Re: [asterisk-users] mixmonitor extension
for the record. info about opus from Lorenzo Mniero (author of Opus patch for asterisk) with his permission --cite-- Opus is just a codec. In order to save an audio file using Opus, you need a container, which for Opus is OGG. Asterisk supports OGG, but I think it is implemented to only dump Vorbis audio, and so the existing module would need to be extended to support Opus as well. I haven't checked how complex this could be, to be honest, so I have no idea about how much effort would be needed for this. Right now we don't need it, so I really can't say if and when we'll start working on this. Lorenzo --cite-- Dne 24.1.2014 10:42, Gareth Blades napsal(a): On 23/01/14 23:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? what about patch for Opus? uncle google doesnt know MP3 is only supported for reading not writing. Its a patent issue as Asterisk cannot distribute the software to write to mp3 under its own license. Its a similar issue with Opus as the codec is covered by a couple of patents in the USA. What most people do is use MixMonitor to record to .wav (alaw) and then in the 'h' extension call a program which runs a background task to convert the .wav file to whatever format they wish. Thats what we do but we actually use the Monitor application and we end up with both legs of the call and multiple sets of recordings if people pause and unpause. We then move these files off to a different server when they get mixed and converted to mp3 and then emailed out to our customers. We do it this way to reduce the load on the Asterisk boxes but also keep all the call recordings in a central location. -- --- Marek Cervenka === -- _ -- 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] AsteriskNOW with AX1600P card
Hi all! I'm new with telephony cards and DAHDI drivers. I have installed Asterisk NOW 3.0.0 and update to Asterisk 11.7.0, modules are update too. I'm following the installation guide of Atcom [1] for AX1600P analogic card, modules are loaded [root@pbx ~]# lsmod | grep -E hisax|netjet|dahdi netjet 14618 0 isdnhdlc4523 1 netjet mISDNipac 33989 1 netjet mISDN_core 73118 3 netjet,mISDNipac hisax 410162 0 isdn 119265 1 hisax dahdi_transcode 5240 1 wctc4xxp dahdi_voicebus 49368 2 wctdm24xxp,wcte12xp dahdi 207790 15 xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wcaxx,wctdm24xxp,wcte11xp,wct1xxp,wcte13xp,wcte12xp,dahdi_voicebus,wcte43x,wct4xxp,oct612x crc_ccitt 1369 4 isdnhdlc,hisax,wctdm24xxp,dahdi and there is the output of lspci command: [root@pbx ~]# lspci -vv -s 02:03.0 02:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device b300:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- INTx- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 12 Region 0: I/O ports at a000 [size=256] Region 1: Memory at fb001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0+,D1-,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel modules: hisax, netjet When I run dahdi_genconf I get the following error: [root@pbx ~]# dahdi_genconf Empty configuration -- no spans /usr/sbin/dahdi_span_types: línea 158: cd: /sys/bus/dahdi_devices/devices/*: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/location: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/hardware_id: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/spantype: No existe el fichero o el directorio Empty configuration -- no spans /usr/sbin/dahdi_span_assignments: línea 183: cd: /sys/bus/dahdi_devices/devices/*: No existe el fichero o el directorio cut: /sys/bus/dahdi_devices/devices/*/spantype: No existe el fichero o el directorio Empty configuration -- no spans Empty configuration -- no spans Which is the better way to install Asterisk NOW? I need install with AX card inserted before? 1 - http://www.atcom.cn/cn/download/cards/ax1600p/AX-1600P-AXE1600P-Centos6.0-Dahdi-User%20Manual-V1.1-EN.pdf Thanks for all. Regards. Fernando. -- _ -- 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] AsteriskNOW with AX1600P card
On Monday 27 January 2014, Fernando Pizarro wrote: Hi all! I'm new with telephony cards and DAHDI drivers. I have installed Asterisk NOW 3.0.0 and update to Asterisk 11.7.0, modules are update too. I'm following the installation guide of Atcom [1] for AX1600P analogic card, modules are loaded [root@pbx ~]# lsmod | grep -E hisax|netjet|dahdi netjet 14618 0 isdnhdlc4523 1 netjet mISDNipac 33989 1 netjet mISDN_core 73118 3 netjet,mISDNipac hisax 410162 0 isdn 119265 1 hisax dahdi_transcode 5240 1 wctc4xxp dahdi_voicebus 49368 2 wctdm24xxp,wcte12xp dahdi 207790 15 xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wcaxx,wctdm24xxp,wcte11xp,wct1xxp,w cte13xp,wcte12xp,dahdi_voicebus,wcte43x,wct4xxp,oct612x crc_ccitt 1369 4 isdnhdlc,hisax,wctdm24xxp,dahdi and there is the output of lspci command: [root@pbx ~]# lspci -vv -s 02:03.0 02:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device b300:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- INTx- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 12 Region 0: I/O ports at a000 [size=256] Region 1: Memory at fb001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0+,D1-,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel modules: hisax, netjet When I run dahdi_genconf I get the following error: [root@pbx ~]# dahdi_genconf Empty configuration -- no spans /usr/sbin/dahdi_span_types: línea 158: cd: /sys/bus/dahdi_devices/devices/*: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/location: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/hardware_id: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/spantype: No existe el fichero o el directorio Empty configuration -- no spans /usr/sbin/dahdi_span_assignments: línea 183: cd: /sys/bus/dahdi_devices/devices/*: No existe el fichero o el directorio cut: /sys/bus/dahdi_devices/devices/*/spantype: No existe el fichero o el directorio Empty configuration -- no spans Empty configuration -- no spans Which is the better way to install Asterisk NOW? I need install with AX card inserted before? That looks like the result of a part-finished upgrade. You probably need to rebuild DAHDI *while the card is installed in the machine*, so it can detect the hardware properly; and then after doing that, you may need to rebuild Asterisk again. -- 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] AsteriskNOW with AX1600P card
El 27/01/14 13:26, A J Stiles escribió: On Monday 27 January 2014, Fernando Pizarro wrote: Hi all! I'm new with telephony cards and DAHDI drivers. I have installed Asterisk NOW 3.0.0 and update to Asterisk 11.7.0, modules are update too. I'm following the installation guide of Atcom [1] for AX1600P analogic card, modules are loaded [root@pbx ~]# lsmod | grep -E hisax|netjet|dahdi netjet 14618 0 isdnhdlc4523 1 netjet mISDNipac 33989 1 netjet mISDN_core 73118 3 netjet,mISDNipac hisax 410162 0 isdn 119265 1 hisax dahdi_transcode 5240 1 wctc4xxp dahdi_voicebus 49368 2 wctdm24xxp,wcte12xp dahdi 207790 15 xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wcaxx,wctdm24xxp,wcte11xp,wct1xxp,w cte13xp,wcte12xp,dahdi_voicebus,wcte43x,wct4xxp,oct612x crc_ccitt 1369 4 isdnhdlc,hisax,wctdm24xxp,dahdi and there is the output of lspci command: [root@pbx ~]# lspci -vv -s 02:03.0 02:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device b300:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- INTx- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 12 Region 0: I/O ports at a000 [size=256] Region 1: Memory at fb001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0+,D1-,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel modules: hisax, netjet When I run dahdi_genconf I get the following error: [root@pbx ~]# dahdi_genconf Empty configuration -- no spans /usr/sbin/dahdi_span_types: línea 158: cd: /sys/bus/dahdi_devices/devices/*: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/location: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/hardware_id: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/spantype: No existe el fichero o el directorio Empty configuration -- no spans /usr/sbin/dahdi_span_assignments: línea 183: cd: /sys/bus/dahdi_devices/devices/*: No existe el fichero o el directorio cut: /sys/bus/dahdi_devices/devices/*/spantype: No existe el fichero o el directorio Empty configuration -- no spans Empty configuration -- no spans Which is the better way to install Asterisk NOW? I need install with AX card inserted before? That looks like the result of a part-finished upgrade. You probably need to rebuild DAHDI *while the card is installed in the machine*, so it can detect the hardware properly; and then after doing that, you may need to rebuild Asterisk again. Hi, I'm going to try reinstall with card installed. Regards. Fernando. -- _ -- 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] IOPS required by Asterisk for Call Recording
Thanks for response. How do I derive the requirement? I need to size IO system to record multiple calls concurrently. I ran test with following configuration Quad Core Xeon with 4GB RAM 250GB SATA disk (No RAID) Linux (CentOS 5.9) Asterisk 1.8.20 I failed to record more than 80 calls. If I run test with simple IVR, I achieved 400+ calls with same server. So write seem to be an issue. Is there any way to tune / optimize / configure for better write performance? I am not sure if I need to post this query on developers list? Please guide... Regards Amit Patkar Message: 1 Date: Fri, 24 Jan 2014 11:46:39 -0400 From: Mikeispbuil...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IOPS required by Asterisk for Call Recording Message-ID:52e28adf.8020...@gmail.com Content-Type: text/plain; charset=iso-8859-1 On 14-01-24 11:16 AM, Amit wrote: If I assume that Asterisk will write data on disk every second for each call, I will need disk array to support minimum of 500 IOPS. Where as if Asterisk push data every 2 seconds, I can deal with array supporting 250 IOPS. But if I assume that Asterisk will write data on disk for every RTP packet received, as and when received, I will need disk IO system with approx 25000 IOPS assuming 20 ms RTP packet. You're assuming that asterisk will perform an fsync() after each write. If asterisk writes without an fsync after each write, then the OS will schedule writes intelligently based on RAM/disk IO available rather than scheduling each one as a separate write. Looking at the code for ast_writestream() there doesn't appear to be an fsync() type call after each write, but someone more familiar with the internals of Asterisk would be better able to verify that. -- Looking for (employment|contract) work in the Internet industry, preferably working remotely. Building / Supporting the net since 2400 baud was the hot thing. Ask for a resume! ispbuil...@gmail.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20140124/590804b7/attachment-0001.html -- Message: 2 Date: Fri, 24 Jan 2014 16:34:17 + From: A J Stiles asterisk_l...@earthshod.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IOPS required by Asterisk for Call Recording Message-ID: 201401241634.17311.asterisk_l...@earthshod.co.uk Content-Type: Text/Plain; charset=iso-8859-6 On Friday 24 January 2014, Mike wrote: On 14-01-24 11:16 AM, Amit wrote: If I assume that Asterisk will write data on disk every second for each call, I will need disk array to support minimum of 500 IOPS. Where as if Asterisk push data every 2 seconds, I can deal with array supporting 250 IOPS. But if I assume that Asterisk will write data on disk for every RTP packet received, as and when received, I will need disk IO system with approx 25000 IOPS assuming 20 ms RTP packet. You're assuming that asterisk will perform an fsync() after each write. If asterisk writes without an fsync after each write, then the OS will schedule writes intelligently based on RAM/disk IO available rather than scheduling each one as a separate write. Looking at the code for ast_writestream() there doesn't appear to be an fsync() type call after each write, but someone more familiar with the internals of Asterisk would be better able to verify that. If you are running on Linux, don't forget that Linux's default behaviour is to cache all disk writes until the machine is rebooted or the RAM is needed for something else, and service read operations from the cache. In fact, it's entirely possible for a temporary file to be written, read and deleted without ever going anywhere near a molecule of oxide. Solaris has the opposite default caching strategy -- it assumes the worst about filesystem integrity, and write operations block until decaching and verifying have finished. -- AJS Answers come *after* questions. *Thanks Regards,* Amit Patkar On 1/24/2014 8:46 PM, Amit wrote: Hi What are the disk IOPS required for Asterisk call recording? I am trying to find out number of disks required in RAID array to record 500 calls. Is there any formula to calculate IOPS required by Asterisk call recording? This will help me to find IOPS for different scale. If I assume that Asterisk will write data on disk every second for each call, I will need disk array to support minimum of 500 IOPS. Where as if Asterisk push data every 2 seconds, I can deal with array supporting 250 IOPS. But if I assume that Asterisk will write data on disk for every RTP packet received, as and when received, I will need disk IO system with approx 25000 IOPS assuming 20 ms RTP packet. Please assist me on this requirement. *Thanks Regards,* Amit Patkar --
Re: [asterisk-users] IOPS required by Asterisk for Call Recording
On 25/1/14 5:26 am, Amit wrote: How do I derive the requirement? I need to size IO system to record multiple calls concurrently. I suspect this might be your problem: 250GB SATA disk (No RAID) Is there any way to tune / optimize / configure for better write performance? Perhaps consider recording to a ramdisk first, then periodically write out completed files to HDD at your leisure (e.g. during slack periods)? Or, given the relatively low cost of 250GB SSDs these days, swap out the spinning disc for an SSD. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] IOPS required by Asterisk for Call Recording
On 14-01-25 01:26 AM, Amit wrote: 250GB SATA disk (No RAID) If you care enough to record the calls, you should care enough to get some fast and redundant storage. SSDs would be best, 15K SAS drives second choice. Even a good RAID10 of SATA drives would help a lot. A RAID card with battery backed cache would be helpful as well. -- Looking for (employment|contract) work in the Internet industry, preferably working remotely. Building / Supporting the net since 2400 baud was the hot thing. Ask for a resume! ispbuil...@gmail.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
Re: [asterisk-users] IOPS required by Asterisk for Call Recording
On 25-01-14 06:26, Amit wrote: Thanks for response. How do I derive the requirement? I need to size IO system to record multiple calls concurrently. I'm not aware of 400+ calls being recorded succesfully on an Asterisk box. If there is it probably has tons of RAM, enterprise grade SSDs or 15K RPM FC/SAS drives in a battery backed RAID setup or a fast SAN saving the calls in native format (via a tmpfs) with the transcoding probably done on another box. I ran test with following configuration Quad Core Xeon with 4GB RAM Add more RAM and much much more if you are going to use tmpfs. 250GB SATA disk (No RAID) Well you get the performance you pay for. CentOS comes with various utilities that allow you to analyze that. Linux (CentOS 5.9) Imo CentOS 6.5 (x86_64) has better performance. Asterisk 1.8.20 In 9 months Asterisk 1.8 will only get security fixes. I would use Asterisk 11. It will get regular bug fixes for a much longer time. I failed to record more than 80 calls. Hardly surprising. If I run test with simple IVR, I achieved 400+ calls with same server. A simple IVR is not the same as call recording. The comparison makes as much sense as saying that copying to /dev/null is faster than to a disk. So write seem to be an issue. Is there any way to tune / optimize / configure for better write performance? I am not sure if I need to post this query on developers list? Please guide... No, this is a user question and does not belong on the developer list. Since you seem to work for a call center business perhaps investigate a commercial solution like Orecx (I have no affiliation): http://www.orecx.com/OrecX-for-Asterisk.php HTH, Patrick -- _ -- 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] Application Queue context that calls the extensions
Hello! I wonder what the default context that the Queue application uses to call extensions. If there is a possibility to change this into a context created by me possible? Would you like to get this load value to variables before calling the extension. tks, 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
Re: [asterisk-users] IOPS required by Asterisk for Call Recording
On 01/24/2014 11:26 PM, Amit wrote: Thanks for response. How do I derive the requirement? I need to size IO system to record multiple calls concurrently. I ran test with following configuration Quad Core Xeon with 4GB RAM 250GB SATA disk (No RAID) Linux (CentOS 5.9) Asterisk 1.8.20 I'd suggest testing your system while monitoring with top and iotop (which should be a yum install away). That should show you your bottlenecks. It looks to me like Asterisk doesn't do compression until the call is ended, so recording to a compressed format would actually increase IO load (write, read and compress, write compressed data). -- Daniel Taylor -- _ -- 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] IOPS required by Asterisk for Call Recording
Can you get a reading of the total number of I/Os during your test? Peak IOPS? That might tell you very quickly about the storage pattern that Asterisk uses. Can you configure a RAM drive to see if disk is really the bottleneck. May need to add some more RAM memory to your configuration. What is your network capacity? Usually one can write faster than the network can deliver - just to make sure that you are chasing the right bottleneck. What happens at 80 calls to tell you that you have run out of IOPS? Sorry for more questions than answers. Ron On 25/01/2014 12:26 AM, Amit wrote: Thanks for response. How do I derive the requirement? I need to size IO system to record multiple calls concurrently. I ran test with following configuration Quad Core Xeon with 4GB RAM 250GB SATA disk (No RAID) Linux (CentOS 5.9) Asterisk 1.8.20 I failed to record more than 80 calls. If I run test with simple IVR, I achieved 400+ calls with same server. So write seem to be an issue. Is there any way to tune / optimize / configure for better write performance? I am not sure if I need to post this query on developers list? Please guide... Regards Amit Patkar Message: 1 Date: Fri, 24 Jan 2014 11:46:39 -0400 From: Mikeispbuil...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IOPS required by Asterisk for Call Recording Message-ID:52e28adf.8020...@gmail.com Content-Type: text/plain; charset=iso-8859-1 On 14-01-24 11:16 AM, Amit wrote: If I assume that Asterisk will write data on disk every second for each call, I will need disk array to support minimum of 500 IOPS. Where as if Asterisk push data every 2 seconds, I can deal with array supporting 250 IOPS. But if I assume that Asterisk will write data on disk for every RTP packet received, as and when received, I will need disk IO system with approx 25000 IOPS assuming 20 ms RTP packet. You're assuming that asterisk will perform an fsync() after each write. If asterisk writes without an fsync after each write, then the OS will schedule writes intelligently based on RAM/disk IO available rather than scheduling each one as a separate write. Looking at the code for ast_writestream() there doesn't appear to be an fsync() type call after each write, but someone more familiar with the internals of Asterisk would be better able to verify that. -- 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
Re: [asterisk-users] Application Queue context that calls the extensions
app_queue dials Channels and not extensions unless your adding them to the queue as members using a local channel. I believe you can call Macro's and Gosubs from app_queue to set variables before the channels are bridged. On Mon, Jan 27, 2014 at 11:17 AM, Eduardo Leones edua...@ypytecnologia.com.br wrote: Hello! I wonder what the default context that the Queue application uses to call extensions. If there is a possibility to change this into a context created by me possible? Would you like to get this load value to variables before calling the extension. tks, 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 -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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] AsteriskNOW with AX1600P card
El 27/01/14 13:44, Fernando Pizarro escribió: El 27/01/14 13:26, A J Stiles escribió: On Monday 27 January 2014, Fernando Pizarro wrote: Hi all! I'm new with telephony cards and DAHDI drivers. I have installed Asterisk NOW 3.0.0 and update to Asterisk 11.7.0, modules are update too. I'm following the installation guide of Atcom [1] for AX1600P analogic card, modules are loaded [root@pbx ~]# lsmod | grep -E hisax|netjet|dahdi netjet 14618 0 isdnhdlc4523 1 netjet mISDNipac 33989 1 netjet mISDN_core 73118 3 netjet,mISDNipac hisax 410162 0 isdn 119265 1 hisax dahdi_transcode 5240 1 wctc4xxp dahdi_voicebus 49368 2 wctdm24xxp,wcte12xp dahdi 207790 15 xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wcaxx,wctdm24xxp,wcte11xp,wct1xxp,w cte13xp,wcte12xp,dahdi_voicebus,wcte43x,wct4xxp,oct612x crc_ccitt 1369 4 isdnhdlc,hisax,wctdm24xxp,dahdi and there is the output of lspci command: [root@pbx ~]# lspci -vv -s 02:03.0 02:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device b300:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- INTx- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 12 Region 0: I/O ports at a000 [size=256] Region 1: Memory at fb001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0+,D1-,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel modules: hisax, netjet When I run dahdi_genconf I get the following error: [root@pbx ~]# dahdi_genconf Empty configuration -- no spans /usr/sbin/dahdi_span_types: línea 158: cd: /sys/bus/dahdi_devices/devices/*: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/location: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/hardware_id: No existe el fichero o el directorio cat: /sys/bus/dahdi_devices/devices/*/spantype: No existe el fichero o el directorio Empty configuration -- no spans /usr/sbin/dahdi_span_assignments: línea 183: cd: /sys/bus/dahdi_devices/devices/*: No existe el fichero o el directorio cut: /sys/bus/dahdi_devices/devices/*/spantype: No existe el fichero o el directorio Empty configuration -- no spans Empty configuration -- no spans Which is the better way to install Asterisk NOW? I need install with AX card inserted before? That looks like the result of a part-finished upgrade. You probably need to rebuild DAHDI *while the card is installed in the machine*, so it can detect the hardware properly; and then after doing that, you may need to rebuild Asterisk again. Hi, I'm going to try reinstall with card installed. Regards. Fernando. Hi again! I'm tried reinstall with card installed but I get the same error when I run dahdi_genconf. I noticed the leds are on now and dahdi_scan doesn't show anything while before it print 99.97% 99.95%... continuously. How I can obtain more information? some command? debugging options? Thanks a lot. Regards. Fernando. -- _ -- 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] grp_lock error when compiling against pjproject
On Sat, Jan 25, 2014 at 1:35 PM, Ira i...@extrasensory.com wrote: Hello Asterisk, Would someone be kind enough as to add the issue: grp_lock error when compiling against pjproject and solution: delete the rogue install in /usr/local/include To the WIKI page about installing pjsip. I tried to update the WIKI but don't seem to have a way to do it. I know it's not supposed to happen and I know what I did wrong, but it's hard to imagine I'll be the last person to make that mistake. Do you have the exact error message that pjproject gave when you ran into this problem? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] IOPS required by Asterisk for Call Recording
On Mon, Jan 27, 2014 at 1:02 PM, Ron Wheeler rwhee...@artifact-software.com wrote: Can you get a reading of the total number of I/Os during your test? Peak IOPS? That might tell you very quickly about the storage pattern that Asterisk uses. Can you configure a RAM drive to see if disk is really the bottleneck. May need to add some more RAM memory to your configuration. What is your network capacity? Usually one can write faster than the network can deliver - just to make sure that you are chasing the right bottleneck. What happens at 80 calls to tell you that you have run out of IOPS? Dovetailing on this question, I'll add one as well: Are you recording using MixMonitor, or Monitor? Depending on your answer to the what happens at 80 calls, you may get better results with MixMonitor over Monitor. MixMonitor offloads the recording of the media to a separate thread; Monitor attempts to record the audio on the thread servicing the channel(s). Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] B410P BRI card testing with patgen / pattest
Hello, I have a card here at the office that I'm trying to validate with some tests. For now I've followed the instructions available here: http://kb.digium.com/articles/Configuration/Back-to-Back-Pattern-Test-for-BRI-Adapters . I get this kind of output: [..] (.Error 44119): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44120): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44121): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44122): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44123): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44124): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44125): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44126): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44127): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44128): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44129): Unexpected result, 255 != 0, 1 bytes since last error. (Error 44130): Unexpected result, 255 != 0, 1 bytes since last error. [...] This goes on continuously, like a massive flood, not a small burst in the beginning like the KB article mentions. Would this be the kind of output that matches a problem card or something may be wrong in my setup? I'm using DAHDI 2.8.0.1. Any input appreciated, thanks! Regards, Rodrigo -- _ -- 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] IOPS required by Asterisk for Call Recording
Hi,
MixMonitor takes a parameter of a system command to run when the recording
finishes. Like Chris said, you can write to ramdisk, and run a script that
will move the file into final position only when the call has done recording
Here we use:
Set(recordFile=${UNIQUEID}_${NUMBER}.gsm);
Set(recordPath=/var/log/asterisk/recordings/${CALLERID(dnid)}/${STRFTIME(${EPOCH},GMT+0,%F)});
MixMonitor(/ramdrive/${recordFile},,/usr/local/bin/mixmon ${recordFile}
${recordPath});
SIPAddHeader(X-REC-FILE:
${recordPath}/${recordFile});
and /usr/local/bin/mixmon will move the file to $recordPath and whatever
else needs done on that file...
On 27 January 2014 21:55, Matthew Jordan mjor...@digium.com wrote:
On Mon, Jan 27, 2014 at 1:02 PM, Ron Wheeler
rwhee...@artifact-software.com wrote:
Can you get a reading of the total number of I/Os during your test? Peak
IOPS?
That might tell you very quickly about the storage pattern that Asterisk
uses.
Can you configure a RAM drive to see if disk is really the bottleneck.
May
need to add some more RAM memory to your configuration.
What is your network capacity? Usually one can write faster than the
network
can deliver - just to make sure that you are chasing the right
bottleneck.
What happens at 80 calls to tell you that you have run out of IOPS?
Dovetailing on this question, I'll add one as well:
Are you recording using MixMonitor, or Monitor?
Depending on your answer to the what happens at 80 calls, you may
get better results with MixMonitor over Monitor. MixMonitor offloads
the recording of the media to a separate thread; Monitor attempts to
record the audio on the thread servicing the channel(s).
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
--
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