Re: [asterisk-users] Queue Agent Getting Additional Calls When on the Phone
Sorry for the top-post... If you do a core show application AddQueueMember from the cli, you'll see the option I was referring to. You'll also need to make sure you're properly reporting device state to asterisk. I think this means you need to set a call-limit for each sip peer that you want to monitor in sip.conf (we use 25 so there are no accidental limits actually applied), and setup hints in your extensions.conf for each peer. Thanks, --Warren Selby On Oct 14, 2010, at 11:36 PM, Matt Darnell mattdarn...@gmail.com wrote: Warren, I tried using AddQueueMember to add agents. If they a user is on a call asterisk shows: Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers We are using 1.4.36. What did you use to keep track of the extension state? Didn't see any option for that at http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember Thanks for the help. -Matt On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Thanks, --Warren Selby On Oct 14, 2010, at 10:13 PM, Matt Darnell mattdarn...@gmail.com wrote: We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Agent Getting Additional Calls When on the Phone
You'll also need to make sure you're properly reporting device state to asterisk. I think this means you need to set a call-limit for each sip peer that you want to monitor in sip.conf (we use 25 so there are no accidental limits actually applied), and setup hints in your extensions.conf for each peer. Warren, Setting the call limits was my issue. I am on a test machine and didn't have it set. Thanks for the help! -Matt -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? Thanks, --Warren Selby On Oct 14, 2010, at 11:36 PM, Matt Darnellmattdarn...@gmail.com wrote: Warren, I tried using AddQueueMember to add agents. If they a user is on a call asterisk shows: Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers We are using 1.4.36. What did you use to keep track of the extension state? Didn't see any option for that at http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember Thanks for the help. -Matt On Thu, Oct 14, 2010 at 6:04 PM, Warren Selbywcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Thanks, --Warren Selby On Oct 14, 2010, at 10:13 PM, Matt Darnellmattdarn...@gmail.com wrote: We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)
Hi, I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN card. The system starts without any errors. I discovered a severe issue. The kernel panics on a very small load. The first call normally gets through. If I start the second or third call and sometimes when I terminate the first call, the system panics (Oops text on console). After solving some difficulties (the relevant part of the Oops text scrolls out of the monitor, no serial interface), I get the text via netconsole. It seems to me, that the panic occurred in oslec (function oslec_update). But maybe I am wrong with this. In the oslec code there is a patch to enable MMX. After switching this off, the problem disappeared. AFAIK the cpu supports mmx. Where should I address this issue to? Is it a known issue? Here comes one example for the oops: /- BUG: unable to handle kernel NULL pointer dereference at (null) IP: [c0103dd6] __math_state_restore+0x56/0x90 *pde = Oops: [#1] PREEMPT SMP last sysfs file: /sys/module/configfs/initstate Modules linked in: netconsole configfs dahdi_echocan_oslec echo capifs loop wcb4xxp rtc_cmos i2c_i801 rtc_core dahdi 8250_pnp 8139too floppy 8250 rtc_lib mii serial_core i2c_core processor pcspkr rng_core button ide_pci_generic ide_core sd_mod crc_t10dif thermal [last unloaded: netconsole] Pid: 1268, comm: clip.agi Not tainted 2.6.35.4 #1 P4Dual-915GL/P4Dual-915GL EIP: 0060:[c0103dd6] EFLAGS: 00010046 CPU: 0 EIP is at __math_state_restore+0x56/0x90 EAX: EBX: c5b2 ECX: cd461960 EDX: ESI: cd461960 EDI: c01045a0 EBP: 0080 ESP: c5b21cb0 DS: 007b ES: 007b FS: 00d8 GS: 00e0 SS: 0068 Process clip.agi (pid: 1268, ti=c5b2 task=cd461960 task.ti=c5b2) Stack: c5b21cd0 0027 c01045a0 c01045e5 0200 cfadd500 c0432273 0 cfadd500 cfadd200 0008 0027 0080 0080 cf33fa00 007b 0 007b c02d00d8 00e0 d0ae2153 0060 00010002 005a Call Trace: [c01045a0] ? do_device_not_available+0x0/0x60 [c01045e5] ? do_device_not_available+0x45/0x60 [c0432273] ? error_code+0x73/0x80 [c02d00d8] ? DAC960_V1_ProcessCompletedCommand+0x1108/0x1510 [d0ae2153] ? oslec_update+0xe3/0x5c0 [echo] [d0aeb038] ? echo_can_process+0x28/0x40 [dahdi_echocan_oslec] [d0aeb010] ? echo_can_process+0x0/0x40 [dahdi_echocan_oslec] [d0a08a18] ? dahdi_ec_span+0x268/0x2a0 [dahdi] [d0a9136c] ? b4xxp_interrupt+0x11c/0x358 [wcb4xxp] [c0175ded] ? handle_IRQ_event+0x2d/0xc0 [c02dd71d] ? scsi_decide_disposition+0x16d/0x180 [c0177b85] ? handle_fasteoi_irq+0x65/0xd0 [c0105a55] ? handle_irq+0x15/0x30 [c01050a7] ? do_IRQ+0x47/0xc0 [c0103d30] ? common_interrupt+0x30/0x40 [c01300e0] ? load_balance+0x550/0x7d0 [c0431614] ? _raw_spin_unlock_irq+0x4/0x20 [c012d9ba] ? finish_task_switch+0x3a/0x90 [c042f5c9] ? schedule+0x1c9/0x520 [c0103d30] ? common_interrupt+0x30/0x40 [c042facf] ? preempt_schedule+0x2f/0x50 [c0198a60] ? do_wp_page+0x160/0x960 [c0199c02] ? handle_mm_fault+0x5d2/0xaa0 [c01244b0] ? do_page_fault+0x0/0x370 [c01245f0] ? do_page_fault+0x140/0x370 [c01b7b2f] ? copy_strings+0x17f/0x1a0 [c01b935e] ? do_execve+0x2be/0x310 [c01b935e] ? do_execve+0x2be/0x310 [c010aa80] ? sys_execve+0x40/0x70 [c01244b0] ? do_page_fault+0x0/0x370 [c0432273] ? error_code+0x73/0x80 Code: 89 c2 0f ae 2f 85 c9 75 27 83 4b 0c 01 80 86 98 00 00 00 01 8b 1c 24 8b 74 24 04 8b 7c 24 08 83 c4 0c c3 66 90 8b 86 50 02 00 00 0f ae 08 eb d9 e8 c0 ed 01 00 90 83 c8 08 e8 c7 ed 01 00 90 b8 EIP: [c0103dd6] __math_state_restore+0x56/0x90 SS:ESP 0068:c5b21cb0 CR2: ---[ end trace 65c27cd3a6b7bd8a ]--- \- Thanks, Karsten -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
On 10-10-15 04:10 AM, Сикорский Сергей wrote: 15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? The correct answer is to use ringinuse=no in queues.conf and callcounter=yes in sip.conf. Leif. -- _ -- 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] drop dead fix
pbx$ man sox allpass frequency[k] width[h|k|o|q] Apply a two-pole all-pass filter with central frequency (in Hz) frequency, and filter-width width. An all- pass filter changes the audio's frequency to phase relationship without changing its frequency to amplitude relationship. The filter is described in detail in [1]. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux -- _ -- 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] drop dead fix
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: Friday, October 15, 2010 9:10 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] drop dead fix On 10/15/2010 09:59 AM, Danny Nicholas wrote: Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Thanks Danny Nicholas hiring someone to re-record 304 prompts is not simpler and far faster than redeploying an entire system ? sounds like about a 4hr job. or find a better converter. Option 2 is what I have in mind (BTW, with the talent I have, your 4 hrs is closer to 80, after normalizing, trimming and prodding). What I do now is record the file using soundrec, normalize it with Audiograbber, then trim it with Audacity before converting it with Sox. Which of these is letting me down, (or it is the loose nut on the keyboard)? -- _ -- 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] drop dead fix
On Fri, 15 Oct 2010, Danny Nicholas wrote: Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Why are you converting them to GSM? Why not convert them to the technology you're using for your phones and trunks? That would be much more efficient. (If you're using g729 for trunks, then that will sound better as GSM to g729 conversion does sound bad) Or maybe it's your conversion software? What are you using? Gordon -- _ -- 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] drop dead fix
On Fri, 15 Oct 2010, Danny Nicholas wrote: I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Can you post a link to a sample before and after file as well as the command line you are using to convert the file? -- 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] fraud advice
On Thu, 14 Oct 2010, bruce bruce wrote: But it also sickens me at how badly Asterisk is made to not cope with situations like this and worse than that is FreePBX. Kind of like blaming the gun manufacturer instead of the criminal with their finger on the trigger? Is there some gaping hole in Asterisk security or are you just asleep at the wheel? -- 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] drop dead fix
I never had this problem, and this is certainly not asterisk's fault. Probably your conversion is not good. Can you email me a file and I'll do conversion on my end, and if sounds good, let you know how I did it. Then a script can be written to convert them all. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-15 10:25 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 15 Oct 2010, Danny Nicholas wrote: I am about to have to dump Asterisk in f... Can you post a link to a sample before and after file as well as the command line you are using to convert the file? -- 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
Re: [asterisk-users] drop dead fix
You want to pay attention the low-pass and high-pass filter A step conversion will help you see the issues. Go halfway first and look for the change and adjust your filter. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Fri, Oct 15, 2010 at 11:18 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Fri, 15 Oct 2010, Danny Nicholas wrote: Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Why are you converting them to GSM? Why not convert them to the technology you're using for your phones and trunks? That would be much more efficient. (If you're using g729 for trunks, then that will sound better as GSM to g729 conversion does sound bad) Or maybe it's your conversion software? What are you using? Gordon -- _ -- 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] fraud advice
For future I would highly recommend to have at least fail2ban installed. This way sipvicous IPs will be blocked instantly before they could create any damage. Also I prefer to limit International calling to only certain limit, e.g. only for $10 per account, but this depends upon how your business deals with international calls. I get a few IPs blocked everyday by fail2ban, though by default no new connections are allowed international calls on my system. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-15 10:40 AM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 14 Oct 2010, bruce bruce wrote: But it also sickens me at how badly Asterisk is made to n... Kind of like blaming the gun manufacturer instead of the criminal with their finger on the trigger? Is there some gaping hole in Asterisk security or are you just asleep at the wheel? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Pr... -- _ -- 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] Audio problems on cable modem link
We have a small office installation running over a cable modem. (8M down, 500k up confirmed with numerous speed test sites) When a single call is up, call quality is fine. When a second call is up, outbound audio is immediately choppy. We're using ulaw, and confirmed that traffic with 2 calls is 175kbps in/out. (IAX connection out) Asterisk doesn't report any dropped frames, the internet connection looks fine, etc. We have a linux router in place running wondershaper that seems to be running fine (same as our other installations). Can someone suggest where to look? Could this be the ITSP? -- _ -- 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] drop dead fix
On Fri, 15 Oct 2010, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Friday, October 15, 2010 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] drop dead fix On Fri, 15 Oct 2010, Danny Nicholas wrote: Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Why are you converting them to GSM? Why not convert them to the technology you're using for your phones and trunks? That would be much more efficient. (If you're using g729 for trunks, then that will sound better as GSM to g729 conversion does sound bad) Or maybe it's your conversion software? What are you using? Gordon I did the proof of concept recordings as gsm files. Now that we want to actually do a finished product, the gsm recordings don't sound good enough to make a viable product. Here is a sample Original file http://www.4shared.com/audio/PDGcMDUt/firstmenuwav.html Seems very quiet to me, but I don't have any tools to meansure it where I am right now. The GSM one didn't sounds too bad either, but are you then listening to it after a G729 conversion? Gordon -- _ -- 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] Echo on PRI after upgrading to Asterisk 1.6.2 from 1.2
On 10/15/2010 08:55 AM, Jared Geiger wrote: I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a full reformat and recompile) and I started getting echo over the PRI. I've tried the default settings for echo in the system.conf file as well as I've compiled OSLEC to try and see if thats any better. I'm not sure what to try next. Does anyone have any suggestions? What are the outputs of the following commands when your system is up and running? #] cat /etc/dahdi/system.conf #] grep -E ^echo /etc/asterisk/chan_dahdi.conf #] dahdi_scan #] lsmod | grep dahdi -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] Audio problems on cable modem link
Hi! Can someone suggest where to look? Could this be the ITSP? - turn off IAX trunking mode - test with SIP to find if it IAX causing the trouble - capture the RTP traffice on the other side and let wireshark have a look at that stats (loss, jitter) Philipp -- _ -- 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] drop dead fix
On Fri, Oct 15, 2010 at 11:02 AM, Danny Nicholas da...@debsinc.com wrote: The original one is super quiet - obviously not Allison in a studio... Listen to the gsm in Asterisk to see my quandary... What is the end use here? Who will be listening to the recordings? Users on PSTN and mobile phones? -- _ -- 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] fraud advice
On Fri, Oct 15, 2010 at 10:29 AM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 14 Oct 2010, bruce bruce wrote: But it also sickens me at how badly Asterisk is made to not cope with situations like this and worse than that is FreePBX. Kind of like blaming the gun manufacturer instead of the criminal with their finger on the trigger? Is there some gaping hole in Asterisk security or are you just asleep at the wheel? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 This is nothing new. Trunk to trunk transfers and other exploits could be used on old school phone systems to do the same thing. I would start with getting the current balance, if over $10k call the FBI, call them anyways, it couldn't hurt. You want the Feds to check things out before local police if possible. Gather as much info as possible, along with police and FBI case numbers and then call the carrier and see what can be done. A friend of mine took what was supposed to be my one month rotation to Iraq. I had too much going on to be in Iraq for a month and a half and had taken the last rotation so it wasn't even my turn. The phone bill came for his cell (company provided on Asia Cell) for $4k in just a couple weeks. It turns out that he was not using the cell and one of the cleaning people stole his SIM. After contacting Asia Cell a few times about the matter, they credited the whole amount back. So you never know. As for security, I assume you need to allow these extensions to register from outside the LAN? If not, then only allow them to register via a LAN IP, I would do it with iptables, only allow the provider IP through. I am curious what your user:pass was? something like 1000:1000, I see many systems setup like this and am surprised they haven't been hit yet. In the future, you could use a scheme that makes it much more secure and also pretty easy to maintain. The username could be the MAC and the pass could be the serial number or asset tags if you use them. I know there must be dozens of people reading this that have had the same issue but are embarrassed to speak up. (BTW Sierra Leone is in West Africa, not the Middle East.) Thanks, Steve T -- _ -- 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] drop dead fix
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Deneen Sent: Friday, October 15, 2010 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] drop dead fix On Fri, Oct 15, 2010 at 11:02 AM, Danny Nicholas da...@debsinc.com wrote: The original one is super quiet - obviously not Allison in a studio... Listen to the gsm in Asterisk to see my quandary... What is the end use here? Who will be listening to the recordings? Users on PSTN and mobile phones? End use is Telephone Banking, so you've nailed the target audience. BTW, the highpass and lowpass filters seem to help, but since I stopped math at pre-calculus, the explanation of the Butterworth filter is beyond my pay grade. Care to offer a better explanation? -- _ -- 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] drop dead fix
On 10/15/2010 08:59 AM, Danny Nicholas wrote: Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? In addition to all the other comments you've received (including the fact that Asterisk does not require GSM format files), keep in mind that *any* product that plays these files over the PSTN is going to have to downsample them to 8KHz and, at a minimum, use G.711 companding. That is what the PSTN uses, so it's not possible to have higher fidelity than that. There were some comments in other replies about your files being 'quiet' (low average volume level)... this won't help your situation at all, because it means that any artifacts caused by resampling and compression/decompression will end up at a relatively high amplitude compared to the original signal (resulting in a low signal-to-noise ratio), and when the listener increases the volume level on their listening device, the noise level will be increased along with it. For these sorts of tasks, you really do want the source material recorded at a fairly high volume level. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] drop dead fix
On Fri, Oct 15, 2010 at 11:20 AM, Danny Nicholas da...@debsinc.com wrote: End use is Telephone Banking, so you've nailed the target audience. BTW, the highpass and lowpass filters seem to help, but since I stopped math at pre-calculus, the explanation of the Butterworth filter is beyond my pay grade. Care to offer a better explanation? While, officially, I completed up to calculus 3, the serious lack of use is not helping. You'd be better off taking the highpass number from low to high and listen to the difference, and then do the same for the lowpass number. Your ears will tell you when you have it right (you will definitely hear what each one does), and you can still consider Butterworth inexpensive pancake syrup. -M -- _ -- 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] Echo on PRI after upgrading to Asterisk 1.6.2 from 1.2
[r...@voice ~]# cat /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Fri Sep 24 21:44:03 2010 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCT1/0 Wildcard TE120P Card 0 (MASTER) span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=OSLEC,1-23 # Global data loadzone= us defaultzone = us This might be my problem?*** [r...@voice ~]# grep -E ^echo /etc/asterisk/chan_dahdi.conf * *So I added this under [channels]: echocancel=yes echocancelwhenbridged=no echotraining=800* [r...@voice ~]# dahdi_scan [1] active=yes alarms=OK description=Wildcard TE120P Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE120P location=PCI Bus 03 Slot 12 basechan=1 totchans=24 irq=217 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [r...@voice ~]# lsmod | grep dahdi dahdi_echocan_oslec 6912 27 echo9600 1 dahdi_echocan_oslec dahdi_transcode12164 1 wctc4xxp dahdi_voicebus 45760 2 wctdm24xxp,wcte12xp dahdi 196552 78 dahdi_echocan_oslec,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 6337 2 wctdm24xxp,dahdi On Fri, Oct 15, 2010 at 11:03 AM, Shaun Ruffell sruff...@digium.com wrote: On 10/15/2010 08:55 AM, Jared Geiger wrote: I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a full reformat and recompile) and I started getting echo over the PRI. I've tried the default settings for echo in the system.conf file as well as I've compiled OSLEC to try and see if thats any better. I'm not sure what to try next. Does anyone have any suggestions? What are the outputs of the following commands when your system is up and running? #] cat /etc/dahdi/system.conf #] grep -E ^echo /etc/asterisk/chan_dahdi.conf #] dahdi_scan #] lsmod | grep dahdi -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] fraud advice
We took a pretty nasty hit one time, a system administrator didnt listen to us about changing the passwords. Luckily they took part of the blame in that, and we split the 1800$ it cost us in half. We could have changed them, and she didnt change them, so we were both at fault. Like said previously, fail2ban is a pretty good start. Weak secrets definitely dont help. An interesting project to look into and i'm working with right now, i've got a honeypot set up in the wild, but havent gotten anything really worth while yet... http://www.infiltrated.net/voipabuse/defensive.html I'd also suggest, if you dont *have* to have international dialing on the trunk. Turn it off, put a pin on it, or just send it to a dummy trunk that doesnt do anything or route anywhere. I really hope this helps, and best of luck with cleaning up from the aftermath. I know ours was a pretty good wake up call to us to really start locking things down. I know its lame, but from Network Security Hacks. Security isn't a noun, it's a verb; not a product, but a process --Matt On Fri, Oct 15, 2010 at 11:50 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Fri, 2010-10-15 at 11:20 -0400, Steve Totaro wrote: This is nothing new. Trunk to trunk transfers and other exploits could be used on old school phone systems to do the same thing. I would start with getting the current balance, if over $10k call the FBI, call them anyways, it couldn't hurt. You want the Feds to check things out before local police if possible. Gather as much info as possible, along with police and FBI case numbers and then call the carrier and see what can be done. A friend of mine took what was supposed to be my one month rotation to Iraq. I had too much going on to be in Iraq for a month and a half and had taken the last rotation so it wasn't even my turn. The phone bill came for his cell (company provided on Asia Cell) for $4k in just a couple weeks. It turns out that he was not using the cell and one of the cleaning people stole his SIM. After contacting Asia Cell a few times about the matter, they credited the whole amount back. So you never know. As for security, I assume you need to allow these extensions to register from outside the LAN? If not, then only allow them to register via a LAN IP, I would do it with iptables, only allow the provider IP through. I am curious what your user:pass was? something like 1000:1000, I see many systems setup like this and am surprised they haven't been hit yet. In the future, you could use a scheme that makes it much more secure and also pretty easy to maintain. The username could be the MAC and the pass could be the serial number or asset tags if you use them. I know there must be dozens of people reading this that have had the same issue but are embarrassed to speak up. Thanks Steve - that is the kind of advice I was looking for. I'm willing to take my lumps for the weak passwords on those accounts, and the lack of any filtering. I do understand the issues and the steps I need to take to better secure the switches in service, and just need to get off my a$$ and do it. Mainly I am hoping to hear from someone who has gone through the aftermath - as you mention above. So far I have had a discussion with the carrier who is opening an investigation. I'll contact the FBI today as well. I'll send an update when this is all over for posterity. (BTW Sierra Leone is in West Africa, not the Middle East.) True ;) Most of the calls were Iraq, UAE, Lebanon... Found another one today that was 2.5 DAYS long to Chile. Bizarre. j -- _ -- 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] fraud advice
On Fri, Oct 15, 2010 at 11:50 AM, Jeff LaCoursiere j...@sunfone.com wrote: snipped (BTW Sierra Leone is in West Africa, not the Middle East.) True ;) Most of the calls were Iraq, UAE, Lebanon... Found another one today that was 2.5 DAYS long to Chile. Bizarre. j Not bizarre at all. You being in the Virgin Islands should know what that is probably about. http://www.snopes.com/fraud/telephone/809.asp I have a general questionnaire prior to planning the installation. One question is about international calls and using a PIN (Authenticate(1234356)), totally blocking, having a few phones in a separate context that can dial international. Usually, I will explain the nature of an IP PBX and the dangers of fraud, then go over what they NEED. If you do this along with locking things down, hopefully you won't run into any more fraud, but as you have seen first hand, there is big money to be made, so assume you are defending against an international crime ring with lots of time and knowledge. Once you do your bit and cover your bases, then if there is fraud, you save face and provide guidance rather than damage control. http://www.infiltrated.net/asterisk-ips.html found that link while looking googling for Nufone. It appears there is may be more to the story than I knew. I know JerJer claimed to have received a bill for $500k due to fraud. I am not sure what happened after that but I am seeing information about charges against him for mail fraud. Thanks, Steve T -- _ -- 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] SIP - no audio behind nat problem
Hello, We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this natted network. We have the issue with calls to these SIP phones - no audio. It is probably the problem with port forwarding on router - but I am not sure how can I forward same sip ports (5004 to 5100) to two phones (nat addresses?)? Any help appreciated! Z. Zivanovic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP - no audio behind nat problem
On Fri, Oct 15, 2010 at 06:22:07PM +0200, Zarko Zivanovic wrote: We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this natted network. The simplest solution will be to stick another Asterisk box inside the NAT and tunnel IAX or SIP over a VPN. R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP - no audio behind nat problem
On Friday 15 Oct 2010, Zarko Zivanovic wrote: Hello, We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this natted network. We have the issue with calls to these SIP phones - no audio. It is probably the problem with port forwarding on router - but I am not sure how can I forward same sip ports (5004 to 5100) to two phones (nat addresses?)? Simple answer, don't run SIP through NAT. Have another Asterisk server on the outside and run IAX2 through NAT instead. Much cleaner :) -- AJS -- _ -- 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] Audio problems on cable modem link
On Fri, Oct 15, 2010 at 8:53 AM, Michelle Dupuis mdup...@ocg.ca wrote: When a single call is up, call quality is fine. When a second call is up, outbound audio is immediately choppy. We're using ulaw, and confirmed that traffic with 2 calls is 175kbps in/out. (IAX connection out) Asterisk doesn't report any dropped frames, the internet connection looks fine, etc. We have a linux router in place running wondershaper that seems to be running fine (same as our other installations). Can someone suggest where to look? Could this be the ITSP? It could be your traffic shapper, the ITSP, your local network, the ISP's network, or the internet backbone - basically anywhere in-between. You only have control over your local network, so I'd start there. Look for duplex mismatches (hint: if one end is set to auto or not able to be set manually, the other end should also be auto, never full [don't worry, they'll negotiate full, but only if both ends are set to auto; otherwise, the auto end will negotiate half due to the end running full not broadcasting capabilities when hardset]). That said, I've never felt great about using the internet for phone calls - you can't controll anything else in the chain, so the possibility of problems is huge - and most of the time you can't fix it. I know lots of people here do it, but it's going to be problematic. If you want toll-quality voice, you still need either TDM lines or dedicated (non-internet) bandwidth. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP - no audio behind nat problem
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Friday, October 15, 2010 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP - no audio behind nat problem On Friday 15 Oct 2010, Zarko Zivanovic wrote: Hello, We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this natted network. We have the issue with calls to these SIP phones - no audio. It is probably the problem with port forwarding on router - but I am not sure how can I forward same sip ports (5004 to 5100) to two phones (nat addresses?)? Simple answer, don't run SIP through NAT. Have another Asterisk server on the outside and run IAX2 through NAT instead. Much cleaner :) If you ARE going to run SIP through a NAT, you're going to need to designate a chunk of ports in rtp.conf and poke those in your firewall. We did UDP pokes for 10001-10004 to use 1 line. When you do an Asterisk communication, the handshake occurs on 5060; the actual call occurs on 2-4 RTP ports usually in the 1 range. -- _ -- 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] fraud advice
On Fri, 2010-10-15 at 07:29 -0700, Steve Edwards wrote: On Thu, 14 Oct 2010, bruce bruce wrote: But it also sickens me at how badly Asterisk is made to not cope with situations like this and worse than that is FreePBX. Kind of like blaming the gun manufacturer instead of the criminal with their finger on the trigger? Is there some gaping hole in Asterisk security or are you just asleep at the wheel? Asterisk is just doing what you tell it to do, process calls. If you have no authentication or route blocking how do you expect Asterisk to know that there is a problem? I was just in a similar situation where someone guessed the username and password of my SIP trunk. The provider called me the next day to tell me that they detected strange traffic on my line and asked if I was making those calls. Now that is good service from a provider. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] drop dead fix
On Fri, Oct 15, 2010 at 9:35 AM, Danny Nicholas da...@debsinc.com wrote: Don't know if this will make acceptable GSM files, but should help with the WAV ones. Are you using GSM to talk to an ITSP (the idea of banking voice calls going across the internet makes me cringe)? If not, what are you using GSM for? GSM always sounds like garbage (and see below - it's not what you are hearing on your mobile phone! It's not as good as mobile phone codecs). If you are using GSM to save bandwidth, you should really look at a better codec - but I would think a banking system wouldn't use the internet for the voice channel. If you are using a private network and bandwidth is still a concern, I'd look at any of the other codecs (except maybe ilbc, which is even worse than GSM). Any of them would sound better. Somehow, to get to a mobile handset user (who uses GSM), the call will hit the PTSN. The PTSN, as others mentioned, is 8K alaw or ulaw (depending on your country). Get the recordings to sound good on the PTSN (convert to alaw or ulaw 8K, as that's what will happen NO MATTER WHAT when your call hits the PTSN) - don't even try to optimize anything else until then. If you're hitting the PTSN at all (versus a direct connection within an IP-based GSM provider's network - unlikely that you have this), even though the handset user is on GSM, you do NOT want to use GSM as your encoding. Use 8K alaw/ulaw (wav format). I suspect your GSM providers in your area have spent literally millions of dollars on their GSM encoding systems - let them do the work. They'll have to do it even if you played the GSM file, it'll just sound worse if you play GSM, convert it to 8K alaw/ulaw over the PTSN, then have it converted using a different algorithm at the cell site. Finally, not all mobile calls on even GSM networks are gsm format. If they are a different format, converting from one compressed algorithm (gsm) to another (whatever the carrier uses) is going to sound horrible. So don't bother with the GSM format. Few people you are calling/called-by use that codec (not even the mobile phone users). It'll get resampled into something else. You'd be better off using the raw, basically-uncompressed (I know, I know, not quite accurate) alaw/ulaw - which everyone's codecs are designed to handle very well (since every single PTSN call uses it). For reference, Asterisk uses (I believe) the full rate GSM codec. Mobile phones on most GSM networks are using an AMR (not full rate) codec, as it simply sounds better, can deal with bad connections better, and can even use less bandwidth. Of course it is licensed and patented, so Asterisk doesn't implement it. But because of this, Asterisk's gsm doesn't sound as good as a call on a GSM network. Why would you want that? Just don't use it! See http://en.wikipedia.org/wiki/Adaptive_Multi-Rate (What mobile companies use) And http://en.wikipedia.org/wiki/Full_Rate (What Asterisk uses) -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-10-15 04:10 AM, Сикорский Сергей wrote: 15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? The correct answer is to use ringinuse=no in queues.conf and callcounter=yes in sip.conf. Leif, Isn't callcounter for 1.6 and not for 1.4? -Matt -- _ -- 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] Audio problems on cable modem link
On Fri, Oct 15, 2010 at 11:07 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: - turn off IAX trunking mode I would disagree, you want to enable trunking with multiple call. It will reduce patch overhead, leading to less bandwidth. OP could enable jitterbuffer, if not already enabled. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] Queue Agent Getting Additional Calls When on the Phone
2010/10/15 Matt Darnell mattdarn...@gmail.com: On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-10-15 04:10 AM, Сикорский Сергей wrote: 15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? The correct answer is to use ringinuse=no in queues.conf and callcounter=yes in sip.conf. Leif, Isn't callcounter for 1.6 and not for 1.4? If you are using the Local channel, look into the n option. -M -- _ -- 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] Audio problems on cable modem link
Jitterbuffer affects inbound audio only, not outbound (the other side hears the choppiness) so I don't think that will help/ Trunking only reduces overhead after 4+ calls, so that shouldn't help either. (Since this occurs at 2 calls) I can't wireshark the other end since the other end is my ITSP (who says everything looks fine, no lost packets, 60ms latency) still stuck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger [paul.belan...@polybeacon.com] Sent: Friday, October 15, 2010 1:30 PM To: Asterisk Users List Subject: Re: [asterisk-users] Audio problems on cable modem link On Fri, Oct 15, 2010 at 11:07 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: - turn off IAX trunking mode I would disagree, you want to enable trunking with multiple call. It will reduce patch overhead, leading to less bandwidth. OP could enable jitterbuffer, if not already enabled. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio problems on cable modem link
On Fri, Oct 15, 2010 at 1:44 PM, Michelle Dupuis mdup...@ocg.ca wrote: Jitterbuffer affects inbound audio only, not outbound (the other side hears the choppiness) so I don't think that will help/ If your problems with audio are at the far end, I don't expect there is much you can do. Try a different codec (gsm) and see what happens. Otherwise, open a support ticket with your ITSP and have them monitor the path of the call. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] Audio problems on cable modem link
Hi! Trunking only reduces overhead after 4+ calls, so that shouldn't help either. (Since this occurs at 2 calls) Trunking requires a timing source, and you might have trouble with your timing, that is why I suggested this (and because you did not tell us wether you have trunking enabled or not). In general IAX is not as well tested as SIP, and once in a while there are interoperability issues with IAX between Asterisk versions (and/or other IAX protocol implementations). Apart from that: you could also try a different cable modem. Philipp -- _ -- 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] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)
On 10/15/2010 04:00 AM, Karsten Wemheuer wrote: I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN card. The system starts without any errors. I discovered a severe issue. The kernel panics on a very small load. The first call normally gets through. If I start the second or third call and sometimes when I terminate the first call, the system panics (Oops text on console). After solving some difficulties (the relevant part of the Oops text scrolls out of the monitor, no serial interface), I get the text via netconsole. It seems to me, that the panic occurred in oslec (function oslec_update). But maybe I am wrong with this. In the oslec code there is a patch to enable MMX. After switching this off, the problem disappeared. AFAIK the cpu supports mmx. Where should I address this issue to? Is it a known issue? Do you have CONFIG_DAHDI_MMX defined in include/dahdi/dahdi_config.h? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] Echo on PRI after upgrading to Asterisk 1.6.2 from 1.2
On 10/15/2010 10:33 AM, Jared Geiger wrote: This might be my problem?*** [r...@voice ~]# grep -E ^echo /etc/asterisk/chan_dahdi.conf * *So I added this under [channels]: echocancel=yes echocancelwhenbridged=no echotraining=800* Most likely (unless you were including another file that included that definition). Did this resolve your problem? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] Echo on PRI after upgrading to Asterisk 1.6.2 from 1.2
On Fri, Oct 15, 2010 at 9:55 AM, Jared Geiger compuw...@gmail.com wrote: I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a full reformat and recompile) and I started getting echo over the PRI. I did an update on a server last year, had the same problem. I needed to explicitly set echocancel=yes in my configs, before 1.6 it was enabled by default. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] SIP - no audio behind nat problem
On Fri, Oct 15, 2010 at 06:22:07PM +0200, Zarko Zivanovic wrote: We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this natted network. We have the issue with calls to these SIP phones - no audio. Tell us more about your settings. I have a GXP2000 behing NAT connected to an * (behind NAT) without problems and without any portforwards at the Grandstream side. nat=yes and canreinvite=no as always do the trick for me. -- Daniel Tryba -- _ -- 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] drop dead fix
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming There were some comments in other replies about your files being 'quiet' (low average volume level)... this won't help your situation at all, because it means that any artifacts caused by resampling and compression/decompression will end up at a relatively high amplitude compared to the original signal (resulting in a low signal-to-noise ratio), and when the listener increases the volume level on their listening device, the noise level will be increased along with it. For these sorts of tasks, you really do want the source material recorded at a fairly high volume level. On Fri, 15 Oct 2010, Danny Nicholas wrote: This appears to be the resolution to my problem - #1. Get my recording talent in an isolated environment so I can get clean, loud recordings #2. Dump the Audacity and Audiologic steps and just use SOX with the highpass and lowpass filters. 1) firstmenu.wav.wav is recorded so low, it just looks like line noise in Audacity. Unless you can re-record at a reasonable level, you're always going to be fighting this sow's ear. 2) I use normalize (http://normalize.nongnu.org/) to normalize from the command line, but it does not deal with DC offset like Audacity will. Eliminating your DC offset issue should also be a goal of improving your recording environment. Newer (than provided with CentOS 5.5) versions of sox can do dcshift. 3) Stick with ULAW or PCM (wav). You only have to be concerned with supplying audio encoded appropriately for the first hop in your delivery path. -- 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] Echo on PRI after upgrading to Asterisk 1.6.2 from 1.2
I haven't heard if this fixed it yet. However I was seeing the echo cancelers loaded before so I never realized I'd have to do this. Its a FreePBX install also so I checked all the include files and didn't see a reference to these values anywhere. Thanks everyone for the input, I should know soon if it is the fix. ~Jared On Fri, Oct 15, 2010 at 3:56 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Oct 15, 2010 at 9:55 AM, Jared Geiger compuw...@gmail.com wrote: I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a full reformat and recompile) and I started getting echo over the PRI. I did an update on a server last year, had the same problem. I needed to explicitly set echocancel=yes in my configs, before 1.6 it was enabled by default. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
