Re: [asterisk-users] Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2

2013-01-10 Thread Leandro Dardini
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

[asterisk-users] Manager event for hint subscribe

2013-01-10 Thread Leandro Dardini
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

Re: [asterisk-users] Detect Low Quality Calls - Realtime

2013-01-08 Thread Leandro Dardini
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

Re: [asterisk-users] Monitor extensions status.

2013-01-08 Thread Leandro Dardini
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

Re: [asterisk-users] Monitor extensions status.

2013-01-08 Thread Leandro Dardini
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

Re: [asterisk-users] Limit registration concurrency per friend

2013-01-05 Thread Leandro Dardini
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

Re: [asterisk-users] RES: Auto ban IP addresses

2013-01-03 Thread Leandro Dardini
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

Re: [asterisk-users] DAHDI: How to know since when it is used? How to shutdown after max time?

2013-01-03 Thread Leandro Dardini
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

Re: [asterisk-users] Verizon SIP trunking Field Trial

2013-01-03 Thread Leandro Dardini
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

Re: [asterisk-users] new user help required to build voice recorder with asterisk

2013-01-02 Thread Leandro Dardini
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

Re: [asterisk-users] new user help required to build voice recorder with asterisk

2012-12-31 Thread Leandro Dardini
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

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Leandro Dardini
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

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Leandro Dardini
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

Re: [asterisk-users] disconnect supervision

2012-12-11 Thread Leandro Dardini
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,

Re: [asterisk-users] - configure ring group

2012-12-06 Thread Leandro Dardini
** ** 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

Re: [asterisk-users] CDR - Freepbx - Safe to add primary key to table ?

2012-12-06 Thread Leandro Dardini
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

Re: [asterisk-users] CDR - Freepbx - Safe to add primary key to table ?

2012-12-06 Thread Leandro Dardini
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

Re: [asterisk-users] - configure ring group

2012-12-05 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk not starting (illegal instruction core dumped)

2012-11-27 Thread Leandro Dardini
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

Re: [asterisk-users] leading ghost 0

2012-11-21 Thread Leandro Dardini
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

Re: [asterisk-users] leading ghost 0

2012-11-20 Thread Leandro Dardini
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 =

Re: [asterisk-users] leading ghost 0

2012-11-20 Thread Leandro Dardini
... 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

Re: [asterisk-users] leading ghost 0

2012-11-20 Thread Leandro Dardini
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

[asterisk-users] Extension hints, which info available?

2012-09-29 Thread Leandro Dardini
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

Re: [asterisk-users] Realtime Hints

2012-09-25 Thread Leandro Dardini
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

Re: [asterisk-users] Realtime Hints

2012-09-25 Thread Leandro Dardini
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

[asterisk-users] Realtime Hints

2012-09-24 Thread Leandro Dardini
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

Re: [asterisk-users] graceful restart

2012-08-19 Thread Leandro Dardini
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

Re: [asterisk-users] Segmenting A Configration File

2012-08-12 Thread Leandro Dardini
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

Re: [asterisk-users] Segmenting A Configration File

2012-08-11 Thread Leandro Dardini
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

Re: [asterisk-users] Multiple channel for SIP users

2012-08-11 Thread Leandro Dardini
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

Re: [asterisk-users] IAX with two asterisk boxes

2012-08-09 Thread Leandro Dardini
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

Re: [asterisk-users] Suggestion of Server Specifications for Asterisk

2012-08-08 Thread Leandro Dardini
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

Re: [asterisk-users] Suggestion of Server Specifications for Asterisk

2012-08-04 Thread Leandro Dardini
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

Re: [asterisk-users] Suggestion of Server Specifications for Asterisk

2012-08-04 Thread Leandro Dardini
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

Re: [asterisk-users] Suggestion of Server Specifications for Asterisk

2012-08-04 Thread Leandro Dardini
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

Re: [asterisk-users] Suggestion of Server Specifications for Asterisk

2012-08-04 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

2012-08-03 Thread Leandro Dardini
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

Re: [asterisk-users] asterisk realtime database structure

2012-08-03 Thread Leandro Dardini
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

Re: [asterisk-users] asterisk realtime database structure

2012-08-03 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

2012-08-03 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

2012-08-03 Thread Leandro Dardini
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

Re: [asterisk-users] Multi-Tenant PBX with Asterisk

2012-07-31 Thread Leandro Dardini
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

Re: [asterisk-users] Multi-Tenant PBX with Asterisk

2012-07-30 Thread Leandro Dardini
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.

Re: [asterisk-users] Multi-Tenant PBX with Asterisk

2012-07-30 Thread Leandro Dardini
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

Re: [asterisk-users] Multiple DID for SIP Trunk

2012-07-28 Thread Leandro Dardini
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

[asterisk-users] Call ID of the second call leg

2012-07-27 Thread Leandro Dardini
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

[asterisk-users] Call ID of the second call leg

2012-07-27 Thread Leandro Dardini
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

[asterisk-users] Call ID of the second call leg

2012-07-26 Thread Leandro Dardini
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

Re: [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk

2012-07-23 Thread Leandro Dardini
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

Re: [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk

2012-07-23 Thread Leandro Dardini
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

Re: [asterisk-users] trouble with asterisk behind router

2012-07-13 Thread Leandro Dardini
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

Re: [asterisk-users] trouble with asterisk behind router

2012-07-13 Thread Leandro Dardini
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

Re: [asterisk-users] port 5060 is blocked by ISP

2012-07-01 Thread Leandro Dardini
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

Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers?

2012-05-23 Thread Leandro Dardini
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

[asterisk-users] Jumping inside a macro with AEL

2012-05-20 Thread Leandro Dardini
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);

Re: [asterisk-users] Asterisk 1.6.2 1.8.12

2012-05-06 Thread Leandro Dardini
,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

Re: [asterisk-users] Asterisk 1.6.2 1.8.12

2012-05-05 Thread Leandro Dardini
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

Re: [asterisk-users] realtime config for general settings in sip.conf

2012-05-02 Thread Leandro Dardini
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

[asterisk-users] Syntax highlight in emacs for editing extensions.ael

2012-05-01 Thread Leandro Dardini
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

Re: [asterisk-users] Set SIP peer state busy

2012-04-26 Thread Leandro Dardini
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:

Re: [asterisk-users] Strange problem on ougoing call

2012-04-25 Thread Leandro Dardini
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

Re: [asterisk-users] Advice on Asterisk Conference

2012-04-20 Thread Leandro Dardini
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

Re: [asterisk-users] Combining multiple SIP providers

2012-04-09 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk ACL

2012-04-02 Thread Leandro Dardini
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

Re: [asterisk-users] Settings problems of Asterisk as client

2012-03-26 Thread Leandro Dardini
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

[asterisk-users] Routing premature media to the calling channel

2012-03-25 Thread Leandro Dardini
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

Re: [asterisk-users] Routing premature media to the calling channel

2012-03-25 Thread Leandro Dardini
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

Re: [asterisk-users] Routing premature media to the calling channel

2012-03-25 Thread Leandro Dardini
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

Re: [asterisk-users] Routing premature media to the calling channel

2012-03-25 Thread Leandro Dardini
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

Re: [asterisk-users] Routing premature media to the calling channel

2012-03-25 Thread Leandro Dardini
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

Re: [asterisk-users] Official numbering plan - where to get?

2012-03-23 Thread Leandro Dardini
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

Re: [asterisk-users] Sip insecure

2012-03-22 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk CDRs

2012-03-02 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk CDRs

2012-03-01 Thread Leandro Dardini
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

Re: [asterisk-users] SER Still recommended for large installs?

2012-02-17 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP

2012-02-10 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP

2012-02-10 Thread Leandro Dardini
/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

Re: [asterisk-users] Does Asterisk permit multiple registrations to the same host?

2012-01-19 Thread Leandro Dardini
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

Re: [asterisk-users] Server-to-server BLF

2012-01-12 Thread Leandro Dardini
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

Re: [asterisk-users] how to stop hacking of my server

2011-12-27 Thread Leandro Dardini
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

Re: [asterisk-users] how to stop hacking of my server

2011-12-27 Thread Leandro Dardini
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

Re: [asterisk-users] how to stop hacking of my server

2011-12-27 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk HoneyPot

2011-10-13 Thread Leandro Dardini
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

Re: [asterisk-users] Problem with multiple sip-peers against the same host

2011-09-23 Thread Leandro Dardini
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

Re: [asterisk-users] Problem with multiple sip-peers against the same host

2011-09-23 Thread Leandro Dardini
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

Re: [asterisk-users] Variables error in 1.8.6.0.

2011-09-05 Thread Leandro Dardini
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) =

[asterisk-users] Asterisk SIP authentication against [section] instead of username

2011-07-29 Thread Leandro Dardini
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

[asterisk-users] Multiple SIP trunks between same pair of asterisk box

2011-07-20 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk-cpu utilization 60 %

2011-05-15 Thread Leandro Dardini
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

Re: [asterisk-users] Asterisk-cpu utilization 60 %

2011-05-14 Thread Leandro Dardini
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

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