Re: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file
Hello Michael, Thanks a lot for your immediate help. After applying patch MixMonitor started works normally, I can understand that this can be Happen in asterisk 10.4 but as a stable and Long support version 1.8.12.0 this should not happen. I got same error in both version. Anyways this patch solved my problem. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Fri, May 25, 2012 at 3:44 AM, Michael L. Young myo...@acsacc.com wrote: - Original Message - From: Jayesh Labade jayesh.lab...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 24, 2012 4:10:29 PM Subject: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file Hello All, I have installaed asterisk 10.4 in my machine. Now suddenly MixMonitor application starts generating 44 Bytes of Recording file. Is this new tye of Bug? Help me.. Best Regards, Jayesh Labade Jayesh, Is this machine x86? There was a bug that was recently fixed and should show up in 10.5. https://issues.asterisk.org/jira/browse/ASTERISK-19727 Regards, Michael -- _ -- 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
Am 24.05.12 23:46, schrieb bilal ghayyad: Thanks for all for the help and kindly reply. One last point that will help me alot: I am thinking to have 4 Servers running Asterisk and 2 Servers to be for database. The load to be distributed on the 4 Asterisk Servers with ability to be redundant (using any redundancy technique). The 4 Asterisk Servers to take the configuration from the Database Server and actually because there is 2 Database servers, then it will be redundant to each other (in case one database failed, the other will take over). My question is: Is it really possible to have the asterisk configuration in the database server instead of having it in conf files? HOW? I am asking this because what I noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that whatever I do configuration in the GUI, then the configuration will be generated in the conf files, so Asterisk will read from the conf files and not from the database directly. Is it right or I am confused and there is something else? If there is a method to let the configuration to be taken from the database (and not from the configuration), then HOW? Because even in AsteriskNow, the configuration will be generated in a conf files. Special thanks for the advise. Regards Bilal Hi Bilal, Without talking about the Gui, i dont think you will be happy using realtime for this kind of load you were talking about. Asterisk using realtime is much slower than using config files, cause there is a big difference if you ask a database for any kind of data or if you allready have it in memory loaded from a config file. When you want to serve 20k peers with 2000 concurrent calls, even spread over 4 servers you still need all users accessable from every asterisk server, which means to find one peer you allways have to check all entries. i have on my systems around 4000 peers spread about three servers with a sip proxy in front but i still use generated config files cause realtime cant take the load. BTW i have also a mysql cluster setup with two master and two slave servers and the load is spread about them. maybe you should think about doing a split load balancing like server 1 and server 2 can do failover and load balancing but only for the half of your peers and the same for server 3 and server 4. another big point when it comes to realtime with this amount of peers is that asterisk will update the contact data with every register it receives into the database. 20k peers means atleast 6 registrations every second and one register normally means 3 or 4 database request (finding the peer to see if there is a secret set, if yes send a 401 back and when the reregister comes also checks the pass for this peer. if every thing is correct then update the contact data). its not so easy at all to build a system for such a big bunch of peers and calls. best regards stefan -- _ -- 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] Function not Registered??
Hi all, I am running the same Asterisk 1.4.21.2 with the same configuration on all the servers in the region. I got this function called func_devstate which I use to control the lights of the Polycom phones. This module works well for all the Asterisk servers except this one. To get it to work, I basically compile this module together with the others and there is no need to explicitly load it in modules.conf. The problem is when my script uses function DEVSTATE, the Asterisk console shows that it is not registered. However, when I did a module show, it was there. I did restart Asterisk or include it in module.conf but it did not resolve the problem. Do you have any clues why this is happening? Thanks in advance. -- Executing [*1223*1**1900@incoming:78] Set(SIP/1900-08ee1da8, DEVSTATE(Custom:cfalw1900)=INUSE) in new stack [May 25 11:59:46] ERROR[8913]: pbx.c:1564 ast_func_write: Function DEVSTATE not registered /usr/lib/asterisk/modules/func_devstate.so /usr/src/asterisk-1.4.21.2/funcs/.func_devstate.makeopts /usr/src/asterisk-1.4.21.2/funcs/.func_devstate.moduleinfo /usr/src/asterisk-1.4.21.2/funcs/.func_devstate.o.d /usr/src/asterisk-1.4.21.2/funcs/func_devstate.c /usr/src/asterisk-1.4.21.2/funcs/func_devstate.o /usr/src/asterisk-1.4.21.2/funcs/func_devstate.so *CLI module show like func_devstate.so Module Description Use Count func_devstate.so Gets or sets a device state in the dialp 0 1 modules loaded -- _ -- 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] Asterisk Atxfer
Hello, I need to return to the original call leg that I wanted to transfer the call to. in case the destination IVR has put me in a rather long queue. Please suggest a way I can hang up the atxfer leg and return to the first call leg. The hangup parameter in dial app using '*' key works only till the destination is ringing... Please share possible ways I could go about this problem. -- 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
Re: [asterisk-users] Disable All Asterisk Features (blind xfer, disconnect, etc)
You can unload the features module maybe On Wed, May 23, 2012 at 9:18 PM, Eduardo Pimenta e...@akivasoftware.com.brwrote: Hi Guys, is there any way to disable all Asterisk Features? We are having false dtmf detections and randon calls being put on-hold and suspect that dtmf features is the cause. Changing features.conf aparently keeps the default options. Since we dont use it, is there any way to disable it? Thanks, Eduardo -- _ -- 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 MixMonitor starts recording 44 bytes file
On Thursday 24 May 2012, Jayesh Labade wrote: Hello All, I have installaed asterisk 10.4 in my machine. Now suddenly MixMonitor application starts generating 44 Bytes of Recording file. Is this new tye of Bug? Help me.. 44 bytes is very interesting, as this is the canonical length of a .wav header. It sounds as though something is crashing immediately after the recording starts, leaving no data. -- AJS Answers come *after* questions. -- _ -- 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
Hi John; For 20,000 users: Is it better to use Asterisk realtime configuration or to use conf files? I readed the below link but did not understand which GUI that works with asterisk realtime? http://www.freepbx.org/trac/wiki/AsteriskRealtime Regards Bilal My question is: Is it really possible to have the asterisk configuration in the database server instead of having it in conf files? HOW? I am asking this because what I noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that whatever I do configuration in the GUI, then the configuration will be generated in the conf files, so Asterisk will read from the conf files and not from the database directly. Is it right or I am confused and there is something else? If there is a method to let the configuration to be taken from the database (and not from the configuration), then HOW? Because even in AsteriskNow, the configuration will be generated in a conf files. Hi Bilal, You want to look the Asterisk realtime configuration features if you want to run your configuration from a database rather than configuration files. This should point you in the right direction and get you started: http://www.voip-info.org/wiki/view/Asterisk+RealTime It should be noted that if you're wanting to use AsteriskNow (which relies on FreePBX for its gui configuration features), then Asterisk realtime configuration will not work as it is not compatible at this time. Other web gui's might work, but I am not familiar with them. FreePBX's sentiment on the subject is shared here: http://www.freepbx.org/trac/wiki/AsteriskRealtime -John On 05/24/2012 05:46 PM, bilal ghayyad wrote: Thanks for all for the help and kindly reply. One last point that will help me alot: I am thinking to have 4 Servers running Asterisk and 2 Servers to be for database. The load to be distributed on the 4 Asterisk Servers with ability to be redundant (using any redundancy technique). The 4 Asterisk Servers to take the configuration from the Database Server and actually because there is 2 Database servers, then it will be redundant to each other (in case one database failed, the other will take over). My question is: Is it really possible to have the asterisk configuration in the database server instead of having it in conf files? HOW? I am asking this because what I noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that whatever I do configuration in the GUI, then the configuration will be generated in the conf files, so Asterisk will read from the conf files and not from the database directly. Is it right or I am confused and there is something else? If there is a method to let the configuration to be taken from the database (and not from the configuration), then HOW? Because even in AsteriskNow, the configuration will be generated in a conf files. Special thanks for the advise. Regards Bilal - 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? AsteriskNOW is a GUI on top of Asterisk; it does not change the ability of the system to handle call load. Modern versions of Asterisk can easily handle 2,000 simultaneous calls, even with media (non-transcoded) passing through the server. We have a community member who has improved chan_sip in Asterisk 10 (and later) to be able to handle 10,000 simultaneous calls. Handling 20,000 registrations is probably more of a concern for Asterisk at this point; I've never heard of anyone attempting to handle that many on one system. In spite of all this, though, the other advice you've received in this thread is sound: even if a single system can handle the load, doing so is asking for a major problem if that system experiences a failure. You'd be much better off to at least split the load across two machines, both of which should be large enough to handle the entire load when necessary. -- 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 --
Re: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file
- Original Message - From: Jayesh Labade jayesh.lab...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 25, 2012 2:09:58 AM Subject: Re: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file Hello Michael, Thanks a lot for your immediate help. After applying patch MixMonitor started works normally, I can understand that this can be Happen in asterisk 10.4 but as a stable and Long support version 1.8.12.0 this should not happen. I got same error in both version. Anyways this patch solved my problem. Jayesh, Glad to hear that the information helped figure out what was going on and also provided a fix. In the 1.8 line, this has been fixed as well and will be in future releases. In an ideal world, there would be no bugs in software. LTS doesn't mean bug free. It means that it will be supported over a longer period of time which should result in more real world use and more bug fixing resulting in a more stable product with time. Regards, Michael -- _ -- 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] Digium's new Community Support Manager - Rusty Newton
We'd like you all to help us welcome Rusty Newton to Digium's Asterisk development and community support team! Rusty has been with Digium for over five years, starting in the Technical Support department and then moving to a sales position where he assisted customers with Asterisk and Switchvox solutions to their business needs. Prior to joining Digium he spent more than five years in the telecom industry, installing, configuring and maintaining PBXs. A couple of weeks ago he moved into a new role (for him and for Digium), Community Support Manager. In this role he'll be the primary person responsible for ensuring that Digium's community services are providing what the community members need, that the systems are operating properly, and that issues and questions are getting the attention they deserve. He'll be working closely with our Community Director as well, especially for events like AstriCon and others. He works directly with the software development team at Digium, which will allow him to focus almost exclusively on technical issues and discussions. We're quite excited that he has taken on this role and we expect that you will soon see the benefits of his activities across the community! -- 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
[asterisk-users] Dual- or Quad ISDN cards for PCI-X Slots
Hello ISDN Users. I am hit by some frustrations because my Server has only two PCI-X slots and my Eicon Diva 4BRI-8M, which should work fine with Asterisk, is only PCI 2.0 standard and does not fit into the PCI-X slot. Currently I use two AVM Fritz! cards, but while my Xeon 604 2000MHz had a load-average of less then 0.5 it is now increasing to more then 4, if I have 20 VoIP and two ISDN calls. So, the cheap AVM cards have to replaced. Since I do not have currenly the money to buy a Eicon Diva Server 3.0 quad-port, can you recommend me inexpensive Dual- or Quad-Port cards which I could get used on eBay? Note: I have to connect my Alice Box (ADSL2+) and two Vodafone EasyBox 803A (using the Huawei K3765-HV USB-Stick) to it. Thanks, Greetings and nice Day/Evening Michelle Konzack -- # Debian GNU/Linux Consultant ## Development of Intranet and Embedded Systems with Debian GNU/Linux Internet Service Provider, Cloud Computing http://www.itsystems.tamay-dogan.net/ itsystems@tdnet Jabber linux4miche...@jabber.ccc.de Owner Michelle Konzack Gewerbe Strasse 3 Tel office: +49-176-86004575 77694 Kehl Tel mobil: +49-177-9351947 Germany Tel mobil: +33-6-61925193 (France) USt-ID: DE 278 049 239 Linux-User #280138 with the Linux Counter, http://counter.li.org/ signature.pgp Description: Digital signature -- _ -- 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] Huawei K3765-HV with Asterisk?
Hi again, does someone use the USB-Stick Huawei K3765-HV with Asterisk? I have an 4-Port TT-Hub (Cypress Chip) and 4 USB-Sticks with 4 different GSM Provider (Germany, France, Turkey, and Iran) Currently I use the Vodafone Easybox 803 A together with an ISDN connection to my Asterisk Server, an analog telephone (PA710) and an USR Sportster Vi 14.400 Fax-Modem (HylaFax). Grmpf, I was not abele to send faxes from my workstations direktly using Asterisk And then the last question: Does someone know, how the OpenSMS API is working? The EasyBox 803A does support it and I need to get the SMS from the USB- Stick. It is enough, if thex could be fetched and send as mail to me. Thanks, Greetings and nice Day/Evening Michelle Konzack -- # Debian GNU/Linux Consultant ## Development of Intranet and Embedded Systems with Debian GNU/Linux Internet Service Provider, Cloud Computing http://www.itsystems.tamay-dogan.net/ itsystems@tdnet Jabber linux4miche...@jabber.ccc.de Owner Michelle Konzack Gewerbe Strasse 3 Tel office: +49-176-86004575 77694 Kehl Tel mobil: +49-177-9351947 Germany Tel mobil: +33-6-61925193 (France) USt-ID: DE 278 049 239 Linux-User #280138 with the Linux Counter, http://counter.li.org/ signature.pgp Description: Digital signature -- _ -- 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] hangup not detected?
Swift() is an asterisk wrapper around the text-to-speech engine cepstral. Looks like this is a dev issue - I'll start a new thread on the dev mailing list. Justin Killen Senior Programmer / Analyst All American Asphalt 951-736-7600 x 2060 jkil...@allamericanasphalt.commailto:jkil...@allamericanasphalt.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada Sent: Thursday, May 24, 2012 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup not detected? Looks like Swift() (whatever that is) is not returning ? -- _ -- 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] Dual- or Quad ISDN cards for PCI-X Slots
On 05/25/2012 11:10 AM, Michelle Konzack wrote: I am hit by some frustrations because my Server has only two PCI-X slots and my Eicon Diva 4BRI-8M, which should work fine with Asterisk, is only PCI 2.0 standard and does not fit into the PCI-X slot. This does not make sense; PCI 2.0 cards should fit just fine into PCI-X slots. Do you mean PCI-Express instead? That's very different. -- 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
[asterisk-users] URA
Hi Recently our asterisk system stopped beign recognized by URA in others telephones exchanges. What's the troubleshoot steps for this issue? -- Att.* *** -- _ -- 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] Function not Registered??
On 5/25/2012 3:18 AM, Lee, John (Sydney) said: -- Executing [*1223*1**1900@incoming:78] Set(SIP/1900-08ee1da8, DEVSTATE(Custom:cfalw1900)=INUSE) in new stack I use 'Set(DEVICE_STATE(Custom:var)=BUSY)' in my 1.4 dialplans to set device state. 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
Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers
Asterisk Realtime is better for administration. Performance, IMHO is the same issue. I'm not lucky to made large implementations to test these. Regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe http://cipher.pe/web/nuestra-experiencia.html El vie, 25-05-2012 a las 07:06 -0700, bilal ghayyad escribió: Hi John; For 20,000 users: Is it better to use Asterisk realtime configuration or to use conf files? I readed the below link but did not understand which GUI that works with asterisk realtime? http://www.freepbx.org/trac/wiki/AsteriskRealtime Regards Bilal My question is: Is it really possible to have the asterisk configuration in the database server instead of having it in conf files? HOW? I am asking this because what I noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that whatever I do configuration in the GUI, then the configuration will be generated in the conf files, so Asterisk will read from the conf files and not from the database directly. Is it right or I am confused and there is something else? If there is a method to let the configuration to be taken from the database (and not from the configuration), then HOW? Because even in AsteriskNow, the configuration will be generated in a conf files. Hi Bilal, You want to look the Asterisk realtime configuration features if you want to run your configuration from a database rather than configuration files. This should point you in the right direction and get you started: http://www.voip-info.org/wiki/view/Asterisk+RealTime It should be noted that if you're wanting to use AsteriskNow (which relies on FreePBX for its gui configuration features), then Asterisk realtime configuration will not work as it is not compatible at this time. Other web gui's might work, but I am not familiar with them. FreePBX's sentiment on the subject is shared here: http://www.freepbx.org/trac/wiki/AsteriskRealtime -John On 05/24/2012 05:46 PM, bilal ghayyad wrote: Thanks for all for the help and kindly reply. One last point that will help me alot: I am thinking to have 4 Servers running Asterisk and 2 Servers to be for database. The load to be distributed on the 4 Asterisk Servers with ability to be redundant (using any redundancy technique). The 4 Asterisk Servers to take the configuration from the Database Server and actually because there is 2 Database servers, then it will be redundant to each other (in case one database failed, the other will take over). My question is: Is it really possible to have the asterisk configuration in the database server instead of having it in conf files? HOW? I am asking this because what I noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that whatever I do configuration in the GUI, then the configuration will be generated in the conf files, so Asterisk will read from the conf files and not from the database directly. Is it right or I am confused and there is something else? If there is a method to let the configuration to be taken from the database (and not from the configuration), then HOW? Because even in AsteriskNow, the configuration will be generated in a conf files. Special thanks for the advise. Regards Bilal - 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? AsteriskNOW is a GUI on top of Asterisk; it does not change the ability of the system to handle call load. Modern versions of Asterisk can easily handle 2,000 simultaneous calls, even with media (non-transcoded) passing through the server. We have a community member who has improved chan_sip in Asterisk 10 (and later) to be able to handle 10,000 simultaneous calls. Handling 20,000 registrations is probably more of a concern for Asterisk at this point; I've never heard of anyone attempting to handle that many on one system. In spite of all this, though, the other advice you've received in this thread is sound: even if a single system can handle the load, doing so is asking for a major problem if
Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers
Realtime is probably better for administration, but do you want to throw a layer of complication into such a large undertaking? I wouldn’t want 20,000 people screaming at me because MYSQL crapped out. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ing CIP. Alejandro Celi Mariátegui Sent: Friday, May 25, 2012 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers Asterisk Realtime is better for administration. Performance, IMHO is the same issue. I'm not lucky to made large implementations to test these. Regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe http://cipher.pe/web/nuestra-experiencia.html El vie, 25-05-2012 a las 07:06 -0700, bilal ghayyad escribió: Hi John; For 20,000 users: Is it better to use Asterisk realtime configuration or to use conf files? I readed the below link but did not understand which GUI that works with asterisk realtime? http://www.freepbx.org/trac/wiki/AsteriskRealtime Regards Bilal My question is: Is it really possible to have the asterisk configuration in the database server instead of having it in conf files? HOW? I am asking this because what I noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that whatever I do configuration in the GUI, then the configuration will be generated in the conf files, so Asterisk will read from the conf files and not from the database directly. Is it right or I am confused and there is something else? If there is a method to let the configuration to be taken from the database (and not from the configuration), then HOW? Because even in AsteriskNow, the configuration will be generated in a conf files. Hi Bilal, You want to look the Asterisk realtime configuration features if you want to run your configuration from a database rather than configuration files. This should point you in the right direction and get you started: http://www.voip-info.org/wiki/view/Asterisk+RealTime It should be noted that if you're wanting to use AsteriskNow (which relies on FreePBX for its gui configuration features), then Asterisk realtime configuration will not work as it is not compatible at this time. Other web gui's might work, but I am not familiar with them. FreePBX's sentiment on the subject is shared here: http://www.freepbx.org/trac/wiki/AsteriskRealtime -John On 05/24/2012 05:46 PM, bilal ghayyad wrote: Thanks for all for the help and kindly reply. One last point that will help me alot: I am thinking to have 4 Servers running Asterisk and 2 Servers to be for database. The load to be distributed on the 4 Asterisk Servers with ability to be redundant (using any redundancy technique). The 4 Asterisk Servers to take the configuration from the Database Server and actually because there is 2 Database servers, then it will be redundant to each other (in case one database failed, the other will take over). My question is: Is it really possible to have the asterisk configuration in the database server instead of having it in conf files? HOW? I am asking this because what I noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that whatever I do configuration in the GUI, then the configuration will be generated in the conf files, so Asterisk will read from the conf files and not from the database directly. Is it right or I am confused and there is something else? If there is a method to let the configuration to be taken from the database (and not from the configuration), then HOW? Because even in AsteriskNow, the configuration will be generated in a conf files. Special thanks for the advise. Regards Bilal - 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? AsteriskNOW is a GUI on top of Asterisk; it does not change the ability of the system to handle call load. Modern versions of Asterisk can easily handle 2,000 simultaneous calls, even with media (non-transcoded) passing through the server. We have a community member who has improved chan_sip in Asterisk 10 (and
Re: [asterisk-users] Vitelity Setup
Howdy, Since the subject is Viteiy Setup, I don't think this is off topic. My big problem with Vitelity is getting my server to register for incoming calls. I can make outgoing calls just fine. My server says it is registered with Vitelity, but no calls come in. Every attempt to call the number generates an email saying there was a failed call. I am using IAX, not SIP, and that is probably part of the problem. IAX should work better in several ways, but few enough people use it. Vitelity support has been unhelpful so far. My suspicion is that there is a setting they need to make in their server so that calls go to the registered IAX server, instead of looking for a SIP registration, which is not there. Has anyone here worked past such a problem? Was there some special thing I need to ask Vitelity? Thanks, Ralph On 5/24/12, Stephen J Alexander sjalexan...@mpbx.com wrote: If I were troubleshooting this, the next thing I would do is verify connectivity on the relevant ports – more plainly, make sure that there's not a firewall rule with unintended consequences somewhere between your asterisk and your ISP. Otherwise, as Alejandro suggests – check with Vitelity support. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- 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] Vitelity Setup
Is your IAX2 peer registered? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralph Green Sent: Friday, May 25, 2012 4:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vitelity Setup Howdy, Since the subject is Viteiy Setup, I don't think this is off topic. My big problem with Vitelity is getting my server to register for incoming calls. I can make outgoing calls just fine. My server says it is registered with Vitelity, but no calls come in. Every attempt to call the number generates an email saying there was a failed call. I am using IAX, not SIP, and that is probably part of the problem. IAX should work better in several ways, but few enough people use it. Vitelity support has been unhelpful so far. My suspicion is that there is a setting they need to make in their server so that calls go to the registered IAX server, instead of looking for a SIP registration, which is not there. Has anyone here worked past such a problem? Was there some special thing I need to ask Vitelity? Thanks, Ralph On 5/24/12, Stephen J Alexander sjalexan...@mpbx.com wrote: If I were troubleshooting this, the next thing I would do is verify connectivity on the relevant ports - more plainly, make sure that there's not a firewall rule with unintended consequences somewhere between your asterisk and your ISP. Otherwise, as Alejandro suggests - check with Vitelity support. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- 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] Vitelity Setup
If your server says it is registered, that could be part of the problem. Vitelity doesn't use trunk registration, only IP authentication. You should not be using a registration string in your trunk definition. I don't know if it will hurt but it won't help. It sounds like you might have only 1 trunk defined, but you need 2; one for inbound and one for outbound. Their servers for incoming calls and for outgoing calls are separate. If fixing that doesn't do the job, make sure that incoming traffic from Vitelity is correctly routed to your PBX (and that they have the correct IP to send SIP traffic to). Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Fri, May 25, 2012 at 4:12 PM, Ralph Green sira...@gmail.com wrote: Howdy, Since the subject is Viteiy Setup, I don't think this is off topic. My big problem with Vitelity is getting my server to register for incoming calls. I can make outgoing calls just fine. My server says it is registered with Vitelity, but no calls come in. Every attempt to call the number generates an email saying there was a failed call. I am using IAX, not SIP, and that is probably part of the problem. IAX should work better in several ways, but few enough people use it. Vitelity support has been unhelpful so far. My suspicion is that there is a setting they need to make in their server so that calls go to the registered IAX server, instead of looking for a SIP registration, which is not there. Has anyone here worked past such a problem? Was there some special thing I need to ask Vitelity? Thanks, Ralph On 5/24/12, Stephen J Alexander sjalexan...@mpbx.com wrote: If I were troubleshooting this, the next thing I would do is verify connectivity on the relevant ports – more plainly, make sure that there's not a firewall rule with unintended consequences somewhere between your asterisk and your ISP. Otherwise, as Alejandro suggests – check with Vitelity support. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- 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] Loss of RTP stream during DTMF collection
I am using asterisk for voice mail. During DTMF collection Asterisk stop sending any RTP Packets. The gap between two consecutive packets are 4 seconds, which is huge enough to screw up the jitter buffer. When ever asterisk stops to receive DTMF, the RTP stream is cut and we loose audio. I don't have this issue when calling from a SIP phone. I only have this issue when calling from one media gateway to the asterisk box. Any suggestions welcome. Can I play some file in the back while collecting DTMF? Dave -- _ -- 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] Loss of RTP stream during DTMF collection
On 05/25/2012 04:30 PM, Dave George wrote: I am using asterisk for voice mail. During DTMF collection Asterisk stop sending any RTP Packets. The gap between two consecutive packets are 4 seconds, which is huge enough to screw up the jitter buffer. When ever asterisk stops to receive DTMF, the RTP stream is cut and we loose audio. I don't have this issue when calling from a SIP phone. I only have this issue when calling from one media gateway to the asterisk box. Any suggestions welcome. Can I play some file in the back while collecting DTMF? You are missing quite a lot of crucial information required for anyone to help you. First, what version of Asterisk are you using? Second, what type of channel is being used to connect to Asterisk? You mention it works from a SIP phone, but not from a media gateway.. is that gateway also using SIP, or something else? What does 'during DTMF collection' mean? Do you mean after a prompt has been played and the voicemail application is waiting for input, or is this during prompt playback, or something else? Quite some time ago Asterisk was changed to ensure that silence would be sent while an application was running and waiting for input from the caller; if your version is older than this, then that could explain what you are seeing. That's just a mildly-educated guess though. -- 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
Re: [asterisk-users] Loss of RTP stream during DTMF collection
Hi Kevin, I have two asterisk boxes with the same issues. Box 1: asterisk ver 1.4.21.2 Box 2: Asterisk 1.8.7.1 setup: CDMA Phone CDMA Media Gateway WCM sip Asterisk voice mail The calls are SIP Based. DTMF collection is when the user is entering a password for voice mail access or voucher to recharge their account. voice mail: user is prompted for a password. After password is entered I can see asterisk playing the voice mail but no audio is heard on the phone. Other scenario user dials into a voucher menu (Asterisk2billing) and is prompted for a voucher. No audio after the voucher is entered. The CDMA guys did a trace on their end and this is what they explained is happening: The voicemail problem is due to the time stamp jump on the RTP steam sending WCM to BSC. There are about 5 seconds gap between two consecutive RTP packets. It was caused by Asterisk not sending any RTP packet to WCM. How can I enable the option to allow asterisk to maintain the RTP stream during DTMF collection? Thanks, Dave Original Message Subject: Re: [asterisk-users] Loss of RTP stream during DTMF collection From: Kevin P. Fleming kpflem...@digium.com Date: Fri, May 25, 2012 5:38 pm To: asterisk-users@lists.digium.com On 05/25/2012 04:30 PM, Dave George wrote: I am using asterisk for voice mail. During DTMF collection Asterisk stop sending any RTP Packets. The gap between two consecutive packets are 4 seconds, which is huge enough to screw up the jitter buffer. When ever asterisk stops to receive DTMF, the RTP stream is cut and we loose audio. I don't have this issue when calling from a SIP phone. I only have this issue when calling from one media gateway to the asterisk box. Any suggestions welcome. Can I play some file in the back while collecting DTMF? You are missing quite a lot of crucial information required for anyone to help you. First, what version of Asterisk are you using? Second, what type of channel is being used to connect to Asterisk? You mention it works from a SIP phone, but not from a media gateway.. is that gateway also using SIP, or something else? What does 'during DTMF collection' mean? Do you mean after a prompt has been played and the voicemail application is waiting for input, or is this during prompt playback, or something else? Quite some time ago Asterisk was changed to ensure that silence would be sent while an application was running and waiting for input from the caller; if your version is older than this, then that could explain what you are seeing. That's just a mildly-educated guess though. -- 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
[asterisk-users] Help ! Audio not stored .
Hi; In Voicemail.conf If am am using format=h263|gsm ,and i want to store only audio , then it is not storing.In log it shows that video is deposite less then 5 second. If i want to store video and audio both then it will store properly. If am using format=gsm|h263 ,then my Xlite softphone will go to haung. I just want to store audio and video both or some time only audio . 1)Plz guide me which combination of codec will be usefull. 2)Is there is any serial number signifance in format,ie one time if i use as format=h263|gsm and second time i am using format=gsm|h263,why is diffrence come? Thanks Durgesh Mishra Rancore Technologies.-- _ -- 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
