Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Wed, 14 Sep 2011, Kaushal Shriyan wrote: Please let me know the correct procedure to get .alaw file format since I belong to India region. Well, let's see... You used '-t ul' and got a 'ulaw.' I wonder what '-t al' will give you? Failing that, I suspect 'sox --help' or Google would be a more responsive resource. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] SNMP problem
Did you copy the asterisk-mib.txt and digium-mib.txt to the proper folder on your distro? I see people forgetting about that step. On Wed, Sep 14, 2011 at 2:18 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using Asterisk 1.8.3.2 and am trying to configure snmp using this as my resource http://ofps.oreilly.com/titles/9780596517342/asterisk-Monitoring.html when I execute the following command snmpwalk -On -v2c -c public 127.0.0.1 .1.3.6.1.4.1.22736 I get the following response .1.3.6.1.4.1.22736 = No Such Object available on this agent at this OID The contents of my /etc/snmp/snmpd.conf is exactly like the book instructs com2sec notConfigUser default public group notConfigGroup v1 notConfigUser group notConfigGroup v2c notConfigUser view allincluded .1 view system included .iso.org.dod.internet.mgmt.mib-2.system access notConfigGroup any noauthexact allnone none master agentx agentXSocket /var/agentx/master agentXPerms 0660 0775 nobody root sysObjectID .1.3.6.1.4.1.22736.1 Does anyone have any pointers as to where I'm going wrong? Thanks in advancde Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert -- _ -- 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] Anyone using Asterisk on VirtualBox ?
On Sat, Sep 3, 2011 at 1:56 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 1 Sep 2011, RSCL Mumbai wrote: I tried and failed with VirtualBox too. Timing seemed impossible to maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and nothing else going on (single instance). I don't think VirtualBox is up to real-time stuff. We use LXC now, and it is fantastic. j Thx Jeff. Kindly share some more details on the kind of hardware you are using, LXC parameters and the kind of load the system can handle. I am sure this will help me and more like myself. Thx Sanjay My main interest of being on Virtual platform is portability / Backup. In case of any h/w issues, or crashes, simply copy the VM on to another box and you are up in minutes. Sanjay Hi Sanjay, LXC is more of a quasi-virtual platform - it doesn't give you hardware virtualization, but instead lets you share the kernel of the host between multiple instances. To me this allows for multiple efficiencies and advantages that you don't get with hardware virtualization: 1) the host's memory is shared between all instances 2) the host's disk is shared between all instances 3) a shell on the host has access to the files in all of the instances So an instance that is truly idle is taking up very little resource on the host. Versus a traditional hardware virt, which even when idle has an appreciable chunk of RAM and CPU in use all the time. For hosting lots of asterisk instances this is VERY efficient. We have it setup such that the host runs an asterisk image that is the PSTN gateway and has dahdi loaded for timing and access to interface cards. It accepts calls for subscribed DIDs and routes them to the appropriate instance. Each instance has an asterisk process that is dedicated to a customer, which includes their own instance of FreePBX. The dedicated asterisk instance uses a SIP peer connection to the asterisk running on the host which is its outbound access to the PSTN (or other instances). The one gotcha I ran into was configuring the instance to allow access to the dahdi kernel module of the host, which is needed for timing for meetme (we still run 1.4). The conf file needs to contain: # dahdi lxc.cgroup.devices.allow = c 196:0 rwm lxc.cgroup.devices.allow = c 196:253 rwm lxc.cgroup.devices.allow = c 196:254 rwm lxc.cgroup.devices.allow = c 196:255 rwm This is still in proof-of-concept mode for us, but we do have a half dozen customers representing about fifty seats running on it in beta. No complaints in over two months, and the load average may as well be zero. The machine is a quad core Xeon (X3450 @ 2.66Ghz) with 8G RAM, running Ubuntu 11.04. Each instance is a subtree of the host's filesystem, by default (at least in Ubuntu) under /var/lib/lxc. We created a template with a full asterisk and FreePBX installation. To create a new instance we simply untar the template and run a sed script over a set of files to give it an IP address, hostname, and minor edits to various asterisk config files. I haven't done it yet, but I intend to create a mirror of the host machine on another box with rsync, which will serve as the backup. At some point I would like to have the instances running on both mirrors with failover. LXC docs basically suck. If you do go down this road, you will have to be prepared to glean as much as possible from notes various people have posted. I settled on Ubuntu 11.04 as a base because a lot of LXC specific scripts have been created to help with management. Even so its kind of flaky shutting down and rebooting the instances. Once they are running as you like it is stable, but I had a lot of weird things happen along the way as I was tweaking. OpenVZ is the older and more mature equivalent, and may be a better choice to start, but it is not built into the kernel as LXC is. I don't have an real comparisons to provide operationally, but I can vouch for LXC being stable enough for production use so far. I haven't stress tested it yet to see how many instances we can provide on a single host, but am hoping it to be a function of the number of simultaneous calls rather than the number of instances... Would love to hear from anyone else that is using LXC, especially in production. Cheers, j -- @Jeff, @Tarek, I finally decided to move away from Virtualization. I have read a lot of posts on various forums which suggests VB is not fully ready for a real time application like Asterisk, and I have been facing issues all the way. LXC was a bit complicated for me and I was short on time. Did a bare metal install and its working good. My Quad Xeon 2.3 GHz CPU hardly hits 10% with 20 concurrent calls I have only 2GB RAM for now and its 50% used. Created a CloneZilla image last night, plan to install it on another similar hardware later today. I am wondering how to resolve ethernet conflict while
Re: [asterisk-users] Anyone using Asterisk on VirtualBox ?
Hi What about OpenVZ. Its good On Thu, Sep 15, 2011 at 12:38, RSCL Mumbai rscl.mum...@gmail.com wrote: On Sat, Sep 3, 2011 at 1:56 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 1 Sep 2011, RSCL Mumbai wrote: I tried and failed with VirtualBox too. Timing seemed impossible to maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and nothing else going on (single instance). I don't think VirtualBox is up to real-time stuff. We use LXC now, and it is fantastic. j Thx Jeff. Kindly share some more details on the kind of hardware you are using, LXC parameters and the kind of load the system can handle. I am sure this will help me and more like myself. Thx Sanjay My main interest of being on Virtual platform is portability / Backup. In case of any h/w issues, or crashes, simply copy the VM on to another box and you are up in minutes. Sanjay Hi Sanjay, LXC is more of a quasi-virtual platform - it doesn't give you hardware virtualization, but instead lets you share the kernel of the host between multiple instances. To me this allows for multiple efficiencies and advantages that you don't get with hardware virtualization: 1) the host's memory is shared between all instances 2) the host's disk is shared between all instances 3) a shell on the host has access to the files in all of the instances So an instance that is truly idle is taking up very little resource on the host. Versus a traditional hardware virt, which even when idle has an appreciable chunk of RAM and CPU in use all the time. For hosting lots of asterisk instances this is VERY efficient. We have it setup such that the host runs an asterisk image that is the PSTN gateway and has dahdi loaded for timing and access to interface cards. It accepts calls for subscribed DIDs and routes them to the appropriate instance. Each instance has an asterisk process that is dedicated to a customer, which includes their own instance of FreePBX. The dedicated asterisk instance uses a SIP peer connection to the asterisk running on the host which is its outbound access to the PSTN (or other instances). The one gotcha I ran into was configuring the instance to allow access to the dahdi kernel module of the host, which is needed for timing for meetme (we still run 1.4). The conf file needs to contain: # dahdi lxc.cgroup.devices.allow = c 196:0 rwm lxc.cgroup.devices.allow = c 196:253 rwm lxc.cgroup.devices.allow = c 196:254 rwm lxc.cgroup.devices.allow = c 196:255 rwm This is still in proof-of-concept mode for us, but we do have a half dozen customers representing about fifty seats running on it in beta. No complaints in over two months, and the load average may as well be zero. The machine is a quad core Xeon (X3450 @ 2.66Ghz) with 8G RAM, running Ubuntu 11.04. Each instance is a subtree of the host's filesystem, by default (at least in Ubuntu) under /var/lib/lxc. We created a template with a full asterisk and FreePBX installation. To create a new instance we simply untar the template and run a sed script over a set of files to give it an IP address, hostname, and minor edits to various asterisk config files. I haven't done it yet, but I intend to create a mirror of the host machine on another box with rsync, which will serve as the backup. At some point I would like to have the instances running on both mirrors with failover. LXC docs basically suck. If you do go down this road, you will have to be prepared to glean as much as possible from notes various people have posted. I settled on Ubuntu 11.04 as a base because a lot of LXC specific scripts have been created to help with management. Even so its kind of flaky shutting down and rebooting the instances. Once they are running as you like it is stable, but I had a lot of weird things happen along the way as I was tweaking. OpenVZ is the older and more mature equivalent, and may be a better choice to start, but it is not built into the kernel as LXC is. I don't have an real comparisons to provide operationally, but I can vouch for LXC being stable enough for production use so far. I haven't stress tested it yet to see how many instances we can provide on a single host, but am hoping it to be a function of the number of simultaneous calls rather than the number of instances... Would love to hear from anyone else that is using LXC, especially in production. Cheers, j -- @Jeff, @Tarek, I finally decided to move away from Virtualization. I have read a lot of posts on various forums which suggests VB is not fully ready for a real time application like Asterisk, and I have been facing issues all the way. LXC was a bit complicated for me and I was short on time. Did a bare metal install and its working good. My Quad Xeon 2.3 GHz CPU hardly hits 10% with 20 concurrent calls I have only 2GB RAM for now and its 50% used. Created a CloneZilla image last night,
Re: [asterisk-users] SNMP problem
Hi Robert I have not seen any reference to those 2 files anywhere before. They are both on the * server in /usr/share/snmp/mibs/ Where should they be copied to? Ish On Thu, 2011-09-15 at 00:44 -0600, Robert Thomas wrote: Did you copy the asterisk-mib.txt and digium-mib.txt to the proper folder on your distro? I see people forgetting about that step. On Wed, Sep 14, 2011 at 2:18 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using Asterisk 1.8.3.2 and am trying to configure snmp using this as my resource http://ofps.oreilly.com/titles/9780596517342/asterisk-Monitoring.html when I execute the following command snmpwalk -On -v2c -c public 127.0.0.1 .1.3.6.1.4.1.22736 I get the following response .1.3.6.1.4.1.22736 = No Such Object available on this agent at this OID The contents of my /etc/snmp/snmpd.conf is exactly like the book instructs com2sec notConfigUser default public group notConfigGroup v1 notConfigUser group notConfigGroup v2c notConfigUser view allincluded .1 view system included .iso.org.dod.internet.mgmt.mib-2.system access notConfigGroup any noauthexact all none none master agentx agentXSocket /var/agentx/master agentXPerms 0660 0775 nobody root sysObjectID .1.3.6.1.4.1.22736.1 Does anyone have any pointers as to where I'm going wrong? Thanks in advancde Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Thu, Sep 15, 2011 at 11:45 AM, Steve Edwards asterisk@sedwards.com wrote: On Wed, 14 Sep 2011, Kaushal Shriyan wrote: Please let me know the correct procedure to get .alaw file format since I belong to India region. Well, let's see... You used '-t ul' and got a 'ulaw.' I wonder what '-t al' will give you? Failing that, I suspect 'sox --help' or Google would be a more responsive resource. Thanks Steve. it worked this time. Also is there a way to verify .alaw file without playing the file. Although i did play obd-demo.alaw it worked fine. I did ran the below command [root@host0040 test]# file obd-demo.alaw obd-demo.alaw: data [root@host0040 test]# [root@host0040 test]# sox obd-demo.alaw -e stat sox: Failed reading obd-demo.alaw: Do not understand format type: alaw [root@host0040 test]# Basically trying to understand the properties of the .alaw file about encoding and details. Please guide. [root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t al -c 1 obd-demo.alaw sox: Detected file format type: wav sox: WAV Chunk fmt sox: WAV Chunk data sox: Reading Wave file: Microsoft PCM format, 1 channel, 44100 samp/sec sox: 88200 byte/sec, 2 block align, 16 bits/samp, 2041438 data bytes sox: 1020719 Samps/chans sox: Input file obd-demo.wav: using sample rate 44100 size shorts, encoding signed (2's complement), 1 channel sox: Output file obd-demo.alaw: using sample rate 8000 size bytes, encoding a-law, 1 channel sox: Output file: comment Processed by SoX sox: resample opts: Kaiser window, cutoff 0.80, beta 16.00 [root@host0040 test]# ls -ltrh total 2.9M -rwxr-xr-x 1 root root 725K Sep 14 06:32 obd-demo.mp3 -rw-r--r-- 1 root root 2.0M Sep 14 06:32 obd-demo.wav -rw-r--r-- 1 root root 181K Sep 15 16:57 obd-demo.alaw -- _ -- 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 PRI hangup
Hello all, Form 2-3 weeks i have some problems with incoming ISDN calls, it interrupts after 1-2 minutes of call. I have tried to debug this with pri set debug on span 1, i have noticied much of this messages: -- Timeout occured, restarting PRI q921.c:468 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending TEI management message 1, TEI=127 Received MDL message TEI assiged to 71 q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending Set Asynchronous Balanced Mode Extended q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH -- Got UA from network peer Link up. q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:805 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED -- Timeout occured, restarting PRI q921.c:468 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending TEI management message 1, TEI=127 Received MDL message TEI assiged to 72 q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending Set Asynchronous Balanced Mode Extended q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH -- Got UA from network peer Link up. q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:805 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED And there is a debug session of an hanged-up incoming call: Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] == Extension Changed 215[ext-local] new state Ringing for Notify User 202 -- SIP/203-0017 is ringing -- SIP/206-0019 is ringing -- SIP/210-001a is ringing -- SIP/205-0018 is ringing -- SIP/201-0015 is ringing -- SIP/215-001b is ringing -- SIP/202-0016 is ringing -- SIP/201-0015 answered DAHDI/1-1 == Extension Changed 201[ext-local] new state InUse for Notify User 202 == Extension Changed 201[ext-local] new state InUse for Notify User 215 == Extension Changed 215[ext-local] new state Idle for Notify User 202 == Extension Changed 210[ext-local] new state Idle for Notify User 202 == Extension Changed 210[ext-local] new state Idle for Notify User 215 == Extension Changed 206[ext-local] new state Idle for Notify User 202 == Extension Changed 206[ext-local] new state Idle for Notify User 215 == Extension Changed 205[ext-local] new state Idle for Notify User 202 == Extension Changed 205[ext-local] new state Idle for Notify User 215 == Extension Changed 203[ext-local] new state Idle for Notify User 202 == Extension Changed 203[ext-local] new state Idle for Notify User 215 -- Executing [s@macro-auto-blkvm:1] Set(SIP/201-0015, __MACRO_RESULT=) in new stack == Extension Changed 202[ext-local] new state Idle for Notify User 215 -- Executing [s@macro-auto-blkvm:2] NoOp(SIP/201-0015, Deleting: BLKVM/600/DAHDI/1-1 TRUE) in new stack -- Stopped music on hold on DAHDI/1-1 q931.c:2951 q931_connect: call 93 on channel 1 enters state 8 (Connect Request) Protocol Discriminator: Q.931 (8) len=11 Call Ref: len= 1 (reference 93/0x5D) (Terminator) Message type: CONNECT (7) [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit Other Spare: 0 Exclusive Dchan: 0 ChanSel: B1 channel ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 93/0x5D) (Originator) Message type: CONNECT ACKNOWLEDGE (15) q931.c:3711 q931_receive: call 93 on channel 1 enters state 10 (Active) -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:805 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 93/0x5D) (Originator) Message type: STATUS ENQUIRY (117) YYY Here we get reset YYY Protocol Discriminator: Q.931 (8) len=7 Call Ref: len= 1 (reference 93/0x5D) (Terminator) Message type: STATUS (125) [14 01 00] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Null (0) -- Got reject requesting packet 0... Retransmitting. Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 93/0x5D) (Originator) Message type: STATUS ENQUIRY (117) YYY Here we get reset YYY Protocol Discriminator: Q.931 (8) len=7 Call Ref: len= 1 (reference 93/0x5D)
Re: [asterisk-users] High delay from Asterisk as PSTN simulator
I understand. I'm interested in simulate the real situation because I'm doing an academic comparative between algorithms, and is really interesting have all possible situations. In the real situation I use a E1 to connect a PBX through a R2 link, so I want to try change DAHDI to 10 ms... What should I modify? Any other sugestion to make it works? 2011/9/14 Kevin P. Fleming kpflem...@digium.com On 09/14/2011 02:37 PM, Gustavo Santos wrote: I'm trying to simulate the situation: SIP Asterisk --- PSTN In this case 16 ms works? I've read in voip-info: Simplistically, you'd need a tail circuit (the distance between your echo canceller and the source of the echo) of over 2500 miles to acheive an echo path of 30ms [...] Asterisk's default of 128taps will therefore handle echo paths of up to 16ms, and is therefore probably good for most things.. You are missing some basic details of the environment here. I'll try to explain. In the diagram you've shown above (assuming there is an FXO port in the Asterisk server connected to an FXO line from the PSTN), there are potentially two sources of line echo: the 2/4 wire hybrid in the FXO port, and the 2/4 wire hybrid at the far end of the FXO line (and potentially even farther into the PSTN, but we can ignore that here). Echo caused by the FXO port hybrid would be heard by the person at the other end of the FXO line (across the PSTN), and would not be cancelled by any echo canceller on the FXO card or in DAHDI. Echo caused by the far end would be heard by the user of the SIP phone, and could potentially be cancelled by an echo canceller on the FXO card or in DAHDI. That quote you've included above is correct: assuming a *TRADITIONAL* PSTN link (no VoIP, no packetization of audio, all circuits either analog or TDM), the echo generated by the far end of the FXO line will likely not be more than 16ms after the transmission. In this case, a 16ms echo canceller window will be adequate. If an echo (primary or secondary) is generated by the real far end (across the PSTN), it could easily be delayed by 30ms (or much more). In these cases, having a 64ms or 128ms echo canceller window is beneficial, and with modern hardware is not expensive to provide (or harmful in any way). However... using Asterisk with an FXS card and the Echo() application is *NOT* a 'PSTN simulator'. When an audio signal is received into the FXS card, it will take 20-40ms to be sent back out the FXS card, depending on packetization intervals, scheduling delays and other factors. This is because, as I stated previously, Asterisk is internally a 'voice over packet' system, and it does not have any way to forward audio in anything less than reasonable size packets. For cards driven by DAHDI, 'reasonable' defaults to 20ms, although it could be changed to 10ms with a corresponding increase in CPU overhead... but even if you did change it, it is likely that under many situations the echoed audio would be delayed by more than 16ms. If you *need* to test an echo canceller configured with a tiny 16ms window, you'll have to find another way of generating echo for it to be tested against. -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Atenciosamente, Gustavo Santos. -- _ -- 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] testing simultaneous calls
Asterisk 1.4.26 keeps randomly crashing then restarting itself on my live production. I cannot run valgrind and I do not have the right flags set in menuselect. I can however at the dead of the night run stress tests. I want to simulate x-amount of concurrent calls to both a dtmf dialplan, which is working, as well as MoH dialplan to see if this could be the cause of crashing. How do I test this? Is it a call file that can handle this without ringing my extension first, like internal system calling? Thanks, --Eric -- _ -- 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] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Thu, 15 Sep 2011, Kaushal Shriyan wrote: I did ran the below command [root@host0040 test]# file obd-demo.alaw obd-demo.alaw: data [root@host0040 test]# sox obd-demo.alaw -e stat sox: Failed reading obd-demo.alaw: Do not understand format type: alaw [au]law are 'headerless' file formats so the 'file' command can't help. Sox needs a clue as well. (See what sox says if you change the file name from example.alaw to example.al.) Basically trying to understand the properties of the .alaw file about encoding and details. Please guide. Personally, I prefer to use WAV (not the funky 'gsm in wav' kind) files throughout. I don't think 'transcoding' between [au]law and wav is a big deal CPU wise and I like easy to use file formats. Call me lazy, but being able to just type 'play example.wav' or 'audacity example.wav' has value to me. So does being able to email an audio file to a 'non-techie' and not having to explain anything. If you're looking to squeeze every last CPU cycle, there are probably better places to look. If you are really that tight on resources, maybe you should reconsider your hardware choices. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
+1 (at least) Steve -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, September 15, 2011 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 On Thu, 15 Sep 2011, Kaushal Shriyan wrote: I did ran the below command [root@host0040 test]# file obd-demo.alaw obd-demo.alaw: data [root@host0040 test]# sox obd-demo.alaw -e stat sox: Failed reading obd-demo.alaw: Do not understand format type: alaw [au]law are 'headerless' file formats so the 'file' command can't help. Sox needs a clue as well. (See what sox says if you change the file name from example.alaw to example.al.) Basically trying to understand the properties of the .alaw file about encoding and details. Please guide. Personally, I prefer to use WAV (not the funky 'gsm in wav' kind) files throughout. I don't think 'transcoding' between [au]law and wav is a big deal CPU wise and I like easy to use file formats. Call me lazy, but being able to just type 'play example.wav' or 'audacity example.wav' has value to me. So does being able to email an audio file to a 'non-techie' and not having to explain anything. If you're looking to squeeze every last CPU cycle, there are probably better places to look. If you are really that tight on resources, maybe you should reconsider your hardware choices. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Monitoring second leg being dialed?
Hello My ISP provides an FXS port to plug a handset, which can be used to make free calls to (GSM) cellphones, similar to the Billion ADSL modems: http://au.billion.com/product/voip.php My plan is to install an SIP client on a smartphone, so that when I'm travelling, I can connect to a good wifi hotspot, register with an Asterisk server at home which has an FXO card, tell Asterisk the number I wish to dial, and have it dial out through the FXO card and the FXS port on the ADSL modem. Here's the diagram: http://img844.imageshack.us/img844/3308/asterisksippstncallback.png Problem is, Dahdi/Zaptel doesn't provide call progression, so that 1) when Asterisk passes the call to Dahdi/Zaptel, Asterisk considers the call answered although there's no actual phone connection yet, and 2) Dahdi/Zaptel doesn't trigger an event so we know if the call was answered (and if yes, by a live human being rather than an answering machine) or if the line is still ringing. A so-so solution is to simply tell Asterisk to loop through a voice message (This is a call from Joe Allen. Please hit any key and you will be connected), so we know that a human being has answered the call, but I was wondering if there were a better solution. Is it possible for Asterisk to somehow play on channel #1 what's happening on channel #2 while Dahdi/Zaptel is actually still dialing, so that I handle call progression manually from my cellphone and the callee doesn't end up hearing that odd recorded message? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Android?
On Thu, 08 Sep 2011 14:52:06 -0400, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 08/09/11 02:19 PM, Cobra 2 wrote: I've chrooted debian onto a Motorola Droid running Cyanogenmod 7 and I've gotten asterisk to run on that just fine. I think the question is, can you answer your incoming calls with the Asterisk running on the device? Yes, that's the plan. I'd like Asterisk to run an IVR to screen incoming calls. Cobra: Out of curiosity, what did you use Asterisk for on that Motorola phone if not to handle incoming calls? -- _ -- 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
