Re: [asterisk-users] Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2
I see asterisk is finding res_jabber.so not compiled for your asterisk version. As Tim just said, remove all the modules from /usr/lib/asterisk/modules and reinstall asterisk. [2013-01-10 14:20:10] WARNING[27062]: loader.c:804 inspect_module: Module 'res_jabber.so' was not compiled with the same compile-time options as this version of Asterisk. ** [2013-01-10 14:20:10] WARNING[27062]: loader.c:805 inspect_module: Module 'res_jabber.so' will not be initialized as it may cause instability. [2013-01-10 14:20:10] WARNING[27062]: loader.c:895 load_resource: Module 'res_jabber.so' could not be loaded. Leandro 2013/1/10 Tim Nelson tnel...@rockbochs.com First thing to *ALWAYS* check is if you have any Asterisk version specific modules (Fax for Asterisk, G.729, etc). Ensure these are not loaded (noload in modules.conf, or simply move them out of the asterisk modules dir). Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 -- After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root@localhost asterisk-11.1.2]# asterisk -vvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Parsing '/etc/asterisk/logger.conf': Found == Manager registered action DBGet == Manager registered action DBPut == Manager registered action DBDel == Manager registered action DBDelTree == Registered custom function 'MESSAGE' == Registered custom function 'MESSAGE_DATA' == Registered application 'MessageSend' == Manager registered action MessageSend == Manager registered action DataGet == Parsing '/etc/asterisk/codecs.conf': Found Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found == Parsing '/etc/asterisk/dnsmgr.conf': Found [2013-01-10 14:20:10] ERROR[27062]: config_options.c:512 aco_process_config: Unable to load config file 'acl.conf' == Parsing '/etc/asterisk/http.conf': Found == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Login == Manager registered action Challenge == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action GetConfig == Manager registered action GetConfigJSON == Manager registered action UpdateConfig == Manager registered action CreateConfig == Manager registered action ListCategories == Manager registered action Redirect == Manager registered action Atxfer == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action PresenceState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Manager registered action SendText == Manager registered action UserEvent == Manager registered action WaitEvent == Manager registered action CoreSettings == Manager registered action CoreStatus == Manager registered action Reload == Manager registered action CoreShowChannels == Manager registered action ModuleLoad == Manager registered action ModuleCheck == Manager registered action AOCMessage == Manager registered action Filter == Registered custom function 'AMI_CLIENT' == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_humbug.conf': Found [2013-01-10 14:20:10] NOTICE[27062]: manager.c:7545 __init_manager: Invalid keyword displaysystemname = yes in manager.conf [general] == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/cdr.conf': Found [2013-01-10 14:20:10] NOTICE[27062]: cdr.c:1613 do_reload: CDR logging disabled, data will be lost. -- CEL logging disabled. == Parsing '/etc/asterisk/udptl.conf': Found [2013-01-10 14:20:10] WARNING[27062]: udptl.c:1413 removed_options_handler: t38faxudpec in udptl.conf is no longer supported; use the t38pt_udptl configuration option in sip.conf instead. [2013-01-10 14:20:10] WARNING[27062]: udptl.c:1415 removed_options_handler: t38faxmaxdatagram in udptl.conf is no longer supported; value is now supplied by T.38
[asterisk-users] Manager event for hint subscribe
Hello, I am playing with the manager interface and it seems I cannot catch the event of a phone subscribing to an hint. Is there a way to catch this kind of event using the manager interface? I use custom device states, so when a phone subscribe to a hint, the device is created on the fly. I'd like to catch these subscription to check if the custom device is valid or not and set it to INVALID if not authorized. Leandro -- _ -- 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] Detect Low Quality Calls - Realtime
2013/1/8 Lenz Emilitri lenz.lo...@gmail.com 2013/1/5 joachim zoach...@securax.org You are pretty much limited to measuring the delay and the jitter. The delay you can somewhat estimate prior to the call (with qualify for example). The jitter / packetloss you can only figure out when the call is already up for a while. (e.g. you might have no issues the first minute, but maybe packet loss will come in bursts after a minute). A few years ago I spoke to a Finnish company that had a commercial solution for automated MOS estimation. So something exists though I have not tested it first-hand. l. For MOS calculation I use voipmonitor, but it computer it at the end of the call. The voipmonitor guy is very handsome, maybe you can sponsor a patch to have the MOS calculation in real time. An external software can get it and halt the call if needed. Leandro -- _ -- 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] Monitor extensions status.
2013/1/8 Luis H. Forchesatto luisforchesa...@gmail.com Greetings. I got two extensions on my asterisk that autenticates from outside our network, via internet. Is there a way to monitor, in certain time periods, if they are available (online) and send some sort of notification if they don't? There are two extensions to monitor, they belong to the same queue. Both must be available to receive calls at the same time and if one or both are offline I must be notified. They stand behind NAT so making Nagios monitor will either report wrong extension status (monitoring the NATing server/router) or simply useless (unless there's a plugin to monitor asterisk extensions). But anyway...I'll be open to opinions. My environment: - Asterisk 1.6.2.13 - Server running Elastix 2.0.0 - DAHDI v. 2.3.0.1 Doing a nagios probe to check for extension status is a matter of just few lines... I think you can have it done by a developer for less than $30 Leandro -- _ -- 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] Monitor extensions status.
Top and bottom post in the same email... don't open again the thread :-) #!/bin/bash res=`sudo /usr/sbin/asterisk -rx 'sip show peer $1' | grep Status | cut -d\: -f 2 | cut -d\ -f 2` if [ $res == OK ] then echo OK is registered exit 0 else echo WARNING peer not registered exit 1 2013/1/8 Luis H. Forchesatto luisforchesa...@gmail.com Hmmmlooks good, but I'm looking for something that I could do. I'm not much of outsorcing. 2013/1/8 Leandro Dardini ldard...@gmail.com 2013/1/8 Luis H. Forchesatto luisforchesa...@gmail.com Greetings. I got two extensions on my asterisk that autenticates from outside our network, via internet. Is there a way to monitor, in certain time periods, if they are available (online) and send some sort of notification if they don't? There are two extensions to monitor, they belong to the same queue. Both must be available to receive calls at the same time and if one or both are offline I must be notified. They stand behind NAT so making Nagios monitor will either report wrong extension status (monitoring the NATing server/router) or simply useless (unless there's a plugin to monitor asterisk extensions). But anyway...I'll be open to opinions. My environment: - Asterisk 1.6.2.13 - Server running Elastix 2.0.0 - DAHDI v. 2.3.0.1 Doing a nagios probe to check for extension status is a matter of just few lines... I think you can have it done by a developer for less than $30 Leandro -- _ -- 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 -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com -- _ -- 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 -- _ -- 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] Limit registration concurrency per friend
2013/1/5 XBrian bobo...@yahoo.co.uk Can I restrictthe number of concurrent registrations per friend? Your question has no meaning. The registration is the way a peer says to asterisk which is the IP address and port to use to contact him. There can be just one registration active at time. If two or more peers attempt to register at the same time, the last one is the only one working. Leandro -- _ -- 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] RES: Auto ban IP addresses
I am using fail2ban on all my asterisk server, but beware, fail2ban can be a dangerous software. The problem rely on the fact that SIP uses UDP, so it is possible to send messages with a forged source IP address. This way the bad guy out there can ban all your IP addresses. I say it is possible without having investigated in deep details what is really needed to do. Leandro 2013/1/3 Éder e...@openminds.com.br Howto fail2ban in asterisk http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Frank Enviada em: quarta-feira, 2 de janeiro de 2013 20:50 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] Auto ban IP addresses Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? Thank you -- _ -- 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 -- _ -- 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 -- _ -- 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: How to know since when it is used? How to shutdown after max time?
2013/1/3 bilal ghayyad bilmar...@yahoo.com Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected (stuck) for long time. I know how to write the extensions.conf in a way to handle the hangup properly, also I send the incoming calls to the voicemail to be sure it is hanged up properly. One more thing, I set the rtptimeout in case there is any problem in the sip phone and the network .. But, still after sometime, I am surprised that some channels are stuck and stayed connected and then I have to reset it manually !! This is happening only in the analoge channels. What other than the rtptimeout, the hangup in the extensions.conf, the voicemail? Is there anything I have to take care for it that might cause this stuck and keeping the channel openned? By the way, for such cases, what should I place the value of the rtpkeepalive as currently it is 0? What other things I have to take care for it? Regards Bilal I checked on my PBX and I find no way to identify the duration of a call involving a DAHDI channel like it happens on SIP channels. I think the only way will be to assign a not so huge AbsoluteTimeout to each call. Leandro -- _ -- 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] Verizon SIP trunking Field Trial
2013/1/3 Steven Howes steve-li...@geekinter.net On 3 Jan 2013, at 15:13, Michael L. Young wrote: So, I am asking the community for any input. I have read on here and seen on IRC that some in the community are successfully using Asterisk with Verizon SIP. Verizon was going to check and see if they have any notes about that and those particular setups. Can anyone help share any information or tidbits on how they were able to sucessfully work with Verizon? I *think* Verizon require IPSEC for the signalling, so it may be worth reading up on configuring IPSEC in Linux (or acquiring additional hardware) whilst you're looking at the Asterisk part. This could have just been for a specific product / contract or something, I don't recall the details exactly. S -- I have no direct experience with Verizon, but another big player asks for a long series of tests, like call and answer, call and don't answer, call and cancel. It took me two full days of work to accomplish all the tasks. For every call I have to dump the Call-ID, the date and the hours... So, don't be scared by the field test, it will be probably long and tedious, but doable. Leandro -- _ -- 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] new user help required to build voice recorder with asterisk
I don't know how many I/O can be achieved on a modern hardware, but I don't think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of data. However can be a good idea to start loading a server and be prepared to share the load on another server. Leandro 2013/1/2 Steve Totaro stot...@asteriskhelpdesk.com Top post for the New Year. Yes, if you might scale up to 60 or more simultaneous calls, definitely look at OrecX or RTPTap because you will run into I/O issues. Not sure what current hardware can accommodate but it is best not to find out. Considering the very low cost of hardware these days compared with the cost of possible downtime, poor audio, lost recordings or whatever else you can assign a monetary value, I always suggest a separate machine for Passive recording when dealing with more than a handful of simultaneous calls. Thanks, Steve Totaro On Wed, Jan 2, 2013 at 6:18 AM, Lenz Emilitri lenz.lo...@gmail.com wrote: With just one PRI card this should not be an issue, but for larger systems you may consider using something like Oreka to offload the I/O from the Asterisk server l. 2012/12/31 Vinod Nadiadwala thinw...@gmail.com Hi, I am new to asterisk, i want to know that is it possible to use asterisk for build voice recording system. Scenario : ISDN PRI line (30 line) I want every incoming outgoing call has to recorded, but without manual action. system has to auto receive the call. Please suggest, how should i start and with which hardware / cards it is possible. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- 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 -- _ -- 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 -- _ -- 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] new user help required to build voice recorder with asterisk
You should start getting a PRI card. I have good result with both Sangoma
and Digium one. After having configured the card in the system (libpri,
dahdi and asterisk part), it is a matter of few asterisk configuration row
to save all calls to a wav file.
For example, if your incoming calls are put in the incoming context and
your PRI card is identified with the g1 group, the dialplan can be as easy
as the following:
context incoming {
_X. = {
MixMonitor(${UNIQUEID}.wav);
Dial(DAHDI/g1/${EXTEN});
}
}
Leandro
2012/12/31 Vinod Nadiadwala thinw...@gmail.com
Hi,
I am new to asterisk, i want to know that is it possible to use asterisk
for build voice recording system.
Scenario :
ISDN PRI line (30 line)
I want every incoming outgoing call has to recorded, but without manual
action. system has to auto receive the call.
Please suggest, how should i start and with which hardware / cards it is
possible.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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--
_
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Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
Have you configured the canreinvite=yes in sip peer? I am currently off work for two days, but a 100% fail means a configuration problem for sure. Leandro 2012/12/27 Eric Wieling ewiel...@nyigc.com We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings 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 -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
I usually set directmedia=yes with good results... Leandro 2012/12/27 Christopher Harrington ch...@acsdi.com On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling ewiel...@nyigc.com wrote: sip.conf settings: directmedia=yes I know you've said you tried it both ways, but consensus seems to be that directmedia needs to be =no when using UDPTL. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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 -- _ -- 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] disconnect supervision
If you are using an analogue/sip ata, then the problem is on the ata. Run a packet capture and you'll see the invite coming from the ata without nobody using the phone... I am typing from my mobile phone... Il giorno 11/dic/2012 18:55, Joseph syscon...@gmail.com ha scritto: On 12/11/12 11:48, Danny Nicholas wrote: In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and hanguponpolarityswitch lines. If they aren't present, the default values are being used. If they are, tweak them and restart asterisk and dahdi. I do this - service asterisk stop; service dahdi restart; service asterisk start. I'm not using dahdi.conf I'm using extension.conf sip.conf with analog AudioCodes gateway FXO/FXS -- Joseph -Original Message- From: asterisk-users-bounces@lists.**digium.comasterisk-users-boun...@lists.digium.com [mailto:asterisk-users-**boun...@lists.digium.comasterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] disconnect supervision On 12/11/12 11:30, Danny Nicholas wrote: Could be, but I'd check the easier to fix polarity settings. How do I do that? Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week. -- Joseph -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] - configure ring group
100 extension on a row is not feasible... the queue strategy is the only possible solution. If you check the queue.conf file you'll find you can define a Queue and add as many members you like. One of the strategy available is the Ring all where all the members in the queue will be ring. You can let your peers to log in/log out of the queue via dialplan Leandro 2012/12/6 Paolo De Michele pa...@paolodemichele.it hi all, thanks for your replies if you have 100 extensions, put them all into a single string? so: (SIP/1001SIP/1002SIP/1003...until you get to 100? It is very difficult to manage such a thing, no? I don't understand the queues,ringall. can someone explain? thanks in advance On 12/05/2012 10:59 PM, Danny Nicholas wrote: You “can” do the queues/ringall, but you’re increasing your pay grade by doing so. ** ** *From:* asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com] *On Behalf Of *Carlos Rojas *Sent:* Wednesday, December 05, 2012 3:58 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] - configure ring group ** ** Maybe, ** ** You can do that, with queues, and ringall strategy. On Wed, Dec 5, 2012 at 4:53 PM, Leandro Dardini ldard...@gmail.com wrote: You can dial all the extensions at once, putting all them in the dial string, separated by . There is no other method. ** ** Leandro 2012/12/5 Paolo De Michele pa...@paolodemichele.it hi all, I want have an information about ring group in asterisk (1.8.16 - centos 6.3) I have configured skypeforasterisk for incoming call to one extension and it works now,my chan_skype.conf is: [general] default_user=user-skype [user-skype] secret=x context=from-skype exten= disallow=all allow=ulaw allow=alaw my extensions.conf: [from-skype] exten = ,1,Verbose(2,Incoming Skype Call) same = n,Answer() same = n,Dial(SIP/1000SIP/2000SIP/3000,30) same = n,Playback(useris-curntly-unavail) same = n,Hangup() at right time the internal ring are 1000, 2000 and 3000 I have the extension from 1000 to 1005, 2000 to 2005 and from 3000 to 3005 I can ring him all? I can group the configuration into a single string? let me know something thanks in advance ** ** -- _ -- 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 ** ** -- _ -- 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 ** ** -- _ -- 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 -- _ -- 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 -- _ -- 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] CDR - Freepbx - Safe to add primary key to table ?
Yes, go for it. However I have added another autoincrement column and created the primary key on it. On the other columns I need to search I have created just an index. Leandro 2012/12/6 Olivier oza_4...@yahoo.fr Hello, I need to develop an application that will query (mostly reading) an existing MySQL CDR database. This database (named asteriskcdrdb) was created during Freepbx 2.10 install on my asterisk 1.8 setup. This database has a single CDR table which is filled by Asterisk. The tools I'm planning to use require this table to include a Primary Key. Is it safe to Alter this table telling it to use UniqueID column as a Primary Key ? (Sure, I'll test this on a database copy but I'm not confident my tests will cover everything) Regards -- _ -- 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 -- _ -- 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] CDR - Freepbx - Safe to add primary key to table ?
The reason I add a new column autoincrement is due to the fact I trust more mysql about uniquness than asterisk. Leandro I am typing from my mobile phone... Il giorno 06/dic/2012 19:11, Ron Wheeler rwhee...@artifact-software.com ha scritto: It seems like a safe thing to do. You could also ask about the impact of making an existing column a primary key, in a MySQL forum. Leandro's solution seems to be a good one as well and does guarantee uniqueness. Ron On 06/12/2012 12:25 PM, Leandro Dardini wrote: Yes, go for it. However I have added another autoincrement column and created the primary key on it. On the other columns I need to search I have created just an index. Leandro 2012/12/6 Olivier oza_4...@yahoo.fr Hello, I need to develop an application that will query (mostly reading) an existing MySQL CDR database. This database (named asteriskcdrdb) was created during Freepbx 2.10 install on my asterisk 1.8 setup. This database has a single CDR table which is filled by Asterisk. The tools I'm planning to use require this table to include a Primary Key. Is it safe to Alter this table telling it to use UniqueID column as a Primary Key ? (Sure, I'll test this on a database copy but I'm not confident my tests will cover everything) Regards -- _ -- 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 -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 -- _ -- 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] - configure ring group
You can dial all the extensions at once, putting all them in the dial string, separated by . There is no other method. Leandro 2012/12/5 Paolo De Michele pa...@paolodemichele.it hi all, I want have an information about ring group in asterisk (1.8.16 - centos 6.3) I have configured skypeforasterisk for incoming call to one extension and it works now,my chan_skype.conf is: [general] default_user=user-skype [user-skype] secret=x context=from-skype exten= disallow=all allow=ulaw allow=alaw my extensions.conf: [from-skype] exten = ,1,Verbose(2,Incoming Skype Call) same = n,Answer() same = n,Dial(SIP/1000SIP/2000SIP/3000,30) same = n,Playback(useris-curntly-unavail) same = n,Hangup() at right time the internal ring are 1000, 2000 and 3000 I have the extension from 1000 to 1005, 2000 to 2005 and from 3000 to 3005 I can ring him all? I can group the configuration into a single string? let me know something thanks in advance -- _ -- 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 -- _ -- 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] Asterisk not starting (illegal instruction core dumped)
I suspect you have something wrong in your server hardware... have you tried running a memtest? Leandro 2012/11/27 Adolphus Enaboifo adolphus.enabo...@osenkorp.com Hi List members, Thanks for the support so far as I try to install and test my first asterisk system. I was able to finally install asterisk-1.8.18.0 with libpri-1.4.13 and dahdi-linux-complete-2.6.1+2.6.1 according to the instructions given in the online documentation (asterisk the definitive guide). But while trying to start asterisk with the following command /usr/sbin/asterisk -cvvv or /usr/sbin/asterisk -c I get the message Illegal instruction (core dumped) Kindly advice on what to do. thanks Adolphus Enaboifo -- _ -- 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 -- _ -- 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] leading ghost 0
I am not really sure, restarting asterisk and dahdi can be the most obvious thing to do, but restarting the dahdi kernel module can be useless if you haven't changed the kernel module configuration and reloading the module in asterisk can be enough if you have changed just the chan_dahdi.conf Leandro 2012/11/21 Frederic Van Espen frederic...@gmail.com Then if you did not restart dahdi and asterisk, then the changes to the parameters in chan_dahdi.conf and system.conf were never taken into account. There is no other way than really restarting asterisk and dahdi. Frederic On Wed, 2012-11-21 at 09:08 +0100, gincantalupo wrote: I cannot restart dahdi because the PBX is in production, all I can do is a module reload chan_dahdi.so. -- _ -- 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 -- _ -- 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] leading ghost 0
2012/11/20 gincantalupo gincantal...@fgasoftware.com Hi all, I have problems dialling out because my new telco (the previous gave no problems) tells me my PBX adds a leading 0 and that's why I cannot dial out (but I can receive calls). I make a small extensions.conf as a test: exten = 666,1,Dial(DAHDI/g1/339xx) but cannot dial out Curious thing is that exten = 666,1,Dial(DAHDI/g1/**0233xx) and exten = 666,1,Dial(DAHDI/g1/233xx) call the same number!!! Line in use is a PRI. My Asterisk version is 1.4.26.2 dahdi version: 2.2.0.2 wanpipe-3.4.6 I checked with intense pri debug and see no 0 inside frames How can I really be SURE Asterisk is not adding some leading zero? Thank you. Giorgio. I have never heard of a way to automatically add digits when using PRI, however can you check your chan_dahdi.conf about the following lines: internationalprefix = nationalprefix = localprefix = If presents, try messing with them. If you are using the PRI in Italy, every provider has PRI configured in its own way, some time even the same provider is configuring PRI lines in multiple times, but often the problems are on receiving the calls (like calls with and without the area code, with or without the leading zero, etc. etc.) Leandro -- _ -- 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] leading ghost 0
That is a real mistery! I like a lots these cases when all seems not working despite all being correctly configured, but you know first or later you'll find the answer. From your website, it seems you are selling/renting PBX based on asterisk, so you can be sure nobody has messed with the asterisk or dahdi source code adding a zero... I am sure you have already tried with a brand new server. Have you checked the pridialplan and prilocaldialplan setting? If I was in your shoes, I'll get another server, with a PRI configured as master and hook it at your PBX to really check if the zero is sent. Does the technician try to make phone calls from the same network cable you are using? Leandro 2012/11/20 gincantalupo gincantal...@fgasoftware.com ** Hi Leandro, thanks for your answer. I already have tried those parameters but without any positive result. The telco technician has tried the line with its machine and it worked...remote telco technicians say they get a leading zero... I'm thinking there is something strange in the middle that adds the zero but do not know what it is. Strange is the fact that you can call some numbers with or without the prefix zero... Moreover we had no problem with the previous telco (fastweb). So we can only call PTSN numbersnot mobile phones. Giorgio On 11/20/2012 11:12 AM, Leandro Dardini wrote: 2012/11/20 gincantalupo gincantal...@fgasoftware.com Hi all, I have problems dialling out because my new telco (the previous gave no problems) tells me my PBX adds a leading 0 and that's why I cannot dial out (but I can receive calls). I make a small extensions.conf as a test: exten = 666,1,Dial(DAHDI/g1/339xx) but cannot dial out Curious thing is that exten = 666,1,Dial(DAHDI/g1/0233xx) and exten = 666,1,Dial(DAHDI/g1/233xx) call the same number!!! Line in use is a PRI. My Asterisk version is 1.4.26.2 dahdi version: 2.2.0.2 wanpipe-3.4.6 I checked with intense pri debug and see no 0 inside frames How can I really be SURE Asterisk is not adding some leading zero? Thank you. Giorgio. I have never heard of a way to automatically add digits when using PRI, however can you check your chan_dahdi.conf about the following lines: internationalprefix = nationalprefix = localprefix = If presents, try messing with them. If you are using the PRI in Italy, every provider has PRI configured in its own way, some time even the same provider is configuring PRI lines in multiple times, but often the problems are on receiving the calls (like calls with and without the area code, with or without the leading zero, etc. etc.) Leandro -- _ -- 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 -- _ -- 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 -- _ -- 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] leading ghost 0
Not only, you have to restart dahdi/zaptel as well. Leandro 2012/11/20 Frederic Van Espen frederic...@gmail.com On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote: I'm sure nobody has added something... tried prilocaldialplan and pridialplan but nothing changed. Question: if pridialplan or prilocaldialplan would work, should I see the 0 inside PRI frame with intense debug or it is hidden? Somebody correct me if I'm wrong but I think you have to restart asterisk when you change these settings on dahdi. Keep that in mind. Cheers, Frederic -- _ -- 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 -- _ -- 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
[asterisk-users] Extension hints, which info available?
Hello,
I want to manage hints in a different way, putting all the hints in the
same context and trying to recognize the subscribing peer, but I can't find
any variable set about the calling peer. Peers need to be authenticated to
be able to subscribe to the hint, but I am not able to access any of the
info usually available when a registered peer place a call, like
${CDR(accountcode)} or ${CHANNEL(peername)} ... the only variable I can use
is ${EXTEN}
Leandro
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Re: [asterisk-users] Realtime Hints
Thank you,
I think I'll surrender in trying to use the realtime extension and use
instead the simple ODBC interface. However I'd like to access some channel
variables. Which ones are available inside the extension hint porcessing? I
tried ${CDR(accountcode)} and it is not available, nor the
${CHANNEL(peername)} ... what is the sd.name that you are referring?
Leandro
2012/9/25 Stephen Collier stephen.coll...@foxaus.com
We use something like below
[blf]
exten =_ZXX!,hint,SIP/${ODBC_FINDEXTN(sd.name,${EXTEN})}
This uses an odbc call to create the hint when the phone asks for it.
Using snom 760 and 821
Cheers
Stephen
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Re: [asterisk-users] Realtime Hints
Thank you again,
The problem in this setup is the inability to isolate a group of extensions
from others. I mean, if all hints are in the same context, each extension
can subscribe to any of the hints. The reason I prefer a completely
realtime hints was I'd like to dynamically create hints in dynamically
created context to jail an extension to only the hints I decide.
I have tried to contact Digium on this topic and I am ready to setup a
bounty to have a true realtime hints management coded if not already
available.
Leandro
Il giorno 26/set/2012 00:03, Stephen Collier stephen.coll...@foxaus.com
ha scritto:
I use the following in func_odbc.conf
[FINDEXTN]
dsn=asterisk
readsql=SELECT ${ARG1} FROM extension_map as em left join sip_devices as
sd on s
d.id = em.name_id WHERE em.extension ='${ARG2}' and name_id IS NOT NULL
this is for our own extension_map table which is part of our mapping to
our Avaya users.
A simple one would be
[FINDEXTN]
dsn=asterisk
readsql=SELECT ${ARG1} FROM sip_devices as sd WHERE sd.name ='${ARG2}'
This allows pulling any field from sip_devices which is our realtime sip
table.
You could pull some of the other data you are looking for.
Cheers
Stephen
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[asterisk-users] Realtime Hints
Hello, I'd like to start using realtime hints in my asterisk 1.8 dialplan, but I am unable. I haven't understood if they have to be put inside the extensions realtime table (with priority -1) or if a dedicated realtime hints table can be made. Neither ways seem to work. Have you any working example? Leandro -- _ -- 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] graceful restart
You can see if asterisk has been restarted by checking the number of calls processed. If almost zero, it has been restarted. core show calls Leandro 2012/8/19 Jan Blom jan.b...@peopleinteractive.se Hello, ** ** Is there a way to detect, via cli or any other way, that Asterisk is in “graceful shutdown” mode, not accepting any new calls? Or to put the question a different way, how can I know that Asterisk has restarted again after the command “core restart graceful” in an automated way? ** ** ** ** Best regards, Jan Blom -- _ -- 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 -- _ -- 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] Segmenting A Configration File
One of my clients uses thirdlane. The web interface is clean and nice, but
asterisk completely locks when one of the client change the config and
reloads during peak hours. It is possible my client uses an old version or
hasn't applied all the patches or hasn't configured asterisk in the right
way or the underlying hardware is not enough powerful, but I'd not suggest
it to manage over than few hundred peers.
I have seen several software solutions, all configuration file based and
they all have the same problem. They locks asterisk on reload, so you'll
end not altering the configuration during peak hours and you'll avoid
giving the clients the ability to change their config. The only key
solution is to have a completely realtime version.
Leandro
2012/8/12 Carlos Rojas crt.ro...@gmail.com
Hi
Have you seen thirdlane?
Thirdlane has a multitenant version.
Regards
On Aug 11, 2012 11:11 AM, Carlos Alvarez car...@televolve.com wrote:
On Sat, Aug 11, 2012 at 3:16 AM, Kannan vasdevelo...@gmail.com wrote:
I am planning a multi-tenant VoIP services system with Asterisk, using
configuration tweaks. Having all the tenant configurations in one
configuration file is overwhelming. I would like to segment the
configuration files and include them in the main configuration file. Is it
possible?
For e.g. I would like to have the main extenstions.conf file to include
tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy
to manage the configurations of each tenant.
We put each tenant's sip and extensions config files in
/etc/asterisk/accounts and then do an include for that directory in the
main files.
We keep all the voicemail.conf in one because changes to passwords will
NOT be saved to included files. We used to use includes for voicemail but
that meant no password changes.
The main file has a list of all phone numbers in the system in numerical
order where we set the account name, and then we send them to the proper
context like this:
exten = 12015551212,1,Set(CDR(accountcode)=johnsmith)
exten = _X.,n(cont),Goto(${CDR(accountcode)}#did,${EXTEN},1)
There's a bunch of other stuff in there where we do line counting and
such.
--
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Segmenting A Configration File
Sure, you can include multiple files from the general extension.conf. You can do the same for the sip.conf. Leandro I am typing from my mobile phone... Il giorno 11/ago/2012 12:17, Kannan vasdevelo...@gmail.com ha scritto: Hi List, I am planning a multi-tenant VoIP services system with Asterisk, using configuration tweaks. Having all the tenant configurations in one configuration file is overwhelming. I would like to segment the configuration files and include them in the main configuration file. Is it possible? For e.g. I would like to have the main extenstions.conf file to include tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy to manage the configurations of each tenant. Thanks. -- _ -- 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 -- _ -- 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] Multiple channel for SIP users
2012/8/11 Hatos Gabor ha...@ggki.hu Hi Team, I use Asterix 1.6.2.9-2 what is running on debian squeeze. I completely statisfied this software. I did everything I want so far. I love it so much, but there is a point where I can not step through. 1) I have connected to my telephone provider as a SIP client, but my Asterisk only one call make to the world in same time. My provider does not limit the number of simultaneous calls. The only limit is the bandwidth of my local internet link. How can I configure my asterisk to create more than one simultaneous calls through my provider? Asterisk has no limitation on the number of simultaneous calls. Just place another call while one call is already going... 2) If I use an ATA, which has 2 SIP clients. These SIP clients is the same asterisk user, but asterisk register only the last one. May I got chance for registering ATA with the same users in the asterisk or every ATA must have two different asterisk user for working well? Ata I have found so far allows to set two distinct SIP account for each one of the FXS/FXO ports they have. Leandro Thanks for any hints in advance! Best regards, Gabor Hatos -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] IAX with two asterisk boxes
2012/8/9 Ashish Agarwal ashisha...@gmail.com Hello, I have two asterisk boxes running and both are using DAHDI PRI Card. I wish to know if IAX is the best method to connect both the boxes? IAX2 is a great protocol, it can do amazing things in saving bandwidth (with the trunking feature) and it plays friendly with NAT. If you haven't NAT or bandwitdh problem, I'll prefer SIP over IAX2. Also, need some help with the following? 1. For incoming call on server2 I wish to run an IVR to the user for which all my prompt sound files resides on server1. Is there a way I can achieve this? There are several ways. The simplest will be to share via NFS the directory holding the sound files. A more tricky one was to actually forward the call to server 1 and play the sound files on it 2. I am also using .call file at times to make outgoing call to the user where IVR will be played but I will initiated the .call file from server1 spool but the call should use server2 dahdi lines and also stream the file from server1? The call can go from server1 to server2 and then use the local dahdi lines. The audio part can be build on server1 as well. Asterisk is a really flexible software, you can do always what you want and usually, for every problems, there are few solutions... Leandro Please suggest -- Regards, Ashish -- _ -- 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 -- _ -- 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] Suggestion of Server Specifications for Asterisk
Let us know how does it performs... Leandro 2012/8/6 Shahid H shah...@gmail.com I have bought a new server today: i7-2600 CPU, 8GB and 2 x 256GB SSDs. 100Mbit Connection. I hope CPU is powerful enough for 200 concurrent calls. On Sun, Aug 5, 2012 at 1:57 AM, Michelle Dupuis mdup...@ocg.ca wrote: That's how we do it - write to a memory based (ramdisk) disk then write to HDD upon call completion. We haven't tried a SSD but that may be necessary depending on your call volumes. -- *From:* asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner [ rswago...@gmail.com] *Sent:* Saturday, August 04, 2012 7:34 PM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Suggestion of Server Specifications for Asterisk On Sat, Aug 4, 2012 at 1:22 PM, Shahid H shah...@gmail.com wrote: Instead of buying expensive disk.. I might setup a ramdisk (about 2GB) to do 200 calls recordings. Once the call hangup/completed it will then move recording file to SATA HDD. What do you think of this? You want some form of raid for redundancy. I usually go with two 15K SAS drives in raid 1 or four 7.2k SATA drives in raid 10. Performance between the two should be similar. With drives being as cheap as they are skip raid 5. Ryan -- _ -- 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 -- _ -- 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 -- _ -- 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] Suggestion of Server Specifications for Asterisk
The busiest server I am managing reaches 120 concurrent channels (with mixed recording). It is a dual processor, dual core Intel 5150 with 16 GB of ram and raid sas controller. The load reaches rarely 3.0. Having to double the number of channels and due to the 100% call recordings, I'll go with a 16 cores. Memory will not a big issue and so the disk. 64kbit/s x 200 (even adding the overhead of the SIP and IP) will be under 20 Mbit/s, so a 100 Mbit/s will be fine. About UK provider, I can't be of any help... I know very good providers in Germany and Canada, where I am laying my servers, but none in UK. Leandro 2012/8/4 Shahid H shah...@gmail.com What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -- _ -- 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 -- _ -- 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] Suggestion of Server Specifications for Asterisk
It is not necessary to use an high performance drive. The bottleneck will be the processor, not the disk. A single disk can handle ten times the load of 200 ulaw channels. Leandro Il giorno 04/ago/2012 12:39, Shahid H shah...@gmail.com ha scritto: Would a SSD drive be enough or do I need like Raid 10 (4 hard drives)? On Sat, Aug 4, 2012 at 10:17 AM, Leandro Dardini ldard...@gmail.comwrote: The busiest server I am managing reaches 120 concurrent channels (with mixed recording). It is a dual processor, dual core Intel 5150 with 16 GB of ram and raid sas controller. The load reaches rarely 3.0. Having to double the number of channels and due to the 100% call recordings, I'll go with a 16 cores. Memory will not a big issue and so the disk. 64kbit/s x 200 (even adding the overhead of the SIP and IP) will be under 20 Mbit/s, so a 100 Mbit/s will be fine. About UK provider, I can't be of any help... I know very good providers in Germany and Canada, where I am laying my servers, but none in UK. Leandro 2012/8/4 Shahid H shah...@gmail.com What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] Suggestion of Server Specifications for Asterisk
A single sata disk will be an unacceptable single point of failure. Get three disks and get in raid5 configuration. You'll gain in safety and speed. About the CPU model, I am a bit lazy, check the latest CPU released from intel or amd (I love amd cpu). Leandro Il giorno 04/ago/2012 14:30, Shahid H shah...@gmail.com ha scritto: Ahh I see. So I might as well get a normal sata disk? I thought I/O will be Bottleneck as well because 200 channels WAV recordings to disk at the same time. Which intel model 16 cores do you recommend? how about 12 cores? Thanks! On Sat, Aug 4, 2012 at 1:19 PM, Leandro Dardini ldard...@gmail.comwrote: It is not necessary to use an high performance drive. The bottleneck will be the processor, not the disk. A single disk can handle ten times the load of 200 ulaw channels. Leandro Il giorno 04/ago/2012 12:39, Shahid H shah...@gmail.com ha scritto: Would a SSD drive be enough or do I need like Raid 10 (4 hard drives)? On Sat, Aug 4, 2012 at 10:17 AM, Leandro Dardini ldard...@gmail.comwrote: The busiest server I am managing reaches 120 concurrent channels (with mixed recording). It is a dual processor, dual core Intel 5150 with 16 GB of ram and raid sas controller. The load reaches rarely 3.0. Having to double the number of channels and due to the 100% call recordings, I'll go with a 16 cores. Memory will not a big issue and so the disk. 64kbit/s x 200 (even adding the overhead of the SIP and IP) will be under 20 Mbit/s, so a 100 Mbit/s will be fine. About UK provider, I can't be of any help... I know very good providers in Germany and Canada, where I am laying my servers, but none in UK. Leandro 2012/8/4 Shahid H shah...@gmail.com What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] Suggestion of Server Specifications for Asterisk
Benny is right, if writes are smaller than the stripe size, there is no gain in speed in using raid5. Not only, but you can have lower performance than a single disk. The ramdisk can be a good idea, but if the load is somewhat constant, you end only moving the slow write ahead of time. 200 calls at 64kbit/s are just 1.5 Mbyte/s ... even the slowest disk can accomplish this. Leandro 2012/8/4 Shahid H shah...@gmail.com Instead of buying expensive disk.. I might setup a ramdisk (about 2GB) to do 200 calls recordings. Once the call hangup/completed it will then move recording file to SATA HDD. What do you think of this? On Sat, Aug 4, 2012 at 5:51 PM, Benny Amorsen benny+use...@amorsen.dkwrote: Leandro Dardini ldard...@gmail.com writes: A single sata disk will be an unacceptable single point of failure. Get three disks and get in raid5 configuration. You'll gain in safety and speed. RAID-5 is slower than single disks when it comes to write IOPS (a commit is not done until the slowest disk has answered). Avoid it for write heavy workloads at all costs unless you are writing sequentially in one file with write caching enabled. /Benny -- _ -- 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 -- _ -- 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 -- _ -- 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] Asterisk realtime don't support 'n' as extension's next priority
It is reasonable 'n' is not usable as priority number. How can asterisk know the second priority if all other priority have 'n' as priority number? In a relational database there is no 'sequential read'. In other words, you need to assign the priority to all entries. Leandro Il giorno 03/ago/2012 06:27, virendra bhati virbh...@gmail.com ha scritto: Hi Team, I want to used *'n*' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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 -- _ -- 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] asterisk realtime database structure
If you check the contrib/realtime direco 2012/8/3 Daniel-Constantin Mierla mico...@gmail.com Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like: https://wiki.asterisk.org/**wiki/display/AST/SIP+Realtime,** +MySQL+table+structurehttps://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure https://wiki.asterisk.org/**wiki/display/AST/ODBC+**Voicemail+Storagehttps://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage I have several table structures from the Asterisk 1.6, I dug for them in the code or found on the web when I wrote the tutorial about integration with Kamailio 3.1 (http://kb.asipto.com/**asterisk:realtime:kamailio-3.** 1.x-asterisk-1.6.2-astdbhttp://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb), but hopefully now it is an easy way to get the db structure. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] asterisk realtime database structure
If you check the contrib/realtime/mysql directory in the source tree, you'll find scripts for almost all the tables. Leandro 2012/8/3 Daniel-Constantin Mierla mico...@gmail.com Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like: https://wiki.asterisk.org/**wiki/display/AST/SIP+Realtime,** +MySQL+table+structurehttps://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure https://wiki.asterisk.org/**wiki/display/AST/ODBC+**Voicemail+Storagehttps://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage I have several table structures from the Asterisk 1.6, I dug for them in the code or found on the web when I wrote the tutorial about integration with Kamailio 3.1 (http://kb.asipto.com/**asterisk:realtime:kamailio-3.** 1.x-asterisk-1.6.2-astdbhttp://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb), but hopefully now it is an easy way to get the db structure. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Asterisk realtime don't support 'n' as extension's next priority
I am kissing every inch of land where each one of the asterisk's developer is putting his feet. In the last 10 years I have worked thanks to the availability of the asterisk code. Most of my income was possible just thanks to asterisk, so I am pretty biased when trying to evaluate if the asterisk code is good or not. You can understand if I love the way asterisk has been coded. Nevertheless things can be better and they can be better thanks to you. Asterisk is open source and Mark is a very kind person when you submit patches, so put your ideas in new code and send to him. If you don't know how to code, hire some developer and have him to code your view of a better RT code. If it will be accepted by the core developer, all us will be happy. if it will not accepted, you'll be happy with you own personal branch. I run for a small period of time my personal asterisk tree because the italian telephony system is flawed and clients want services not suitable for the general asterisk audience, so there is nothing to worry to have your personal asterisk code. Leandro PS I think your idea of extension RT can be accomplished with some triggers and replacing the extension table with a view on your own n-enabled extension table 2012/8/3 Bryant Zimmerman brya...@zktech.com Leandro I have to disagree reasonable designers would have done a better job with this one. But we developers are not always so reasonable. The issue is many developers when pushing to put features in they don't put on their designers hat and think out side the box first.Heaven knows I have been guilty of this one over the years and had to go back and refactor. It is not so reasonable to think that this limitation has to exist developers have been putting order by fields in db driven systems for years. What of the guy who want's to use n(target) or 4(target) (I know this may have not been an option when RT was first done now it is) so they can add specialized jumping code. If I had been designing the Realtime (today) I would have added a field for the priority and made it a full alpha / numeric and added an order by field. As it sits now how do you do n, i, h or tags ect It kinda sucks and limits the Realtime. Not to bash on the developer who did this I get that we don't always think out side the box all the time nor was some of this ability available when the RT was written. but know it does so what do we do. Unfortunately I am not a ansi C guy or I could probably fix it . Thanks Bryant Zimmerman (ZK Tech Inc.) -- *From*: Leandro Dardini ldard...@gmail.com *Sent*: Friday, August 03, 2012 2:18 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority It is reasonable 'n' is not usable as priority number. How can asterisk know the second priority if all other priority have 'n' as priority number? In a relational database there is no 'sequential read'. In other words, you need to assign the priority to all entries. Leandro Il giorno 03/ago/2012 06:27, virendra bhati virbh...@gmail.com ha scritto: Hi Team, I want to used *'n*' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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 -- _ -- 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 -- _ -- 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] Asterisk realtime don't support 'n' as extension's next priority
No, numbers have to be in sequence. Leandro I am typing from my mobile phone... Il giorno 03/ago/2012 20:28, Raj Mathur (राज माथुर) r...@linux-delhi.org ha scritto: On Friday 03 Aug 2012, C. Savinovich wrote: You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle? If you ever used BASIC you'd remember the trick is to increment line numbers (priorities) by 10. I presume a dialplan would work fine even if the priorities aren't sequential, as long as they're increasing monotonically. Could someone confirm? Having said that, I use n with abandon. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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 -- _ -- 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] Multi-Tenant PBX with Asterisk
Hello Bryant, it is nice to hear someone with different experience, so I am happy to know the cloud is indeed a feasible environment even for VoIP. Can you share with us some of your configuration magic? Like the cloud service you are using, the power of each node and the load you are experiencing on them in regards to the number of channels active and phone registered? Leandro 2012/7/31 Bryant Zimmerman brya...@zktech.com Kannan I have to disagree with Leanrod. We are a hosted (cloud) PBX company we successfully run our Multi-tenant systems in Virtual machines and have no issues with them. It comes down to designing your virtual environment for your target loads and then not exceeding them. This allows for fail over of hardware and scalability. We have moved our virtual phone switches live with full call loads and have no call drops. We do not usually dedicate a single Virtual Machine to each customer either. We have built our own Multi-tenant PBX on top of asterisk. We achieve many of the features available in freepbx/trixbox (not all). This method allows us to cost effectively service our customers with a presence of scale in mind. It is not uncommon to have 5000 + extensions per virtual switch. This method does require highly skilled engineering to achieve stability. Bryant -- *From*: Kannan vasdevelo...@gmail.com *Sent*: Tuesday, July 31, 2012 12:37 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Multi-Tenant PBX with Asterisk Thanks Leandro for your comments. On Mon, Jul 30, 2012 at 6:35 PM, Leandro Dardini ldard...@gmail.comwrote: 2012/7/30 Kannan vasdevelo...@gmail.com Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. Which one of the method above is generally used by hosted PBX service providers? Isn't the second option with ARA a good choice for dynamic creation of multiple small PBX tenants? Is the last option alone or combination of options 2 and 3 good for cloud based hosted PBX service offering? Thanks, Kannan. Working in the voip field from a lots of years, I have found all three type of business. The first is maybe the easier and most common. Hardware is cheap and it is easier to sell a service like the PBX if it is sold together with a piece of iron. Usually the hardware is placed on client's network, using the bandwidth of the client. Usually together with the PBX is sold also a router/firewall/traffic shaper/vpn endpoint to try to optimize the traffic on the client's DSL. The major pros about this solution is you can use a normal PBX like freepbx/trixbox, the client can mess the config how he likes, without disrupting other services, you can install VoIP card to connect landlines,. The major cons is the cost of the hardware, the cost of the g.729 licenses (if any) and the maintenance cost of replacing hardware failures and the need to be physically near each client. The second is the holy grail of the VoIP providers. The major pros is the cost. Having a single hardware is cheap and it is still cheap also if you decide to get two to be ready in case of an hardware failure. The major cons is the software. You cannot use the award winning freepbx/trixbox family and you need to deal with sometime limited or incomplete developed interfaces. The client always asks for the missing feature. One other major cons is the reload. If the PBX software is not made using ARA, then every time you add a new peer or a new DID, you need to reload the entire PBX and that is a resource killer. Again, if the pbx interface is not made using ARA, then you cannot let your clients to change the configuration or they will trigger continuous reload (and delaying reload for example every 10 minutes is not a solution) The last one is sometime the chosen compromise, but from my point of view, pbxes are not good software to virtualize. They are too sensible to delays and your voice quality can go down if the real server is overloaded. The same for the cloud based solutions (I have yet to found). I suspect the cloud is good for services like http, not for real time applications. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
2012/7/30 Kannan vasdevelo...@gmail.com Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. Which one of the method above is generally used by hosted PBX service providers? Isn't the second option with ARA a good choice for dynamic creation of multiple small PBX tenants? Is the last option alone or combination of options 2 and 3 good for cloud based hosted PBX service offering? Thanks, Kannan. Working in the voip field from a lots of years, I have found all three type of business. The first is maybe the easier and most common. Hardware is cheap and it is easier to sell a service like the PBX if it is sold together with a piece of iron. Usually the hardware is placed on client's network, using the bandwidth of the client. Usually together with the PBX is sold also a router/firewall/traffic shaper/vpn endpoint to try to optimize the traffic on the client's DSL. The major pros about this solution is you can use a normal PBX like freepbx/trixbox, the client can mess the config how he likes, without disrupting other services, you can install VoIP card to connect landlines,. The major cons is the cost of the hardware, the cost of the g.729 licenses (if any) and the maintenance cost of replacing hardware failures and the need to be physically near each client. The second is the holy grail of the VoIP providers. The major pros is the cost. Having a single hardware is cheap and it is still cheap also if you decide to get two to be ready in case of an hardware failure. The major cons is the software. You cannot use the award winning freepbx/trixbox family and you need to deal with sometime limited or incomplete developed interfaces. The client always asks for the missing feature. One other major cons is the reload. If the PBX software is not made using ARA, then every time you add a new peer or a new DID, you need to reload the entire PBX and that is a resource killer. Again, if the pbx interface is not made using ARA, then you cannot let your clients to change the configuration or they will trigger continuous reload (and delaying reload for example every 10 minutes is not a solution) The last one is sometime the chosen compromise, but from my point of view, pbxes are not good software to virtualize. They are too sensible to delays and your voice quality can go down if the real server is overloaded. The same for the cloud based solutions (I have yet to found). I suspect the cloud is good for services like http, not for real time applications. Leandro -- _ -- 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] Multi-Tenant PBX with Asterisk
ARA is an acronym for Asterisk Realtime Architecture and is a different way to keep configuration files in asterisk. Instead of reading configuration from plain files at startup, asterisk read them from database, in realtime. This mean, if you need to add a peer, you drop a new line in the sippeers table and you are fine. You start defining an ODBC source in res_odbc.conf and then configure the ARA source for each plain configuration files in extconfig.conf About the config reload, reloading only the module changed is a good idea, but the commercial GUI I have meet so far doesn't support it. I have clients with very simple dialplan, able to reload it even if more than 130.000 rows long, others, with more complicated dialplan cannot reload it during work hours even if only 30.000 rows long. You are right about freeware PBX for hosted services. Independent from the fact a GUI is free or needs a payment, I think it is important to have the source for it to be able to customize it and also it is important to have a clean dialplan, so you can debug and customize it as well. I am a developer selling software. I never protect my code obfuscating or compiling it and my clients enjoy it and never steal my work (so far). Leandro 2012/7/31 Carlos Alvarez car...@televolve.com I don't know what ARA is. We use just bare Asterisk, no GUI, and from the context it seems that's related to a GUI. We have no problem doing a config reload during production hours. We never do a full reload, just the relevant module (SIP, dialplan, voicemail, etc). I don't believe there is any freeware PBX software that is good for hosted services unless they are kept tiny and limited. Switchvox is excellent as a hosted platform, but extremely expensive and totally closed so you can't customize as needed. And at least 50% of our customers have customization that wouldn't fit into any of the GUI-based systems. You'll need to decide what your market is and your value proposition as well as your ability to learn Asterisk (which I don't think anyone would argue is easy or fast). On Mon, Jul 30, 2012 at 9:41 PM, Kannan vasdevelo...@gmail.com wrote: Thanks Carlos, it is good to hear from one who is in a similar business. Are you getting use of ARA too in similar hosted PBX offerings? On Mon, Jul 30, 2012 at 10:00 PM, Carlos Alvarez car...@televolve.comwrote: On Mon, Jul 30, 2012 at 2:36 AM, Kannan vasdevelo...@gmail.com wrote: 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. We use number two. We dabbled with number three but didn't like the results for a lot of different reasons. As others have mentioned, there is a certain level of danger when you mix companies so closely. We have in the past made a mistake and brought down the whole system, but it's been many years since we've done that. Part is improved skill and part is that Asterisk has improved and no longer commits suicide for certain minor errors. To do this, you need to plan out a good naming convention for everything that will be unique to customers accounts. SIP accounts, macros, contexts, etc etc. We use the accountcode feature and prepend the accountcode through the dial plan and accounts. accountcode.301 would be a SIP account accountcode#function would be a context name We do deploy custom hardware for specific functions or customers who are particularly large in some cases. We just need a good reason to. Like they want to self-manage, or they make a lot of changes, need custom integration with databases, etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Multiple DID for SIP Trunk
Asterisk has some configuration files, like sip.conf holding the peers and trunks details and the most important one, available in several flavours, extensions.conf, extensions.ael... these latest ones are merged toghether at run time. The extension conf file is referred also as 'dialplan' and it contains instructions in how and what to do once a call is received. Asterisk is an extemely flexible systen allowing you to do whatever you like.. Leandro I am typing from my mobile phone... Il giorno 28/lug/2012 11:40, Mitul Limbani mi...@enterux.in ha scritto: by writing dialplan Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 On Sat, Jul 28, 2012 at 2:49 PM, Kannan vasdevelo...@gmail.com wrote: Hi List, We are planning an Asterisk installation with SIP clients on one side, and a SIP trunk on the other side. Is it possible to configure each SIP client with a DID? How to configure each client with the DID? How to configure the SIP trunk with multiple DIDs? Thanks, Kannan. -- _ -- 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 -- _ -- 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 -- _ -- 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
[asterisk-users] Call ID of the second call leg
Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr table) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you Leandro -- _ -- 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
[asterisk-users] Call ID of the second call leg
Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr table) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you Leandro -- _ -- 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
[asterisk-users] Call ID of the second call leg
Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you Leandro -- _ -- 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] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk
15k users are quite a big number. To my clients with a large user base I advice always to partition the load on multiple servers. This has a list of advantages, like the ability to power cycle a node without impacting all your users, easier debug and tests of problems and solutions, abiity to scale up to higher number of peers without changing the overall architecture. Obviously you need a software to manage the multi tenant PBX with asterisk and a special setup to handle the high availability and load balanced configuration. About the fax license, yes, Digium provides fax solutions at cheap prices. However it depends by what you need. If you need T.38 support, Digium fax for asterisk can be the answer, if instead you just need a fax2mail and mail2fax solution, not involving T.38, I always suggest a T1/E1/J1 ISDN network card, far more reliable than a software fax. Leandro 2012/7/23 Kannan vasdevelo...@gmail.com Thanks SamyGo and Mitul for your prompt responses. I have been vested with the responsibility to evaluate Asterisk for a VOIP solution. I was just going through couple of documents and got impressed by the features it has to offer. Our user base is around 15000 and the system should support 1000 concurrent calls, with BHCA projected at 16000. I plan to use OpenSER for SIP registrations and Asterisk for media processing. RTP Proxy will be used for handling NAT traversal. DNS based load balancing/fail over is preferred over Ultramonkey based one. I have quite a few more questions on Asterisk. 1. Can we setup virtual private PBX inside Asterisk. I.e. One installation of Asterisk will handle many configured hosted PBXs. Each virtual private PBX should be able to be configured and managed separately. 2. Do we have to buy Fax licence, if Asterisk is to support end-to-end Fax. I.e. Asterisk will not be used in Fax media procession, but only the SIP signals will be handled by Asterisk. Thanks again for your support. Kind Regards, Kannan. On Mon, Jul 23, 2012 at 10:18 AM, Mitul Limbani mi...@enterux.in wrote: Thats precisely what asterisk has to offer. Mitul On Jul 23, 2012 9:53 AM, Kannan vasdevelo...@gmail.com wrote: Hi List, Is it possible for me to setup PBX, IVR and Conferencing platforms from a single installation with Asterisk? Thanks. -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk
Answers in text. 2012/7/23 Kannan vasdevelo...@gmail.com Thanks Leandro for your reply. See my comments inline. On Mon, Jul 23, 2012 at 12:57 PM, Leandro Dardini ldard...@gmail.comwrote: 15k users are quite a big number. To my clients with a large user base I advice always to partition the load on multiple servers. This has a list of advantages, like the ability to power cycle a node without impacting all your users, easier debug and tests of problems and solutions, abiity to scale up to higher number of peers without changing the overall architecture. Obviously you need a software to manage the multi tenant PBX with asterisk and a special setup to handle the high availability and load balanced configuration. According to some articles in the Internet, 15000 users is a big number when Asterisk handles SIP registrations. That is why I planned to use OpenSIP for SIP registrations. In our architecture, two instances of OpenSIP and three instances of Asterisk will handle the load -- both signaling and media. hardware wise, we will be using 3 Nos. of DL 380 servers. Another one is used for redundancy. Yes, I was asking if multi tenancy is possible with Asterisk PBX? Asterisk is open to every kind of configuration. It has no embedded multi tenancy but a lots of configuration were developed with multi tenancy in mind. I know very little about OpenSIP so I am not the right source of info about it. I never find the need of using something different from asterisk from 2004 when I start using it. About the fax license, yes, Digium provides fax solutions at cheap prices. However it depends by what you need. If you need T.38 support, Digium fax for asterisk can be the answer, if instead you just need a fax2mail and mail2fax solution, not involving T.38, I always suggest a T1/E1/J1 ISDN network card, far more reliable than a software fax. For Fax, we will be using Asterisk as a mediator between Fax terminals and SBC. That is Asterisk itself does not handle the RTP stream for Fax. It is just proxying the SIP signals for Fax. Do I need to but Fax licence? As far I know it, you need a license for every active channel. Leandro Regards, Kannan. Leandro 2012/7/23 Kannan vasdevelo...@gmail.com Thanks SamyGo and Mitul for your prompt responses. I have been vested with the responsibility to evaluate Asterisk for a VOIP solution. I was just going through couple of documents and got impressed by the features it has to offer. Our user base is around 15000 and the system should support 1000 concurrent calls, with BHCA projected at 16000. I plan to use OpenSER for SIP registrations and Asterisk for media processing. RTP Proxy will be used for handling NAT traversal. DNS based load balancing/fail over is preferred over Ultramonkey based one. I have quite a few more questions on Asterisk. 1. Can we setup virtual private PBX inside Asterisk. I.e. One installation of Asterisk will handle many configured hosted PBXs. Each virtual private PBX should be able to be configured and managed separately. 2. Do we have to buy Fax licence, if Asterisk is to support end-to-end Fax. I.e. Asterisk will not be used in Fax media procession, but only the SIP signals will be handled by Asterisk. Thanks again for your support. Kind Regards, Kannan. On Mon, Jul 23, 2012 at 10:18 AM, Mitul Limbani mi...@enterux.inwrote: Thats precisely what asterisk has to offer. Mitul On Jul 23, 2012 9:53 AM, Kannan vasdevelo...@gmail.com wrote: Hi List, Is it possible for me to setup PBX, IVR and Conferencing platforms from a single installation with Asterisk? Thanks. -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api
Re: [asterisk-users] trouble with asterisk behind router
2012/7/13 Nikolay G. Petrov r...@dir.bg Hi guys! I have a some non standard problem when I register my asterisk into My SIP Provider . The trouble is: my asterisk stay behind router with port forwarding, who have Public IP (55.55.55.55 - for example), asterisk have a private IP (192.168.1.2) From My SIP Provider cabinet I see: online device 355@192.168.1.2:5060 Asterisk PBX 1.8.13.0 , but I need from (example): online device 355@55.55.55.55:5060 Asterisk PBX 1.8.13.0 From wich trukes in linux or asterisk technology I need? Can you help? -- Best regards, Nikolay G. Petrov! -- _ -- 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 Sure, the problem is asterisk cannot know its public IP address and thus, the IP inserted in the SIP packets is the private one. You have to specify the internal network and the public IP address in the sip.conf configuration file. externip=55.55.55.55 localnet=192.168.1.0/24 Leandro -- _ -- 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] trouble with asterisk behind router
Il giorno 13/lug/2012 14:00, Nikolay G. Petrov r...@mail.bg ha scritto: 13.07.2012 15:01, Leandro Dardini пишет: 2012/7/13 Nikolay G. Petrov r...@dir.bg Hi guys! I have a some non standard problem when I register my asterisk into My SIP Provider . The trouble is: my asterisk stay behind router with port forwarding, who have Public IP (55.55.55.55 - for example), asterisk have a private IP (192.168.1.2) From My SIP Provider cabinet I see: online device 355@192.168.1.2:5060 Asterisk PBX 1.8.13.0 , but I need from (example): online device 355@55.55.55.55:5060 Asterisk PBX 1.8.13.0 From wich trukes in linux or asterisk technology I need? Can you help? -- Best regards, Nikolay G. Petrov! -- _ -- 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 Sure, the problem is asterisk cannot know its public IP address and thus, the IP inserted in the SIP packets is the private one. You have to specify the internal network and the public IP address in the sip.conf configuration file. externip=55.55.55.55 localnet=192.168.1.0/24 Leandro -- _ -- 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 externip=55.55.55.55 localnet=192.168.1.0/24 Cool! It's work! Respect! The trouble is resolve! -- Best regards! -- Medal medal medal! :-) http://www.youtube.com/watch?v=8qkSe4YM7EYfeature=youtube_gdata_player -- _ -- 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] port 5060 is blocked by ISP
Port 5060 when used with the sip protocol is used witj UDP protocol. Telnet is using TCP. I am typing from my mobile phone... Il giorno 01/lug/2012 09:35, alok srivastava alok...@gmail.com ha scritto: dear i have configured properly asterisk. At the one end i am using x-lite soft ph and another end twinkle. call is going properly from both end but after picking the phone not able to listen other one. when i checked the port 5060 on the asterisk server it is always showing closed while i have flushed all the rules from iptables (iptables -F) PORT STATE SERVICE VERSION 5060/tcp closed sip telnet localhost 5060 (could not connect) regards alok -- _ -- 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 -- _ -- 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] twenty thousands (20, 000) users, which asterisk and how many servers?
20.000 users is really a big number, as big as 2000 concurrent calls. As previously stated on this list, it depends... it depends by the type of calls for example. If all media is offloaded from the server letting the phones to reinvite each other, than your server CAN support the call volume. If instead even a tiny portion of the call volume uses service on the pbx, like IVR, music on hold, conferences, queues or even worst, transcoding, then the server is obviously underpowered. From my point of view, servicing 20.000 users with a single piece of hardware is highly risky. It can broke in the middle of the day, leaving all your users without service. I think a better approach will be to have more less powered servers working all together to serving your users. If a day one or two of them broke, you have not to worry because the other will continue to serve your users and nobody notice the little decrease in power. There are a lots of way to achieve the high availability, load sharing, each with its pros and cons. Right now I am building a pbx with high availability and load sharing in mind, for a client who wants to achieve numbers you have just said. Let's see how it works in few months. Leandro 2012/5/23 bilal ghayyad bilmar...@yahoo.com Hi All; I need to use Asterisk for 20 000 users, so which asterisk version to be used? Is there asterisk version that supports 20,000 users on one hardware machine? Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to handle 20 000 users, and concurrent calls 2000? Or I need multiple servers, how much? If I am going to use multiple servers (until now I do not know how much, and I do not know if the barrier will be the asterisk software or the hardware), then do I have to use special SIP proxy or I have to use load balancer)? In this case, I have to use asterisk Database (so all the servers will read/write from the database)? What about AsteriskNow, can it support? Regards Bilal -- _ -- 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 -- _ -- 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
[asterisk-users] Jumping inside a macro with AEL
Hello,
I am not able to jump to a label from inside a macro. The goto is made
inside a catch while the label is in the body of the macro:
macro recordMessage() {
Answer();
recordagain:
Playback(after-the-tone);
Playback(say-temp-msg-prs-pound);
record(/tmp/${UNIQUEID}.wav);
earagain:
Playback(/tmp/${UNIQUEID});
Background(press-1);
Background(to-hear-msg-again);
Background(press-2);
Background(to-rerecord-yr-message);
Background(press-pound-save-changes);
WaitExten(15);
catch 1 {
goto earagain;
};
catch 2 {
goto recordagain;
};
catch # {
AGI(uploadMedia.php,/tmp/${UNIQUEID}.wav,wav,${TENANTID});
Playback(your-msg-has-been-saved);
};
};
The goto earagain fails because the label is searched inside the 1
extension.
How can I jump correctly to the label?
Leandro
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Re: [asterisk-users] Asterisk 1.6.2 1.8.12
You have to create yourself the odbc commands using func_odbc.conf file,
like:
[GET_HUNTLIST_TYPE]
dsn=asterisk1,asterisk2
synopsis=Get the Hunt List type
readsql=SELECT hu_type,hu_ringtime from hu_huntlists where hu_id='${ARG1}'
then you cna use it in your dialplan (I use AEL):
Set(ARRAY(TYPE,DIALTIMEOUT)=${ODBC_GET_HUNTLIST_TYPE(${ID})});
Leandro
2012/5/5 Jonas Kellens jonas.kell...@telenet.be
**
Will ODBC become the default then ?
I see no ODBC-command to use in the dialplan.
Jonas.
On 05/05/2012 11:12 AM, Leandro Dardini wrote:
Use ODBC. Check the func_odbc.conf configuration file.
Leandro
2012/5/5 Jonas Kellens jonas.kell...@telenet.be
Hello,
I notice when upgrading from 1.6.2 to 1.8 that in the menuselect
app_mysql is indicated as deprecated.
If one wants to use the MySQL-command in the dialplan, how to do so if
app_mysql is deprecated ??
Kind regards,
Jonas.
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Re: [asterisk-users] Asterisk 1.6.2 1.8.12
Use ODBC. Check the func_odbc.conf configuration file. Leandro 2012/5/5 Jonas Kellens jonas.kell...@telenet.be ** Hello, I notice when upgrading from 1.6.2 to 1.8 that in the menuselect app_mysql is indicated as deprecated. If one wants to use the MySQL-command in the dialplan, how to do so if app_mysql is deprecated ?? Kind regards, Jonas. -- _ -- 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 -- _ -- 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] realtime config for general settings in sip.conf
2012/5/2 Kamlesh Kumar kamlesh_...@hotmail.com Hi, I need to configure global parameters in sip.conf like rtptimeout, rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time architecture. Please suggest the way to do it. thanks, Kamlesh For what I have discovered, it is not possible. I hope to be wrong, but the sip.conf realtime is limited to peers (or users) registering on the box. It is not suitable even for defining trunks to be used by asterisk. Leandro -- _ -- 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
[asterisk-users] Syntax highlight in emacs for editing extensions.ael
Hello, I was tired of manually aligning ael files in emacs, so I downloaded the .el file on http://www.voip-info.org/wiki/view/EMacs+Asterisk+Syntax+Highlighting Unfortunately there is a problem with switch statement. Do you know of a better .el file or are you good in writing .el files to fix it? Leandro -- _ -- 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] Set SIP peer state busy
Check the command Busy() of the dialplan, it return the busy state at the calling party. Leandro 2012/4/26 Jonas Kellens jonas.kell...@telenet.be ** Hello, can someone please tell me if this is possible and how ? Kind regards, Jonas. On 04/24/2012 12:59 PM, Jonas Kellens wrote: Hello, is there a way to put a certain SIP peer on state busy ? I know you can do this by pressing DND on your IP-phone, but can this state also be set in the dialplan ? Thanks. Jonas. -- _ -- 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 -- _ -- 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] Strange problem on ougoing call
2012/4/25 Olivier CALVANO o.calv...@gmail.com
Sure, sorry for the Confusion ;=)
Server A Trader:
Asterisk Server 1.6.x for call routing only.
IP Adress: 172.16.0.11
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Server B Ipbx
Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
IP Adress: 172.16.0.70
Second IP: 172.16.1.70 (used for phone lan)
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Linksys SPA942 A:
IP Adress: 172.16.1.200
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User01
Linksys SPA942 B:
IP Adress: 172.16.1.220
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User02
On Server A Trader, we have two sip account:
accountname: USER01 for user in group 1
accountname: USER02 for user in group 2
On Server B Ipbx, i use registry:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
for two connection to the Trader Server. Registry is good:
on server A Trader:
trader*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr 2012 15:58:59
On server B Ipbx, i have into my sip.conf after the registry:
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
and in extensions.conf:
[I-User01]
exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
[I-User02]
exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)
When i call with Linksys SPA942 A, i use the context I-User01 and
the call are sent
to SIP account USER01 and No problems.
When i call with Linksys SPA942 B, i use the context I-User02 and
the call are sent
to SIP account USER02 but Server A Trader reject the call
immediatly with this error:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab
Olivier and 906280 is the information that i have on the Linksys
SPA942 B, 906280 is the username used between
best ? hihi
Olivier
Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
Hi,
Lots of mixing and confusing stuff - Can you re-explain the topology you
are
trying to achieve with proper IP addresses and declared sip ext. names.
When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a error:
Somehow it reminds of the same situation I always face when a peer is
declared with the same name as of the dialing one on second server - only
Its just not registered there instead registered on server-1.
So when the call comes in from server-1 to server-2 fromuser=olivier
which
is not registered on server-2 but is declared. Server-2 thinks that this
is
my valid extension but it is not registered with me and so lets
authenticate
this one and here it fails and rejects the call.
BR,
Sammy.
On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has different.
On my second server (phone is connected on this server):
I have in sip.conf:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
i see the registration:
ipbx*CLI sip show registry
Host dnsmgr Username Refresh
Re: [asterisk-users] Advice on Asterisk Conference
1. No, asterisk can act as pbx and as conference server 2. No, just bought a powerful server 3. Not me, sorry 4. You are limited only by the CPU of your server Il giorno 20/apr/2012 19:21, Mitchell Johnson mitch.johns...@gmail.com ha scritto: We're looking into using Asterisk to do our conferencing. Currently we do all our conferencing using Cisco, we have a router with PVDM modules so we can offload the hardware resources. I'm looking for some best practices on how to set it up. 1. DO I need a separate server for the conference server? 2. Do I need to offload the actual conference to a router with PVDM modules. 3. Does anyone have experience with transitioning from Cisco conferencing to Asterisk? 4. How many participants can participate in a conference? Thanks, Mitch -- _ -- 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 -- _ -- 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] Combining multiple SIP providers
I used asterisk with some dialplan customization. It is not difficult. All client asterisk step in the central asterisk to reach the providers. I have a central system to monitor calls, call quality, enforce limits and route versus the best provider. To be sure I have two asterisk servers in active/backup status with heartbeat. Leandro Il giorno 09/apr/2012 16:45, Anita Hall anita.h...@simmortel.com ha scritto: Hi What is the best way to combine multiple SIP providers to achieve 1) Higher concurrency (for eg. 2 providers with 50 concurrent calling limits could be combined to give a limit of 100) 2) Redundancy (use another if one is down) I have a feeling that this will need some SIP Proxy like OpenSIPS but what could be the architecture ? Much thanks! regards, Anita -- _ -- 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 -- _ -- 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] Asterisk ACL
Your understanding of the problem seems incorrect. The problem seems due to the extension not available in your dialplan. Please check carefully in which context the call is placed and if the extension is defined in that context. Maybe it can be useful to define a _X. extension to catch all not defined extensions. Leandro 2012/4/2 Mark Farmer mark.far...@gagenetworks.com Hi ** ** We are trying to accept inbound calls from a SIP provider who sends us calls from various IP’s within a given subnet but they are failing every time with the following message on the console. ** ** chan_sip.c:20006 handle_request_invite: Call from '' to extension 'destination-number' rejected because extension not found ** ** Our understanding is that the deny line blocks every IP and the following permit line then allows calls from the specified subnet but it seems that the peer is never matched when a calls hits the server. It’s almost as if there should be a setting somewhere that we are missing to enable ACL’s. ** ** Can anyone point us in the right direction here please? Is our understanding simply not correct? ** ** In our peer config we have: ** ** host = dynamic type = peer deny = 0.0.0.0/0.0.0.0 permit = xxx.xxx.xxx.xxx/255.255.255.0 context = Test insecure = invite,port ** ** Thanks in advance Mark. ** ** -- _ -- 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 -- _ -- 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] Settings problems of Asterisk as client
Your problem originate from the use of insecure option. Using this option,
the peer is authenticated using the registration ip and not the user and
password.
Leandro
Il giorno 26/mar/2012 05:48, YeungJoe ma_ch1...@hotmail.com ha scritto:
Hello All,
I am Asterisk user, and right now I have some troubles about Asterisk As
Client settings.
Here are my envrionments:
Asterisk-1.8.5.0
---
Server Settings(IP:172.16.70.121)
extensions.conf
[from-internal-200]
exten = _X.,1,Dial(SIP/${EXTEN})
exten = _X.,n,Hangup()
end of extensions.conf/
sip.conf///
[101]
type=friend
username=101
secret=101
host=dynamic
allow=all
context=from-internal-101
[102]
type=friend
username=102
secret=102
host=dynamic
allow=all
context=from-internal-102
[200]
type=friend
username=200
secret=200
host=dynamic
allow=all
context=from-internal-200
end of sip.conf///
---
Client Settings(IP:172.16.70.124:
//extensions.conf//
[from-sip-101]
exten = s,1,Noop(SIP-101)
[from-sip-102]
exten = s,1,Noop(SIP-102)
end of extensions.conf/
/sip.conf//
[general]
register = 101:101@172.16.70.121
register = 102:102@172.16.70.121
[101]
type=peer
username=101
secret=101
insecure=invite,port
host=172.16.70.121
context=from-sip-101
[102]
type=peer
username=102
secret=102
insecure=invite,port
host=172.16.70.121
context=from-sip-102
//end of sip.conf/
---
Right now, I am able to register extensions 101 and 102 to
server(172.16.70.121).
and I can dial from SIP extension 200 to 101 or 102, if I dial 101, it
will be
routed to 101, and 101 is ringing. This is OK. but if I dial 102, it also
be routed 101, I don't know why, because
according to my SIP knowledges it should be routed to 102 as they are
different contexts.
BTW, Client peer is also based on Asterisk.
I am a newbie of SIP, if you need more info I will provide.
Please help! Thanks!
Joe.Yeung
***
*
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[asterisk-users] Routing premature media to the calling channel
Hello, I have a problem with premature media and inband progress audio. I am using the latest 1.8.10.1 and this is the setup: soft phone --- asterisk --- SIP provider The number I call is giving back some hints via inband audio I am not able to ear from the soft phone. They stop on the asterisk and are not routed down the path to the sip phone. The SIP part is simple: soft phone - asterisk: INVITE asterisk - soft phone: TRYING asterisk - provider: INVITE asterisk - soft phone: 180 RINGING provider - asterisk: 183 SESSION PROGRESS provider - asterisk: AUDIO Unfortunately the AUDIO received from the provider by the asterisk box is not sent to the soft phone. I think I have tried every combination of progressinband and prematuremedia, without success. How can I made the audio received from the provider to the asterisk be transmitted to the soft phone? Thank you Leandro -- _ -- 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] Routing premature media to the calling channel
All NAT and firewall problems are already been excluded. All peers are on public IP address and no firewall is active between them. The missing routing of the audio path to the peer has been checked with tcpdump ... nothing is coming out from the asterisk box. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com I assume you have ruled out NAT and firewall issues? Between those two, 99% of the reasons why something may not be routed somewhere correctly are accounted for. If you donapos;t know, your best bet is to take a packet capture or SIP debug on the Asterisk server and find out where that early media is going. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: Hello, I have a problem with premature media and inband progress audio. I am using the latest 1.8.10.1 and this is the setup: soft phone --- asterisk --- SIP provider The number I call is giving back some hints via inband audio I am not able to ear from the soft phone. They stop on the asterisk and are not routed down the path to the sip phone. The SIP part is simple: soft phone - asterisk: INVITE asterisk - soft phone: TRYING asterisk - provider: INVITE asterisk - soft phone: 180 RINGING provider - asterisk: 183 SESSION PROGRESS provider - asterisk: AUDIO Unfortunately the AUDIO received from the provider by the asterisk box is not sent to the soft phone. I think I have tried every combination of progressinband and prematuremedia, without success. How can I made the audio received from the provider to the asterisk be transmitted to the soft phone? Thank you Leandro -- _ -- 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 -- _ -- 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] Routing premature media to the calling channel
The asterisk box has only one interface. I am capturing all the traffic on the box and the only audio traffic is from the provider to the asterisk box. Obviously if I set progressinband=yes, then I get the ringing tone from the asterisk box, but no the audio from the provider I was looking for. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com Are you absolutely sure that nothing is coming out, even on a different interface than the one on which you are capturing? Are you capture on the Asterisk server and not the receiving host? Secondly, are you absolutely positive that something is supposed to be coming out? 183 does not logically imply or mandate backward early media, though 183+SDP is generally used as a convention to indicate that it is about to be sent. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: All NAT and firewall problems are already been excluded. All peers are on public IP address and no firewall is active between them. The missing routing of the audio path to the peer has been checked with tcpdump ... nothing is coming out from the asterisk box. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com I assume you have ruled out NAT and firewall issues? Between those two, 99% of the reasons why something may not be routed somewhere correctly are accounted for. If you donapos;t know, your best bet is to take a packet capture or SIP debug on the Asterisk server and find out where that early media is going. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: Hello, I have a problem with premature media and inband progress audio. I am using the latest 1.8.10.1 and this is the setup: soft phone --- asterisk --- SIP provider The number I call is giving back some hints via inband audio I am not able to ear from the soft phone. They stop on the asterisk and are not routed down the path to the sip phone. The SIP part is simple: soft phone - asterisk: INVITE asterisk - soft phone: TRYING asterisk - provider: INVITE asterisk - soft phone: 180 RINGING provider - asterisk: 183 SESSION PROGRESS provider - asterisk: AUDIO Unfortunately the AUDIO received from the provider by the asterisk box is not sent to the soft phone. I think I have tried every combination of progressinband and prematuremedia, without success. How can I made the audio received from the provider to the asterisk be transmitted to the soft phone? Thank you Leandro -- _ -- 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 -- _ -- 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 -- _ -- 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] Routing premature media to the calling channel
I want to have the early media to pass from the provider down to the soft phone because it contains important information about the call, like Your call cannot go through, please try your call again ... The provider is giving this info via early media, just after the 183 SESSION PROGRESS. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com I think I may have misunderstood your initial question, sorry. You are looking for Asterisk to directly pass through the early media from upstream? Why would it do that? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: The asterisk box has only one interface. I am capturing all the traffic on the box and the only audio traffic is from the provider to the asterisk box. Obviously if I set progressinband=yes, then I get the ringing tone from the asterisk box, but no the audio from the provider I was looking for. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com Are you absolutely sure that nothing is coming out, even on a different interface than the one on which you are capturing? Are you capture on the Asterisk server and not the receiving host? Secondly, are you absolutely positive that something is supposed to be coming out? 183 does not logically imply or mandate backward early media, though 183+SDP is generally used as a convention to indicate that it is about to be sent. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: All NAT and firewall problems are already been excluded. All peers are on public IP address and no firewall is active between them. The missing routing of the audio path to the peer has been checked with tcpdump ... nothing is coming out from the asterisk box. Leandro 2012/3/25 Alex Balashov abalas...@evaristesys.com I assume you have ruled out NAT and firewall issues? Between those two, 99% of the reasons why something may not be routed somewhere correctly are accounted for. If you donapos;t know, your best bet is to take a packet capture or SIP debug on the Asterisk server and find out where that early media is going. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Leandro Dardini ldard...@gmail.com wrote: Hello, I have a problem with premature media and inband progress audio. I am using the latest 1.8.10.1 and this is the setup: soft phone --- asterisk --- SIP provider The number I call is giving back some hints via inband audio I am not able to ear from the soft phone. They stop on the asterisk and are not routed down the path to the sip phone. The SIP part is simple: soft phone - asterisk: INVITE asterisk - soft phone: TRYING asterisk - provider: INVITE asterisk - soft phone: 180 RINGING provider - asterisk: 183 SESSION PROGRESS provider - asterisk: AUDIO Unfortunately the AUDIO received from the provider by the asterisk box is not sent to the soft phone. I think I have tried every combination of progressinband and prematuremedia, without success. How can I made the audio received from the provider to the asterisk be transmitted to the soft phone? Thank you Leandro -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] Routing premature media to the calling channel
Bingo, it was the r option! Thank you Leandro 2012/3/25 isr...@gmail.com Do you have r in your dial string? If yes remove that -Original Message- From: Leandro Dardini ldard...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 25 Mar 2012 11:35:45 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Routing premature media to the calling channel -- _ -- 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 -- _ -- 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 -- _ -- 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] Official numbering plan - where to get?
If you have 10 billing plans from different providers, you have for sure at least almost all the data. Use the prefix from the plans to build your own database of prefixes and destinations. Il giorno 23/mar/2012 05:06, Mikhail Lischuk mlisc...@itx.com.ua ha scritto: ** Is it a problem to parse rates from said 10 providers and create database with all their info? Anyways, speaking of this as a service... I have at least 2 clients, who would love such service: some kind of daily (maybe more often) updated database, which automatically normalizes rates and provides output in parseable format. Maybe even that could include some interactive page, for providers which offer cheaper rates for higher call volumes. But of course 100 Euros/month will be too much for such service. AND some kind of integration with Starbilling will make the whole world happy. BR Don Kelly писал 23.03.2012 01:00: Although I do feel that 100+ Euros/month is more than most of us could manage, I don't think a one-time list is of much value. I would be interested in establishing a database if there was interest from enough users for a modest subscription price. --Don Don Kelly PCF Corp People Come First651 842-1000 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Sent: Thursday, March 22, 2012 5:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Official numbering plan - where to get? I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] Sip insecure
2012/3/22 Zohair Raza engineerzuhairr...@gmail.com Hi, How to allow registered sip users to call without re-authentication insecure =yes/very are deprecated in 1.8 I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch. Please help Regards, Zohair Raza There are other options, like invite and port to be used when you trust the IP of the caller. Leandro -- _ -- 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] Asterisk CDRs
Really interesting finding. From my point of view, it is a good thing. Having spike in cpu load will harm voice quality for sure, but it can hurts if you are relaying on prompt write of cdr records. What cdr backend are you using? Maybe the constant speed you see is the maximal write speed the backend can receive. A silly question ... have you reloaded the cdr module once made the changes? Leandro 2012/3/2 [Digital^Dude] ® millennium@gmail.com I've tried with batch enabled as well as disabled, it seems irrespective of the call burst I send to asterisk. CDR writes at a constant speed, not changing with the call load! On Fri, Mar 2, 2012 at 12:20 PM, Leandro Dardini ldard...@gmail.comwrote: Asterisk can cache cdr records to avoid having to write continuosly in the cdr backend. Writing in bunch instead one at once improves performance. Check the cdr.conf file and disable the option batch if it hurts you. Leandro Il giorno 02/mar/2012 07:24, [Digital^Dude] ® millennium@gmail.com ha scritto: -- _ -- 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 -- _ -- 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 -- _ -- 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] Asterisk CDRs
Asterisk can cache cdr records to avoid having to write continuosly in the cdr backend. Writing in bunch instead one at once improves performance. Check the cdr.conf file and disable the option batch if it hurts you. Leandro Il giorno 02/mar/2012 07:24, [Digital^Dude] ® millennium@gmail.com ha scritto: -- _ -- 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] SER Still recommended for large installs?
I prefer multiple servers sharing the load. All asterisk based. This let me scale up the power of the system just adding more servers. I use asterisk 1.8 realtime with all the data (peers, voicemails, ivr messages and so on) stored in a pair of mysql database with multimaster replication. Phones choose where to connect using SRV records, so I can bring down servers without problems. Leandro 2012/2/17 Jason W. Parks jason.w.pa...@gmail.com: I'm reading some information that recommends using SER / OpenSER for large installation to offload SIP traffic from the Asterisk server. http://www.voip-info.org/wiki/view/Asterisk+at+large However, it looks like the information might be dated. I'm looking at a potential 750 SIP phone and 150 Analog installation, all internal network, PRI trunks, and am trying to nail down an architecture. Opinions? You think I skip the SER box if I'm using 1.8? Thanks! -- I get enough exercise just pushing my luck. -- _ -- 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 -- _ -- 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] Asterisk vs FreeSwitch vs Asterisk + OpenSIP
2012/2/10 Brynjolfur Thorvardsson bi...@itanet.nu: I hope I'm not flogging a dead horse here, but the discussion around the whole scalability issue in Asterisk have opened my eyes to a whole lot of issues, making me increasingly confused! We have a fully functioning and stable installation where we offer PBX services to some 15 small firms (basically medical practices). These are based all over the country, with between 2 and 15 SIP phones each. We have a Web front end where each firm can configure their own queues, menus, forwarding etc. My problem is that my bosses want to expand massively, they are currently talking of at least a tenfold increase in the number of clients. I'm fairly certain our Asterisk server won't be able to handle that. Our current 15 clients all have peak usage at the same time (with 2/3 of all traffic between 8 and 9 in the morning). At peak times, we have 20% CPU load with some 100 concurrent calls and a little under one call/second. I have to solve the scalability problem within a relatively short timeframe so starting from scratch with something new is out of the question. My first thought was to add another Asterisk server and use DUNDi load balancing between the two. But looking around and reading the discussion on this list got me to thinking whether some sort of SIP switch or router/proxy could take some load off the Asterisk server(s). One of my main concerns is to change our current setup as little as possible. It's a mishmash of Asterisk, MySQL, Rails and RAGI/RAMI. The original programmers are no longer available to me and I am still very wet behind the ears when it comes to VOIP. So should I be looking at adding e.g. OpenSIP as a sip proxy to our current setup or adding a second (and then a third and a fourth ...) Asterisk server with DUNDi? Or both? Will adding OpenSIP require a change in the way in which we handle SIP peers or require some major reconfiguration of Asterisk? It seems to me that DUNDi requires minimal configuration changes but I don't really know. Any information and recommendations will be greatly appreciated! Regards Binni There are a lots of solutions to asterisk scalability. Each one with its own pros and cons. If you have several small firms, the easiest path will be to duplicate your installation and share your clients among all the servers. Firm01 to Firm15 will be on server01, Firm16 to Firm25 on server02 and so on... However if you have such big numbers of contemporary calls (the max I recorded on one of my server was 60 active calls), maybe you need to think better to high availability, duplicating each server and putting them in high availability. One other way, the one I prefer is to completely share the load among a bunch of servers using mysql multimaster replication and asterisk realtime. Client's phones will use SRV to locate the best server. This way, you can just increase the capacity adding servers and you are completely fault tolerant. Leandro -- _ -- 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] Asterisk vs FreeSwitch vs Asterisk + OpenSIP
Sorry for the top post, but I am using a silly mail client. I havent talked about ndb tables, just multimaster setup. It is really stable if done with just two mysql servers. I am running a couple of asterisk servers sharing a common cdr and cnam database for at least 3 years without problems. Simple Mysql multimaster replication is really solid and easy to setup and maintain. Dont forget to handle the autoincrement columns with a distinct starting point and a common step, greater than one of course! Leandro Il giorno 10/feb/2012 14:23, Vieri rentor...@yahoo.com ha scritto: --- On Fri, 2/10/12, Leandro Dardini ldard...@gmail.com wrote: mysql multimaster replication and asterisk realtime. Just a word of caution: I've had terrible luck with MySQL NDB tables in a multimaster setup. I'm not a big expert but v.5.0 and 5.1 have given me lots of reliability issues (I lost table data several times). I'd like to try postgresql in a multimaster setup. Realtime with a clustered database is a nice idea but is it reliable? Any long-term success stories? Vieri -- _ -- 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 -- _ -- 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] Does Asterisk permit multiple registrations to the same host?
I can assure you it works. It is important you can set in the [general] section: match_auth_username=yes Leandro 2012/1/19 Frank Church voi...@gmail.com: Does Asterisk permit multiple registrations to the same host? Each registration has a different username and password The purpose is for billing, handling incoming calls is not important, although it will be a bonus. I guess I should also ask the converse, whether the receiving host can accept multiple registrations from the same host to different accounts. I have Googled the issue and the info seems inconclusive. Thanks /voipfc -- _ -- 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 -- _ -- 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] Server-to-server BLF
Me too, an maybe other people on the list are interested in knowing your effort result and maybe appreciate a guide on the topic. Thank you Leandro 2012/1/13 Ronald Cepres rbcep...@gmail.com: Hi Ishfaq, Thanks for your reply. I've already started trying the XMPP method so I can't help you with the AIS method as of the moment. I'll let you know the result of my test. Regards, Ronald On Fri, Jan 6, 2012 at 5:14 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Ronald I took a bit of interest in your problem as I'm going to have to be doing the same thing in a few weeks. oenais is in the yum repositories so you can install from there if using redhat/centos based OS It is also in apt repositories if you're using a debian based OS Let me know how you get on Ish On Thu, 2012-01-05 at 12:07 +0800, Ronald Cepres wrote: Hi Kevin, Thanks for your suggestion. On the website of OpenAIS, it seems that it is not supported anymore and their download links (FTP and SVN) are broken (been trying it for about a month now). Is it still possible to use OpenAIS method? The other solution on the wiki is using XMPP which is for jabber. IMHO, it means that the XMPP solution can't be used on SIP peers, right? Regards, Ronald On Thu, Nov 17, 2011 at 1:01 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 11/16/2011 04:18 AM, Ronald Cepres wrote: Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Here is one way: https://wiki.asterisk.org/wiki/display/AST/Distributed+Device +State+with+AIS There are other methods documented on the wiki as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 -- _ -- 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 -- _ -- 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 stop hacking of my server
2011/12/27 virendra bhati virbh...@gmail.com Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then 1 days. I need your help *CLI log:- * [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '4411 sip:4411@204.152.194.246' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318
Re: [asterisk-users] how to stop hacking of my server
Yes, this is one of my entries: [trunk1] context=fromoutside type=friend deny=0.0.0.0/0.0.0.0 permit=34.2.10.24 qualify=yes 2011/12/27 virendra bhati virbh...@gmail.com Can you give an example how to set these oprion ... On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini ldard...@gmail.comwrote: 2011/12/27 virendra bhati virbh...@gmail.com Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then 1 days. I need your help -- _ -- 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 stop hacking of my server
With deny you'll deny all IP with permit you'll permit only your IP. Yes, it is mandatory to define both deny and permit. Leandro 2011/12/27 virendra bhati virbh...@gmail.com okay, So it is mandatory to define both permit and deny ? if I will update like [trunk1] context=fromoutside type=friend http://0.0.0.0/0.0.0.0 permit=34.2.10.24 qualify=yes So will it be fine or not ? Or it will get rest information from sip.conf general section ? On Tue, Dec 27, 2011 at 2:21 PM, Leandro Dardini ldard...@gmail.comwrote: Yes, this is one of my entries: [trunk1] context=fromoutside type=friend deny=0.0.0.0/0.0.0.0 permit=34.2.10.24 qualify=yes 2011/12/27 virendra bhati virbh...@gmail.com Can you give an example how to set these oprion ... On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini ldard...@gmail.comwrote: 2011/12/27 virendra bhati virbh...@gmail.com Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then 1 days. I need your help -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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 -- _ -- 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] Asterisk HoneyPot
From time to time a similar subject pops up on the list. I'd like to repeat it is extremely dangerous to ban IP based on a suspicious UDP activity. The source IP of an UDP packet can be easily forged, so if you start using fail2ban or other blacklist techniques, it can be very awesome to start sending bogus invite and let you blacklist all major SIP providers... However I am using fail2ban on all my servers :-) Leandro 2011/10/12 Jack Honey Pot j...@asteriskhoneypot.com Hi All, I'm not the first to try to start a VOIP blacklist but currently working on a project for the next 12 hours, hopefully I can get it up soon. What I intend to do is to work with a few reliable Harvester to gather the logs. A simple script to parse it then extract the list of attackers IP, compile them and send them out to the list. If any of you are kind enough to zip and send me a /var/log/asterisk/messages that contain hacker's scan attack, it will be helpful to my research. Do email me at j...@asteriskhoneypot.com . Let me know if you are keen to be a harvester as well.Thanks. Regards, Jackster -- _ -- 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 -- _ -- 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] Problem with multiple sip-peers against the same host
Add match_auth_username=yes in the [general] section of sip.conf Remove from each peer any insecure entry Usually I add also auth, defaultuser and username to the peer definition, but some of these entries are not needed. Leandro 2011/9/23 David Björkevik da...@dynamore.se Dear list, We are switching to a new provider for SIP-trunks. We have 20 numbers, each defined as a separate SIP peer. With the old provider everything works. When switching to the new provider's account data, it only works as long as I only define one of the accounts. If multiple accounts are defined, I can only place outgoing calls on one of them, for the other(s) authentication fails, FORBIDDEN. It is almost like Asterisk is using just one of the defined passwords to authenticate all peers on that host. Any input is very appreciated. Regards David Björkevik, Engineer -- _ -- 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 -- _ -- 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] Problem with multiple sip-peers against the same host
Please check no other peers with insecure entry are registered from the same IP. Asterisk takes some shortcut and try authenticating peers by IP address before authenticating them by username/password. Leandro 2011/9/23 David Björkevik da...@dynamore.se Leandro, Thank you for your input! I tried this and it's still the same. (although I still have _unrelated_ peers with the insecure entry) /David On 2011-09-23 14:24, Leandro Dardini wrote: Add match_auth_username=yes in the [general] section of sip.conf Remove from each peer any insecure entry Usually I add also auth, defaultuser and username to the peer definition, but some of these entries are not needed. Leandro 2011/9/23 David Björkevik da...@dynamore.se mailto:da...@dynamore.se Dear list, We are switching to a new provider for SIP-trunks. We have 20 numbers, each defined as a separate SIP peer. With the old provider everything works. When switching to the new provider's account data, it only works as long as I only define one of the accounts. If multiple accounts are defined, I can only place outgoing calls on one of them, for the other(s) authentication fails, FORBIDDEN. It is almost like Asterisk is using just one of the defined passwords to authenticate all peers on that host. Any input is very appreciated. Regards David Björkevik, Engineer -- _ -- 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 -- _ -- 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 -- David Björkevik, Engineer DYNAmore Nordic AB - http://www.dynamore.se/ Full contact information: http://people.dynamore.se/david Voice: +46 (0)13-23 66 80 On July 1, DYNAmore Nordic AB acquired all of the business of Engineering Research. Read more on www.dynamore.se/dynamore-purchase Note the new @dynamore.se E-mail endings, previous @erab.se endings will work until the end of 2011. -- _ -- 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 -- _ -- 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] Variables error in 1.8.6.0.
2011/9/5 Catalin S. jonsonpla...@gmail.com
Hello,
I have a problem with some variables in 1.8.6.0. I set on extension the
following lines:
exten = h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio,
local_lostpackets)}) ; lost packets by local end **
exten = h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio,
remote_lostpackets)}) ; lost packets by remote end
exten = h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos, audio,
local_jitter)}) ; the Same for jitter
Theoretically this should throw these variables in a table in MySQL but
these values cannot be readed. I think it's a different syntax in 1.8.
I gave this error:
- Executing [h @ macro-special1: 11] Set (SIP/1010-0002, CDR
(LLP) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio,
remote_lostpackets' to CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
remote_lostpackets'
- Executing [h @ macro-special1: 12] Set (SIP/1010-0002, CDR
(PCR) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio, local_jitter' to
CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
local_jitter'
- Executing [h @ macro-special1: 13] Set (SIP/1010-0002, CDR
(ljitt) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio, remote_jitter'
to CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
remote_jitter'
Any idea how I can fix?
Best regards,
Jonson.
--
It is really simple, a patch of few months ago renamed the vars, but forget
to update the documentation. You have to use the source for finding the
new variable names. I paste here the part of the code for your easy
viewing...
{ txcount, INT, { .i4 =
stats.txcount, }, },
{ rxcount, INT, { .i4 =
stats.rxcount, }, },
{ txjitter, DBL, { .d8 =
stats.txjitter, }, },
{ rxjitter, DBL, { .d8 =
stats.rxjitter, }, },
{ remote_maxjitter, DBL, { .d8 =
stats.remote_maxjitter, }, },
{ remote_minjitter, DBL, { .d8 =
stats.remote_minjitter, }, },
{ remote_normdevjitter, DBL, { .d8 =
stats.remote_normdevjitter, }, },
{ remote_stdevjitter,DBL, { .d8 =
stats.remote_stdevjitter, }, },
{ local_maxjitter, DBL, { .d8 =
stats.local_maxjitter, }, },
{ local_minjitter, DBL, { .d8 =
stats.local_minjitter, }, },
{ local_normdevjitter, DBL, { .d8 =
stats.local_normdevjitter, }, },
{ local_stdevjitter, DBL, { .d8 =
stats.local_stdevjitter, }, },
{ txploss, INT, { .i4 =
stats.txploss, }, },
{ rxploss, INT, { .i4 =
stats.rxploss, }, },
{ remote_maxrxploss, DBL, { .d8 =
stats.remote_maxrxploss, }, },
{ remote_minrxploss, DBL, { .d8 =
stats.remote_minrxploss, }, },
{ remote_normdevrxploss, DBL, { .d8 =
stats.remote_normdevrxploss, }, },
{ remote_stdevrxploss, DBL, { .d8 =
stats.remote_stdevrxploss, }, },
{ local_maxrxploss, DBL, { .d8 =
stats.local_maxrxploss, }, },
{ local_minrxploss, DBL, { .d8 =
stats.local_minrxploss, }, },
{ local_normdevrxploss, DBL, { .d8 =
stats.local_normdevrxploss, }, },
{ local_stdevrxploss,DBL, { .d8 =
stats.local_stdevrxploss, }, },
{ rtt, DBL, { .d8 =
stats.rtt, }, },
{ maxrtt,DBL, { .d8 =
stats.maxrtt, }, },
{ minrtt,DBL, { .d8 =
stats.minrtt, }, },
{ normdevrtt,DBL, { .d8 =
stats.normdevrtt, }, },
{ stdevrtt, DBL, { .d8 =
stats.stdevrtt, }, },
{ local_ssrc,INT, { .i4 =
stats.local_ssrc, }, },
{ remote_ssrc, INT, { .i4 =
[asterisk-users] Asterisk SIP authentication against [section] instead of username
Hello, Asterisk seems to try to authenticate incoming INVITE based on the [section] in sip.conf and not the username specified. I just removed the insecure option from my sip.conf requesting every connection to be authenticated. I added the match_auth_username=yes in the [general] section for extra security. To make it work, I have to use the same [section] identifier as username. This is really bad because if multiple provider are giving me the same username, it doesn't work. If I put the following data in sip.conf, it doesn't work. Asterisk return the following error: [2011-07-29 04:55:30] WARNING[9971]: chan_sip.c:13205 check_auth: username mismatch, have GoodProvider, digest has myusername [GoodProvider] username=myusername auth=myusername defaultuser=myusername secret=verydifficultpass type=friend host=pbx.goodprovider.com canreinvite=No dtmfmode=rfc2833 context=from-outside accountcode=GoodProvider disallow=all allow=ulaw If I put the following data in sip.conf, it does work: [myusername] username=myusername auth=myusername defaultuser=myusername secret=verydifficultpass type=friend host=pbx.goodprovider.com canreinvite=No dtmfmode=rfc2833 context=from-outside accountcode=GoodProvider disallow=all allow=ulaw I check the INVITE from the GoodProvider and it is sending myusername Am I doing something wrong or is really asterisk checking the wrong section? Leandro -- _ -- 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
[asterisk-users] Multiple SIP trunks between same pair of asterisk box
Hello, for billing purpose between a multitenant asterisk box and another asterisk, I am in the need to maintain multiple SIP trunks between them. Usually I use insecure=invite,port but I had to remove or the trunks will be selected based on IP address and not with username/password. I had to use the fromuser option or asterisk will try to authenticate the call using the CID and not the username, but this break the outbound CID of the client. Both are asterisk 1.6 Is there any other solution from multiple SIP trunks between two asterisk boxes? Leandro -- _ -- 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] Asterisk-cpu utilization 60 %
2011/5/15 RSCL Mumbai rscl.mum...@gmail.com On Sat, May 14, 2011 at 11:43 AM, Leandro Dardini ldard...@gmail.comwrote: Check if someone is brute forcing your asterisk accounts. It used to happen to me before I install fail2ban. You can easily check the full log of asterisk or with just a tcpdump -i any -n port 5060 or port 4569. Thx for the tcpdump command. Checked, all looks good. Packets coming from trusted domains only. What should be the next step ? Thx Sans Have you tried to restart asterisk? As last chance, install strace and check what is asterisk doing. Get the pid (PID) of the running asterisk and run: strace -p PID -f -F /tmp/strace.log Leave it running for a while then read the strace.log file Leandro -- _ -- 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] Asterisk-cpu utilization 60 %
2011/5/14 RSCL Mumbai rscl.mum...@gmail.com Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans -- Check if someone is brute forcing your asterisk accounts. It used to happen to me before I install fail2ban. You can easily check the full log of asterisk or with just a tcpdump -i any -n port 5060 or port 4569. Leandro -- _ -- 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
