. i've been struggling with it for months now .. plus as per an earlier post i had here i'm having problems convincing trixbox to accept my dia plans on two of my three servers.. while i tred installing elastix more than 10 times on different machins.. i don't have those problems.. besides!!! on trixbox you need to add th ip of the freepbx mirrors to upgrade your modules.. and you have to manipulate your php files to be able to upgrade your box from the website.. Is it trixbox pro or free version ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
2008/10/5 Andrew Kohlsmith (lists) [EMAIL PROTECTED] On October 3, 2008 04:15:26 pm Tariq .. wrote: it is FRING i'm sorry for the mistype... www.fring.com I just downloaded it for the iphone... it's pretty cheap looking, crashes occasionally and appears to force all audio through their server, but I have to say that yes, it does have potential. Thanks for the pointer. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Try Siphon for iphone, but you need to jailbreak it ;( ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
I have tryed skip2pbx 580€ yeastar 60 €, the quality is the way behind of a good sip provider, thay are simply not suitable for business, i hope it would not be the case of asterisk addon. Also i wonder if skype auto relay will be disabled (bandwith), wait and see... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
2008/9/26 randulo [EMAIL PROTECTED] Get Olle to call in for once in his life! Mark did say IM and video, IM first. It's all gonna happen. (just not right away) http://lists.digium.com/mailman/listinfo/asterisk-users Video ? that could be really nice but limited to pc/macasteriskwhatever. There are tonns of 3G phones on the market, so why not to adapt software fot the videocalls over wifi ? such a client is my dream for about a year, and i dont care it it would be a skype or else. A new product for that purpose is not a solution, but adapting software to existing 3G phones will open a HUGE market recently created and closed for 3G operators w/licence. Any suggestions ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
2008/9/26 Kevin P. Fleming [EMAIL PROTECTED] Brian J. Murrell wrote: And so will this channel driver also allow Skype to use my resources (CPU, bandwidth -- i.e. Internet for which many have usage caps, etc.) the way the Skype client does? The Skype engine in Skype For Asterisk does not currently have 'relay' support, so it does not route calls or media any calls that it is not involved in. However, this will be present in the production release of the product, but when it appears we will also document its behavior and the configuration options that can be used to control it. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Will it be packed into the base asterisk package, or to asterisk-addons? or into some third party ? Would it be possible to buy some comminication licences use them while disabling the 'relay' function ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pressing 0 to get an external line
Yo can do it with Playtones(!440) !440 is for france seach yours in
indications.confhere is the example script from asterisk-france, the guy had
the exact same problem
[Appel_Sortant_Isdn]
exten = _0,1,Set(Flag_Playtone = 0)
exten = _0,n,Playtones(!440)
exten = _0,n(Continue),Read(Digits,,1,,,3)
exten = _0,n,GotoIf($[${LEN(${Digits})} != 0]?:Suite)
exten = _0,n,GotoIf($[${Flag_Playtone} = 0 ]?Va_Indexer)
exten = _0,n,Set(Flag_Playtone = 1)
exten = _0,n,StopPlaytones
exten = _0,n(Va_Indexer),Set(Call_Number=$[${Call_Number}${Digits}])
exten = _0,n,Goto(Continue)
exten = _0,n(Suite),Answer
exten = _0,n,Set(CDR(userfield)=${Call_Number})
exten = _0,n,Dial(${Canal_Isdn}/${Call_Number},${dial_tout},T)
exten = _0,n,NoOp(Dial Status: ${DIALSTATUS})
exten = _0,n,Macro(Status_Dial|${DIALSTATUS})
exten = _0,n,hangup
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Re: [asterisk-users] Congestion in Outgoing call through PRI
2008/8/30 Shariq Khan [EMAIL PROTECTED]
When i dial out any number through PRI it gives the following error every
time, while incoming calls works fine
I have sangoma E1 PRI card.
-- Executing Dial(SIP/2000-081b9938, Zap/g0/0501125||) in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/0501125
-- Zap/1-1 is proceeding passing it to SIP/2000-081b9938
-- Zap/1-1 is making progress passing it to SIP/2000-081b9938
-- Channel 0/1, span 1 got hangup request
-- Zap/1-1 is circuit-busy
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/2000-081b9938, ) in new stack
== Spawn extension (default, 920501125, 2) exited non-zero on
'SIP/2000-081b9938'
Zaptel.conf
loadzone=us
defaultzone=us
#Sangoma A101 port 1 [slot:4 bus:5 span:1] wanpipe1
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
Zapata.conf
-
[trunkgroups]
[channels]
context=default
usecallerid=no
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A101 port 1 [slot:4 bus:5 span:1] wanpipe1
switchtype=euroisdn
context=from-pstn
group=0
signalling=pri_cpe
channel =1-15,17-31
extensions.conf
---
[globals]
;CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g0 ; Trunk interface
[from-pstn]
exten = 4392839,1,Answer
exten = 4392839,2,Wait(1000)
exten = 4392839,3,Goto(default,1000,1)
[default]
exten = 1000,1,Playback(transfer)
exten = 1000,2,Hangup
exten = _92X.,1,Dial(${TRUNK}/${EXTEN:2},,)
exten = _92X.,2,Hangup
sip.conf
---
[1000]
type=friend
secret=1000
host=dynamic
disallow=all
allow=alaw
allow=ulaw
Where i m on the mistake
Shariq
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Update to latest libpri and tell us if it still demonstrates the problem,
use HEAD version.
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Re: [asterisk-users] sip conversations overlapping!!!!
Every one PSTN line connected to the FXS port of sipura.. Though these 4 lines comes in one cable if that has to do with anything! Not clear for me, develop some more you topology. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk cdr_mysql inexact values
I have a simple cdr configured with the default tables, here is a row of a
good cdr report
calldate | clid | src |
dst | dcontext | channel | ect . ect
2008-08-29 10:16:49 | C. BOUTON 40 | 40 | XXX | phonesystems |
SIP/40-08776938 | ect . ect
I have replaced the number by XX, but it is there. But sometimes
i get this:
calldate | clid | src |
dst | dcontext | channel | ect . ect
2008-08-29 10:17:06 | C. SAGNIER 60 | 60 | s |
phonesystems | SIP/111-08799690 | ect . ect
You see that s in dst ? I know from where it is coming but i have no idea
how to remove it. I am using one macro for dial out, it is easy for me to
manage multiple outgoing peers and max channels for them. I am using
spriority inside that macro, so somehow cdr SOMETIMES report
s as dst. If you can help me to arange my macro to remove that s from cdr or
by any advice i would be gratefull. My macro:
[macro-phonesystems]
exten = s,1,NoOp(We are calling=${ARG1})
exten = s,2,GotoIf($[${GROUP_COUNT(ph0)}=1]?100:3)
exten = s,3,Set(GROUP()=ph0)
exten = s,4,Dial(Sip/${ARG1:[EMAIL PROTECTED],40,TwW)
exten = s,5,NoOP(PH0)
exten = s,100,GotoIf($[${GROUP_COUNT(ph1)}=1]?200:101)
exten = s,101,Set(GROUP()=ph1)
exten = s,102,Dial(Sip/${ARG1:[EMAIL PROTECTED],40,Tw)
exten = s,103,NoOp(PH1)
exten = s,200,GotoIf($[${GROUP_COUNT(ph2)}=2]?300:201)
exten = s,201,Set(GROUP()=ph2)
exten = s,202,Dial(Sip/${ARG1:[EMAIL PROTECTED],40,Tw)
exten = s,203,NoOp(PH2)
exten = s,300,GotoIf($[${GROUP_COUNT(ph3)}=2]?400:301)
exten = s,301,Set(GROUP()=ph3)
exten = s,302,Dial(Sip/${ARG1:[EMAIL PROTECTED],40,Tw)
exten = s,303,NoOp(PH3)
exten = s,400,GotoIf($[${GROUP_COUNT(ph4)}=2]?400:500)
exten = s,401,Set(GROUP()=ph4)
exten = s,402,Dial(Sip/${ARG1:[EMAIL PROTECTED],40,Tw)
exten = s,403,NoOp(PH4)
exten = s,500,Playback(all-circuits-busy-now)
And my portion of extensions.conf from where we are jumping to that macro
exten =
_00[123459]!,1,Monitor(gsm,${CALLERID(num)}APP-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = _00[123459]!,2,GotoIf($[${DB(internet/disponible)}=1]?3:7)
exten = _00[123459]!,3,GotoIf($[${DB(moyende/telecom)}=0]?4:6)
exten = _00[123459]!,4,Macro(phonesystems,${EXTEN})
exten = _00[123459]!,5,Hangup()
;this hangup is for marcro returning
exten = _00[123459]!,6,GotoIf($[${DB(moyende/telecom)}=1]?7:8)
;case 8 should never happen, just in case.
exten = _00[123459]!,7,Dial(mISDN/g:intern-out/${EXTEN:1})
exten = _00[123459]!,8,Dial(mISDN/g:intern-out/${EXTEN:1})
Thank you.
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Re: [asterisk-users] sip conversations overlapping!!!!
Remove pstn lines from sipura and call sipura to sipura ... any problems ? Still with pstn lines removed call sipura1 sipura2 and after sipura 3sipura1 do you still hear any voices? if not it's you cable to pstn. Give us feedback ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
We had some problems with siemens 675ip with audio, but with the correct setup they disappeared, we are using one base and 2 phones. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
I you have such a problems with siemens you should consider 8 voip port linksys gateway with dect bases, their gateway is rock solid and cheap. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] entering a password to have access to a sip account?!
I have one solution in mind, maybe it is an overkill but:
You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all other sip accounts DB(300/NNN)=0 where NNN are all others sip
accounts numbers. You can use set for this, example
exten = 75,1,Set(DB(300/301)=1)
or
exten = 75,1,Set(DB(300/${Callerid(num)}=1)
exten = 76,1,Set(DB(300/${Callerid(num)}=0)
And just go and call from each phone 75 or 76 , i assume that you callerid
is the same as callerid(num) var. The methos is somehow primitive and will
not work if you have 500 extensions, but for 5 sip accounts is a way to go.
Or create external bash script to speed up.
After this you will have as much db entryes as sip accounts in you astdb,
all we need to is is to verify the value before call
exten = 300,1,GotoIf($[${DB(300/${Callerid(num)})}=1]?2:3)
exten = 300,2,Playback(stop_calling_me)
exten = 300,3,Dial(Sip/300)
And again i assume that your sip peers have the same
Callerid(num)=extensions
Maybe i got some syntax errors, but you get the idea.
Have fun
2008/8/24 RoLaNd RoLaNd [EMAIL PROTECTED]
Hello Steve,
thanks for the advice :)
though one prob! if i add the authenticate line itll require all callers to
enter 1234 to access *ANY* sip account..
even though this would come in handy at some point but at the moment i
just want to deny the extension 300 from being able to call 01 unless the
caller entered a password..
find below wht i did so far..
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,authenticate(1234)
exten = 301,5,WaitExten(8)
exten = 301,6,Dial(SIP/100,15) ; goes to operator
exten = 301,3,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
line is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten = 303,1,VoicemailMain ; voicemail box to be redirected to
Date: Sun, 24 Aug 2008 12:05:02 -0400
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] entering a password to have access to a sip
account?!
You want to use Authenticate() between answer and dial.
http://www.google.com/search?q=asterisk+authenticateie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a
Thanks,
Steve Totaro
On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd [EMAIL PROTECTED]
wrote:
Hi all,
i;m obviously a newbie, its been 2 days that im trying to figure out a
way
to deny a specific extension (300) from calling another specific
extensions
(03) except if the caller punch a specified password.. sorry if im not
explaining myself well.. heres an example:
i called my pstn line(with 300 as its sip account), an attendant
answers and
asks me to punch in an extension number right now if i dial 03 it
rings at
the other end! though i dont want that to happen! i want to set
asterisk up
in a way tht if i dial 03 from 300 to ask me for a password... or
it
wont let the line go through!
can anyone guide me through this issue! im really going crazy to get
this
done! any help would truly and utterly be appreciated:)
ps: find below my extensions.conf
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so
it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box
if line
is busy or unavailable
exten = _1XX,3,HangUp()
exten =
Re: [asterisk-users] entering a password to have access to a sip account?!
I have one solution in mind, maybe it is an overkill but:
You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all other sip accounts DB(300/NNN)=0 where NNN are all others sip
accounts numbers. You can use set for this, example
exten = 75,1,Set(DB(300/301)=1)
or
exten = 75,1,Set(DB(300/${Callerid(num)}=1)
exten = 76,1,Set(DB(300/${Callerid(num)}=0)
And just go and call from each phone 75 or 76 , i assume that you callerid
is the same as callerid(num) var. The methos is somehow primitive and will
not work if you have 500 extensions, but for 5 sip accounts is a way to go.
Or create external bash script to speed up.
After this you will have as much db entryes as sip accounts in you astdb,
all we need to is is to verify the value before call
exten = 300,1,GotoIf($[${DB(300/${Callerid(num)})}=1]?2:3)
exten = 300,2,Playback(stop_calling_me)
exten = 300,3,Dial(Sip/300)
And again i assume that your sip peers have the same
Callerid(num)=extensions
Maybe i got some syntax errors, but you get the idea.
Have fun
previous message have failed for some reasons.
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