[asterisk-users] presence with polycom DND
hi, I have configured asterisk 1.4.21 to control the presence BLF (hint + watch buddy parameter) of Polycom phones (650,550,330) and it works good. But when I set the phones on Do Not Disturb (DND) on the server there arent sip notifications and the presence doesnt change. On the Polycom configuration I have try to use the server based DND option but i dont know how to use this with asterik. What can i do ? Are there some workaround to use the DND button and the BLF on asterisk? thanks cfh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with dahdi
On Tue, Nov 18, 2008 at 07:56:36PM -0500, Jerry Geis wrote: I am installing dahdi on a machine lspci 00:00.0 Host bridge: Advanced Micro Devices [AMD] RS780 Host Bridge 00:01.0 PCI bridge: Hewlett-Packard Company Unknown device 9602 00:04.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 0) 00:05.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 1) 00:06.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 2) 00:07.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 3) 00:11.0 SATA controller: ATI Technologies Inc SB700/SB800 SATA Controller [AHCI mode] 00:12.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller 00:12.1 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI1 Controller 00:12.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller 00:13.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller 00:13.1 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI1 Controller 00:13.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 3a) 00:14.1 IDE interface: ATI Technologies Inc SB700/SB800 IDE Controller 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia 00:14.3 ISA bridge: ATI Technologies Inc SB700/SB800 LPC host controller 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge 00:18.0 Host bridge: Advanced Micro Devices [AMD] Family 11h HyperTransport Configuration (rev 40) 00:18.1 Host bridge: Advanced Micro Devices [AMD] Family 11h Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] Family 11h DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] Family 11h Miscellaneous Control 00:18.4 Host bridge: Advanced Micro Devices [AMD] Family 11h Link Control 01:05.0 VGA compatible controller: ATI Technologies Inc RS780M/RS780MN [Radeon HD 3200 Graphics] 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller 08:00.0 System peripheral: JMicron Technologies, Inc. Unknown device 2382 08:00.2 SD Host controller: JMicron Technologies, Inc. Unknown device 2381 08:00.3 System peripheral: JMicron Technologies, Inc. Unknown device 2383 08:00.4 System peripheral: JMicron Technologies, Inc. Unknown device 2384 09:00.0 Ethernet controller: Atheros Communications Inc. AR242x 802.11abg Wireless PCI Express Adapter (rev 01) 0a:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8101E PCI Express Fast Ethernet controller (rev 02) dahdi complete 2.0.0 compiles fine. I am running centos 5.2 x86_64. the service starts fine. lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231888 1 dahdi_dummy crc_ccitt 35265 1 dahdi dahdi_dummy should be the source of timing (ticks) dahdi_dummy loads as shown. When compiling asterisk 1.4.22 it compiles fine. when running I get the message: ] ERROR[10981]: asterisk.c:3036 main: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have options: 1. You only have to compile DAHDI support into Asterisk if you need it. One option is to recompile without DAHDI support. 2. You only have to load DAHDI drivers if you want to take advantage of DAHDI services. One option is to unload DAHDI modules if you don't need them. 3. If you need DAHDI services, you must correctly configure DAHDI. dahdi_speed gives: Count: 1782120 dahdi_speed is pointless. dahdi_test never somes back DAHDI loaded. Device files exist. But nothing actually ticks. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not reading fast DTMFs, was: PBX - PRI - * - Telco not working
I plug the NEC back straight to the Telco and all works well again. I just got on the phone to Digium and we've raised a ticket with some pri intense debugging going on. I'll update the list on findings. On Wed, Nov 19, 2008 at 10:32 AM, Brent Davidson [EMAIL PROTECTED] wrote: I have a weird thought... Is the PBX possibly passing the digits both inband and via PRI signaling so Asterisk is getting two digit streams at the same time and totally freaking out? You know.. that is probably it What the NEC system is doing I think is when you pick up the POTS phone to dial, you go to the NEC's LCR program (least cost routing). It then reads the first digits of your call. When it determines how to route your call (in our case, we have made it route everything out to the PRI) it then must send the digits out via PRI signaling. Maybe it captures three digits before deciding what to do, so it sends them out via PRI signaling. It would also capture the remaining digits and send them too via PRI signaling, but then the analog phone is ALSO sending the remaining digits via inband audio and then asterisk gets the first three via pri signaling, and the last 5 via inband, and instead of putting the pri signaling first and the inband second, is interleaving it. This must be how the Telco actually managed to router the call. Because it must go 'pri signaled digits first, inband second'. Because if you take the pri signal digits (which we assume are the first three) and put them at the start, you can see the number, all in the correct sequence. Thanks for this idea, I'm going to send it off to Digium and get it added to the ticket. Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] P2P
Hello List, i would like to set up the following concept: Scenario 1: = VOIP-Phone -tcp/udp- VOIP-Phone (direct P2P between two phones. Those phones have be he hard phones. No Software such as KPhone or something) Scenario 2: = VOIP-Phone -tcp/udp- Asterisk -tcp/udp- VOIP-Phone (Those phones also have be he hard phones.) Are this scenarios possible? What hardware do i need for this? Has anyone any recommendations? I guess for Scenario 2 the Asterisk box just need a simple pc with a network card? Thanks, Mario ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forcing repacketization on SIP to SIP call
JT Once again thanks for the help on this. I have found the issue, which was, as they say, carbon based. I was getting mixed up because the default setting allow=alaw, is displayed as follows when I do sip show user : Codec Order : (ulaw:20,alaw:20,g729:20) which I thought was equivalent to having allow=alaw:20, but it is not. Setting the ACLs to alaw:20 explicitly as you described has fixed this issue. W.r.t. you comment: and in the SDP Asterisk should offer a ptime and maxptime I must add that I could not get Asterisk to send a maxptime in the SDP, nor can I find any instance of maxptime in the Asterisk source code (version 1.4.18 so it may have since been added). Thanks again! Richard On Tue, Nov 11, 2008 at 11:29 AM, Richard Brady [EMAIL PROTECTED] wrote: JT Thanks for this detailed response. It's clear I have some more homework to do before going anywhere near Mantis, but I will follow up either way. Regards, Richard On Tue, Oct 28, 2008 at 9:02 PM, John Todd [EMAIL PROTECTED] wrote: This seems like a transcoding issue, and the RTP code may not be clever enough to understand that a repacketization is transcoding and therefore lets the media flow directly and/or passes the RTP packets through without examining or modifying them. This could be an error in the way RTP transcoding is handled - put on your superhero bugtracking cape and post to Mantis! I'd suggest that you document this clearly, and put it on the bugs.digium.com system if you've tried all possible iterations of allow= and deny= for getting this media to transcode. It would seem that alaw:20 is different than alaw:40, and if you've found that they are treated as equal then there seems to be a problem. While not explicitly stated in the doc/rtp-packetization.txt file, it does seem that several things are true: - it seems that if a remote sender is sending 40ms packets, and you have not explicitly denied 40ms packets, that Asterisk should accept those packets. This seems to work. - if you explicitly have deny=all and then allow=alaw:20 in a peer definition, it should be the case that Asterisk takes whatever audio stream and transcode it for the remote peer in that format (and in the SDP Asterisk should offer a ptime and maxptime based on the default and highest ptime acceptable, in this case 20 for both.) Is this broken? - if a remote host sends you a ptime that is not defined or defaulted in the list of allow= codec choices for that peer (or globally) then the call should be refused just like it would be with any other codec mismatch. (Of course, if you don't have a deny=all as the first statement in your peer codec list, Asterisk should let anything through since that's the way those ACLs work. I mention this only as a caution for reporting problems that might not be problems.) Is this broken? This problem is actually fairly important when we start talking about scale. All RTP-based systems start to experience bottlenecks introduced by Packets-Per-Second limits on hardware interfaces. The upper limit of performance starts to be more bound to throughput on interfaces and kernel drivers, rather than in the higher-layer code. PPS, not megabits per second, becomes the number to beat. If you can get RTP packets to go from 20ms to 40ms, it doubles the size of the packet and effectively halves the number of packets you're sending on your interface, which _could_ lead to doubling of total numbers of calls as the limits of interface buffering are reached in the near future. Even if you're just doing this on one leg of a looped call, this still could reduce your overall PPS by 25%, which is nothing to sniff at. Of course, I'm assuming that the load introduced by re- packetizing different packet delays is not significant - this could be a false assumption. JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about connecting with Mobile Base Station
Hi. Probably you should use SS7. It depends on your hardware. On Wed, Nov 19, 2008 at 8:44 AM, mark morreny [EMAIL PROTECTED] wrote: Hi Andrew, Thank you for your info. I am actually looking for connecting mobile base station with asterisk via E1. Any idea on where I should start looking? Thanks, Mark On Wed, Nov 19, 2008 at 1:03 PM, Andrew Joakimsen [EMAIL PROTECTED]wrote: On Tue, Nov 18, 2008 at 22:30, mark morreny [EMAIL PROTECTED] wrote: Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. There are various devices. PCI GSM card, GSM to Ethernet, or the most basic is GSM to analog, then you connect it to asterisk with e.g. X100 card or SPA3000. Either the PCI or Ethernet devices should work very well -- since the call from the GSM network continues to be digital. An analog adapter will have a slower call setup time, can not support SMS or data and might have echo issues and by definition of a digital-to-analog and subsequent analog-to-digital conversion the quality of the call will be worse (but probably not noticeable). Here is one example: http://www.junghanns.net/en/GSM-PCI_produkt.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring
Hello all - We are trying to implement some monitoring systems for our production asterisk boxes. We use whats up gold for all our other stuff. I'd like to be able to monitor the status of PRI's. For example if a PRI is in alarm, i'd like to get an e-mail notification. How are others accomplishing this? Thanks, Jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not reading fast DTMFs, was: PBX - PRI - * - Telco not working
On Wed, Nov 19, 2008 at 9:08 PM, Hakan C [EMAIL PROTECTED] wrote: Did you try relaxdtmf = yes in your Zaptel/DAHDI conf? Yup, no difference. Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring
Hello Jon, Maybe you can think about SNMP support in Asterisk. Also you can develop custom applications in many languages or take a look to Nagios (http://www.nagios.org/) Try that command on your Asterisk box: asterisk -rx 'pri show spans', it returns PRI status. Good lucks On Wed, Nov 19, 2008 at 1:57 PM, Jon Weisman [EMAIL PROTECTED] wrote: Hello all - We are trying to implement some monitoring systems for our production asterisk boxes. We use whats up gold for all our other stuff. I'd like to be able to monitor the status of PRI's. For example if a PRI is in alarm, i'd like to get an e-mail notification. How are others accomplishing this? Thanks, Jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aeterisk NOW 1.5beta1 - CDR problem....
Hello. Enable verbosity first. core set verbosity 10 and then create a test extension: exten = _111,1,Answer exten = _111,n,MusicOnHold exten = _111,n,Hangup then try to dial 111 and hangup phone after 10 secs. and post your CDR configurations if you mind. Thanks On Wed, Nov 19, 2008 at 7:51 AM, Bipin [EMAIL PROTECTED] wrote: hello all, Is there any problem with Aeterisk NOW 1.5beta1 with the cdr logging.My *Code:* *CLI cdr status CDR logging: enabled CDR mode: simple CDR output unanswered calls: no CDR registered backend: cdr_manager CDR registered backend: cdr-custom CDR registered backend: mysql *Code:* *CLI cdr mysql status Connected to [EMAIL PROTECTED], port 3306 using table cdr for 30 minutes, 3 seconds. Wrote 0 records since last restart. shows the CDR is enabled in the CSV and in the MYSQL.But nothing is recording.I checked in the /etc/asterisk/ folder and found that there is no cdr.conf and cdr_custom.conf files.I manually added and tried and the result was same. Also there is no file called Master.csv in the asteriskcdr log.Did any body know what may be the reason?. Thanks, Bipin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Sangoma BRI with zaptel?
Hello. I've never used BRI but you can take a look to wiki.sangoma.com You can kindly ask to them for support. Good luck. On Tue, Nov 18, 2008 at 5:45 PM, Claus Herwig [EMAIL PROTECTED] wrote: Hello, there has been a post to this list somewhere arount april which said that it is possible to use a Sangoma BRI A500 card with zaptel and asterisk bristuff. That is, without sangoma_brid and sangoma_mgd daemons and without woomera channels. Could anybody give me a short hint how to configure this? I tried wanpipe-driver + zaptel + asterisk-bristuffed, but I couldn't get zaptel to recognize the sangoma channels. modprobe wanpipe did load zaptel module and others but no spans appeared in /proc/zaptel or /etc/zaptel.conf. I tried various config options of the wanpipe setup tool, but to no avail. genzaptelconf -d displays correct cardinfo but doesn't seem to get the channels. Config: debian etch with kernel 2.6.18-5-amd64 on x86_64 sangoma a503de (PCIe 6x BRI w/ Echo Cancel) asterisk 1.4.13-BRIstuffed-0.4.0-test4 (from pkg-voip.buildserver.net) zaptel 1.4.7 (from pkg-voip) wanpipe 3.3.14 (newest beta) Same config (without wanpipe of course) works well with a digium TE220 (PCIe 2x PRI). Any hints would be greatly appreciated as I'm banging my head about this for some days now ;-) Claus -- CHECON EDV-Consulting und Redaktion Claus Herwig * Barer Straße 70 * 80799 München +49 89 27826981 * Fax 27826982 * [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring
Thanks Hakan, I was kind of hoping I wouldn't have to write anything. Anybody else got something I could just use? - Original Message - From: Hakan C To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 19, 2008 7:07 AM Subject: Re: [asterisk-users] Monitoring Hello Jon, Maybe you can think about SNMP support in Asterisk. Also you can develop custom applications in many languages or take a look to Nagios (http://www.nagios.org/) Try that command on your Asterisk box: asterisk -rx 'pri show spans', it returns PRI status. Good lucks On Wed, Nov 19, 2008 at 1:57 PM, Jon Weisman [EMAIL PROTECTED] wrote: Hello all - We are trying to implement some monitoring systems for our production asterisk boxes. We use whats up gold for all our other stuff. I'd like to be able to monitor the status of PRI's. For example if a PRI is in alarm, i'd like to get an e-mail notification. How are others accomplishing this? Thanks, Jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with dahdi
On Wed, Nov 19, 2008 at 12:07:06PM +0200, Hakan C wrote: Hello, First step: Stop and Uninstall DAHDI, well theres no uninstall script in DAHDI source, so just stop it and remove kernel modules. /etc/init.d/dahdi stop then go to your /usr/src dont forget to install your kernel headers and sources, and these packages are necessary: gcc Just install build-essential g++ Not needed make libncurses5-dev flex bison Those three are not needed to build dahdi. patch Only if you need to apply a patch linux-source Not needed. linux-headers-$(uname -r) That's the one, indeed. then purge your DAHDI source, and download again. http://downloads.digium.com/pub/telephony/dahdi-linux/dahdi-linux-2.0.0.tar.gz http://downloads.digium.com/pub/telephony/dahdi-tools/dahdi-tools-2.0.0.tar.gz and go to dahdi-linux first. make make install and then go to dahdi-tools: ./configure make make install make config then restart DAHDI with: /etc/init.d/dahdi stop /etc/init.d/dahdi start dahdi_cfg -vvv (2 times) one dahdi_cfg, without those v-s will also do. Not to mention it is run by the dahdi init.d start target. And not even necessary if you just need dahdi_dummy. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IF else
Hi all,
I have the following context in extensions.conf:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,Playback(AR_GetGiveToID)
exten = _X.,22,Wait(2)
exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,24,Wait(2)
exten = _X.,25,Playback(/tmp/asterisk-recording)
exten = _X.,26,Wait(2)
exten = _X.,27,Hangup
If the customer dial 111, it'll be router to the entry with priority 21,
else it'll go to priority 2...I would like to add a third condition that if
the user dial let's say 112 it'll go to the priority 28 let's say
Regards
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Re: [asterisk-users] Monitoring
Hey Jon,
You are asking something too specific.
If you want to monitor your PRI, its not so difficult to script.
?
$checkPRI = exec(asterisk -rx 'pri show spans');
if (ereg('/^Down/', $checkPRI, $match) {
echo OMG, someone call the ambulance\r\n;
echo $match;
} else {
echo working...;
}
?
See?
It doesnt need write something huge.
Hope it helps.
Thanks.
On Wed, Nov 19, 2008 at 2:29 PM, Jon Weisman [EMAIL PROTECTED] wrote:
Thanks Hakan,
I was kind of hoping I wouldn't have to write anything. Anybody else got
something I could just use?
- Original Message -
*From:* Hakan C [EMAIL PROTECTED]
*To:* Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
*Sent:* Wednesday, November 19, 2008 7:07 AM
*Subject:* Re: [asterisk-users] Monitoring
Hello Jon,
Maybe you can think about SNMP support in Asterisk.
Also you can develop custom applications in many languages or take a look
to Nagios (http://www.nagios.org/)
Try that command on your Asterisk box:
asterisk -rx 'pri show spans', it returns PRI status.
Good lucks
On Wed, Nov 19, 2008 at 1:57 PM, Jon Weisman [EMAIL PROTECTED] wrote:
Hello all -
We are trying to implement some monitoring systems for our production
asterisk boxes. We use whats up gold for all our other stuff. I'd like to
be
able to monitor the status of PRI's. For example if a PRI is in alarm, i'd
like to get an e-mail notification. How are others accomplishing this?
Thanks,
Jon
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Re: [asterisk-users] P2P
Hey, Yeah, its possible. You just need a PC with network card and Asterisk. Read the Asterisk book, http://voipspeak.net/index.php?/content/view/33/2/ Good luck On Wed, Nov 19, 2008 at 1:10 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello List, i would like to set up the following concept: Scenario 1: = VOIP-Phone -tcp/udp- VOIP-Phone (direct P2P between two phones. Those phones have be he hard phones. No Software such as KPhone or something) Scenario 2: = VOIP-Phone -tcp/udp- Asterisk -tcp/udp- VOIP-Phone (Those phones also have be he hard phones.) Are this scenarios possible? What hardware do i need for this? Has anyone any recommendations? I guess for Scenario 2 the Asterisk box just need a simple pc with a network card? Thanks, Mario ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring
Thanks!
I'll give this a try
- Original Message -
From: Hakan C
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, November 19, 2008 8:17 AM
Subject: Re: [asterisk-users] Monitoring
Hey Jon,
You are asking something too specific.
If you want to monitor your PRI, its not so difficult to script.
?
$checkPRI = exec(asterisk -rx 'pri show spans');
if (ereg('/^Down/', $checkPRI, $match) {
echo OMG, someone call the ambulance\r\n;
echo $match;
} else {
echo working...;
}
?
See?
It doesnt need write something huge.
Hope it helps.
Thanks.
On Wed, Nov 19, 2008 at 2:29 PM, Jon Weisman [EMAIL PROTECTED] wrote:
Thanks Hakan,
I was kind of hoping I wouldn't have to write anything. Anybody else got
something I could just use?
- Original Message -
From: Hakan C
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, November 19, 2008 7:07 AM
Subject: Re: [asterisk-users] Monitoring
Hello Jon,
Maybe you can think about SNMP support in Asterisk.
Also you can develop custom applications in many languages or take a look
to Nagios (http://www.nagios.org/)
Try that command on your Asterisk box:
asterisk -rx 'pri show spans', it returns PRI status.
Good lucks
On Wed, Nov 19, 2008 at 1:57 PM, Jon Weisman [EMAIL PROTECTED] wrote:
Hello all -
We are trying to implement some monitoring systems for our production
asterisk boxes. We use whats up gold for all our other stuff. I'd like
to be
able to monitor the status of PRI's. For example if a PRI is in alarm,
i'd
like to get an e-mail notification. How are others accomplishing this?
Thanks,
Jon
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[asterisk-users] Asterisk NOW - Where to start
G'Day All, Greetings and best wishes. Many moons ago I had an Asterisk system running. Steve Totaro helped me quite a bit. Just now I installed Asterisk NOW 1.5 Beta, and am at the command prompt.I thought there was a GUI with Asterisk NOW. Anyway, where can I find the install/config documentation or how to launch the GUI, as I have look around on the site but cannot locate it. Thanks and Cheers!! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] P2P
Hi Mario, Hi Valetin, Valentin Bud wrote: Are the VoIP phone mobile on the internet or in fixed locations? If they are in fixed locations and they have to go through internet to reach the asterisk box, the way *i* would do it is with VPN tunnels. If they are in the same location (LAN) it is very simple, you just need the phones and an asterisk box with a network card as you said. You configure the phones to register with the asterisk and configure the dialplan and you are good to go. They are in the same network/lan. Can you recommend and hard phones for this task? Are there phones which can be used without asterisk in between them? I'm new in this VoIP / Asterisk business and the only hard phones i have used are Linksys SPA 901, 921, 922. Stay away from 901, they only bring problems. The 921 are very good and they even have an LCD. The 922 is something like 921 but they know PoE and the have a builtin switch so you can connect the phone to the wall plug and from the phone you connect the computer. The switch is 10/100. My wishlist for 922 would be: 1 Gig switch and the voice vlan that is used on the cisco switches so you can separate the voice traffic from the data traffic, all this if you use the builtin switch. There might be some phones that can handle calls between them without the need of a proxy (asterisk) but honestly i do not know. I repeat i am new in this business but into it :). all the best and a great day, v Thanks, Mario ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with dahdi
Hello, First step: Stop and Uninstall DAHDI, well theres no uninstall script in DAHDI source, so just stop it and remove kernel modules. /etc/init.d/dahdi stop then go to your /usr/src dont forget to install your kernel headers and sources, and these packages are necessary: gcc g++ make libncurses5-dev flex bison patch linux-source linux-headers-$(uname -r) then purge your DAHDI source, and download again. http://downloads.digium.com/pub/telephony/dahdi-linux/dahdi-linux-2.0.0.tar.gz http://downloads.digium.com/pub/telephony/dahdi-tools/dahdi-tools-2.0.0.tar.gz and go to dahdi-linux first. make make install and then go to dahdi-tools: ./configure make make install make config then restart DAHDI with: /etc/init.d/dahdi stop /etc/init.d/dahdi start dahdi_cfg -vvv (2 times) Hope this works, good luck! On Wed, Nov 19, 2008 at 10:22 AM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Tue, Nov 18, 2008 at 07:56:36PM -0500, Jerry Geis wrote: I am installing dahdi on a machine lspci 00:00.0 Host bridge: Advanced Micro Devices [AMD] RS780 Host Bridge 00:01.0 PCI bridge: Hewlett-Packard Company Unknown device 9602 00:04.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 0) 00:05.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 1) 00:06.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 2) 00:07.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 3) 00:11.0 SATA controller: ATI Technologies Inc SB700/SB800 SATA Controller [AHCI mode] 00:12.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller 00:12.1 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI1 Controller 00:12.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller 00:13.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller 00:13.1 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI1 Controller 00:13.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 3a) 00:14.1 IDE interface: ATI Technologies Inc SB700/SB800 IDE Controller 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia 00:14.3 ISA bridge: ATI Technologies Inc SB700/SB800 LPC host controller 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge 00:18.0 Host bridge: Advanced Micro Devices [AMD] Family 11h HyperTransport Configuration (rev 40) 00:18.1 Host bridge: Advanced Micro Devices [AMD] Family 11h Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] Family 11h DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] Family 11h Miscellaneous Control 00:18.4 Host bridge: Advanced Micro Devices [AMD] Family 11h Link Control 01:05.0 VGA compatible controller: ATI Technologies Inc RS780M/RS780MN [Radeon HD 3200 Graphics] 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller 08:00.0 System peripheral: JMicron Technologies, Inc. Unknown device 2382 08:00.2 SD Host controller: JMicron Technologies, Inc. Unknown device 2381 08:00.3 System peripheral: JMicron Technologies, Inc. Unknown device 2383 08:00.4 System peripheral: JMicron Technologies, Inc. Unknown device 2384 09:00.0 Ethernet controller: Atheros Communications Inc. AR242x 802.11abg Wireless PCI Express Adapter (rev 01) 0a:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8101E PCI Express Fast Ethernet controller (rev 02) dahdi complete 2.0.0 compiles fine. I am running centos 5.2 x86_64. the service starts fine. lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231888 1 dahdi_dummy crc_ccitt 35265 1 dahdi dahdi_dummy should be the source of timing (ticks) dahdi_dummy loads as shown. When compiling asterisk 1.4.22 it compiles fine. when running I get the message: ] ERROR[10981]: asterisk.c:3036 main: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have options: 1. You only have to compile DAHDI support into Asterisk if you need it. One option is to recompile without DAHDI support. 2. You only have to load DAHDI drivers if you want to take advantage of DAHDI services. One option is to unload DAHDI modules if you don't need them. 3. If you need DAHDI services, you must correctly configure DAHDI. dahdi_speed gives: Count: 1782120 dahdi_speed is pointless. dahdi_test never somes back DAHDI loaded. Device files exist. But nothing actually ticks. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] Monitoring
Hello Jon, you can see in the proc filesystem, in the same place where the zttool read. The command "cat /proc/zaptel/1 | grep -i Span" give you the status of the span 1 . You can looking for the word RED with a grep command: if it's present the span is KO. You can make a shell script and put it in crontab. Then, if the span is KO, you can use any applications to have to send you a alarm email. Giorgio Ciccarelli Jon Weisman wrote: Thanks Hakan, I was kind of hoping I wouldn't have to "write" anything. Anybody else got something I could just use? - Original Message - From: Hakan C To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 19, 2008 7:07 AM Subject: Re: [asterisk-users] Monitoring Hello Jon, Maybe you can think about SNMP support in Asterisk. Also you can develop custom applications in many languages or take a look to Nagios (http://www.nagios.org/) Try that command on your Asterisk box: asterisk -rx 'pri show spans', it returns PRI status. Good lucks On Wed, Nov 19, 2008 at 1:57 PM, Jon Weisman [EMAIL PROTECTED] wrote: Hello all - We are trying to implement some monitoring systems for our production asterisk boxes. We use whats up gold for all our other stuff. I'd like to be able to monitor the status of PRI's. For example if a PRI is in alarm, i'd like to get an e-mail notification. How are others accomplishing this? Thanks, Jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giorgio Ciccarelli Gruppo Capodarco - Area ICT Voip Ippofono Via Ostiense, 131L asc.B 00154 ROMA Cellulare Aziendale : 3454302411 "Ai sensi e per effetti della legge sulla tutela della riservatezza personale (D.lgs n. 196/2003), questa @mail e' destinata unicamente alle persone sopra indicate e le informazioni in essa contenute sono da considerarsi strettamente riservate. E' proibito leggere, copiare, usare o diffondere il contenuto della presente @mail senza autorizzazione. Se avete ricevuto questo messaggio per errore, siete pregati di rispedire la stessa al mittente. Grazie" ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not reading fast DTMFs, was: PBX - PRI - * - Telco not working
Hi, Did you try relaxdtmf = yes in your Zaptel/DAHDI conf? On Wed, Nov 19, 2008 at 11:46 AM, Mikel Lindsaar [EMAIL PROTECTED] wrote: I plug the NEC back straight to the Telco and all works well again. I just got on the phone to Digium and we've raised a ticket with some pri intense debugging going on. I'll update the list on findings. On Wed, Nov 19, 2008 at 10:32 AM, Brent Davidson [EMAIL PROTECTED] wrote: I have a weird thought... Is the PBX possibly passing the digits both inband and via PRI signaling so Asterisk is getting two digit streams at the same time and totally freaking out? You know.. that is probably it What the NEC system is doing I think is when you pick up the POTS phone to dial, you go to the NEC's LCR program (least cost routing). It then reads the first digits of your call. When it determines how to route your call (in our case, we have made it route everything out to the PRI) it then must send the digits out via PRI signaling. Maybe it captures three digits before deciding what to do, so it sends them out via PRI signaling. It would also capture the remaining digits and send them too via PRI signaling, but then the analog phone is ALSO sending the remaining digits via inband audio and then asterisk gets the first three via pri signaling, and the last 5 via inband, and instead of putting the pri signaling first and the inband second, is interleaving it. This must be how the Telco actually managed to router the call. Because it must go 'pri signaled digits first, inband second'. Because if you take the pri signal digits (which we assume are the first three) and put them at the start, you can see the number, all in the correct sequence. Thanks for this idea, I'm going to send it off to Digium and get it added to the ticket. Mikel -- http://lindsaar.net/ Rails, RSpec and Life blog ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring
On Wed, 19 Nov 2008 15:17:50 +0200
Hakan C [EMAIL PROTECTED] wrote:
Hey Jon,
You are asking something too specific.
If you want to monitor your PRI, its not so difficult to script.
?
$checkPRI = exec(asterisk -rx 'pri show spans');
if (ereg('/^Down/', $checkPRI, $match) {
echo OMG, someone call the ambulance\r\n;
echo $match;
} else {
echo working...;
}
?
Or better (imo):
?
$checkPRI = exec(asterisk -rx 'pri show spans');
if (ereg('/^Up/', $checkPRI, $match) {
echo working...;
} else {
echo OMG, someone call the ambulance\r\n;
echo $match;
}
?
Bye
Federico Fetto
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Re: [asterisk-users] P2P
Hi Valetin, Valentin Bud wrote: Are the VoIP phone mobile on the internet or in fixed locations? If they are in fixed locations and they have to go through internet to reach the asterisk box, the way *i* would do it is with VPN tunnels. If they are in the same location (LAN) it is very simple, you just need the phones and an asterisk box with a network card as you said. You configure the phones to register with the asterisk and configure the dialplan and you are good to go. They are in the same network/lan. Can you recommend and hard phones for this task? Are there phones which can be used without asterisk in between them? Thanks, Mario ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
Then in the hope of stopping searchers following up again: Ssh/telnet to it, log in and type ctrl-z. That gets you a basic CLI. Type '?' for help if you like. Now type lcli and login again when prompted. Now you have a proper CLI with comfortingly IOS-like commands to configure *everything*. Mike On Mon, Nov 17, 2008 at 11:44:33PM -0700, Jesse Molina wrote: Digging up an old issue here, so please disregard. I'm making this statement for historical and searches. I own a couple of Linksys SRW series switches. The modern/updated firmwares on multiple models as of this writing are MSIE v6 compatible only. They will not work with Safari, Firefox/Seamonkey, or even MSIE v7. However, Linksys does not make firmware across models or series standard in any way, so one unit might work with one browser, and another mostly-similar unit may not. Please see the Linksys message boards for more info about this issue. It's a fairly well known gripe from Linksys customers. -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ -- Mike JagdisWeb: http://www.eris-associates.co.uk Eris Associates LimitedTel: +44 7780 608 368 Reading, England Fax: +44 118 926 6974 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Role of asterisk
Hello list, When you have an asterisk box connected between the VoIP phones and an PSTN gateway what is the role of asterisk. Proxy server: stateful or stateless? From what i read in the: Understanding the SIP, second edition from Alan B. Johnston i think that asterisk is a stateful proxy server as well as registration server. Am I right? Can asterisk be configured to work as redirect server or stateless proxy or i am totally in the dark and don't understand correctly? And if you know other (better) books for SIP please tell me. thank you and have a great day, v ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Picked up calls die in exactly 20 seconds
Hello. Set your verbosity to 10 with 'core set verbosity 10' and put a test call, paste your outputs. Thanks. On Tue, Nov 18, 2008 at 3:00 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Mon, Nov 17, 2008 at 6:04 PM, Juan Carlos Castro y Castro [EMAIL PROTECTED] wrote: Weird thing happening when a call is picked up. Whether by *8 feature, or by directed pickup via dialplan, either with Pickup() or with Pickup2(), the same thing happens: the call is picked up successfully, and after exactly 20 seconds talking, the call is terminated. The originating end gets a hangup, while the side that did the pickup goes mute. Anyone experienced anything similar? Throw an answer() in after pickup() and see if it still does the same. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk conference
Hello. Use 'core show application MeetMe' and see how meetme works, may its makes a sense. On Mon, Nov 17, 2008 at 4:43 PM, Giedrius Augys [EMAIL PROTECTED] wrote: Hello, I've asterisk 1.4.22. I need to that the first conference user hears You're the only conference user... . When the second user joins (without recording his name) , the first user only hears new user have join , when the third user joins to conference, others hear new user have join and so on. I'll try to do this with meetme, but it always ask me for recording user name So is it possible to do that with meetme, or use another conference application? thanks -- Pagarbiai / Best Regards, Giedrius Augys ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] P2P
On Wed, Nov 19, 2008 at 1:10 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello List, i would like to set up the following concept: Scenario 1: = VOIP-Phone -tcp/udp- VOIP-Phone (direct P2P between two phones. Those phones have be he hard phones. No Software such as KPhone or something) Scenario 2: = VOIP-Phone -tcp/udp- Asterisk -tcp/udp- VOIP-Phone (Those phones also have be he hard phones.) Are the VoIP phone mobile on the internet or in fixed locations? If they are in fixed locations and they have to go through internet to reach the asterisk box, the way *i* would do it is with VPN tunnels. If they are in the same location (LAN) it is very simple, you just need the phones and an asterisk box with a network card as you said. You configure the phones to register with the asterisk and configure the dialplan and you are good to go. a great day, v Are this scenarios possible? What hardware do i need for this? Has anyone any recommendations? I guess for Scenario 2 the Asterisk box just need a simple pc with a network card? Thanks, Mario ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with dahdi
/ // // dahdi_dummy loads as shown. // // When compiling asterisk 1.4.22 it compiles fine. // // when running I get the message: // ] ERROR[10981]: asterisk.c:3036 main: Asterisk has detected a problem // with your DAHDI configuration and will shutdown for your protection. // You have options: // 1. You only have to compile DAHDI support into Asterisk if you // need it. One option is to recompile without DAHDI support. // 2. You only have to load DAHDI drivers if you want to take // advantage of DAHDI services. One option is to unload DAHDI modules if // you don't need them. // 3. If you need DAHDI services, you must correctly configure DAHDI. // // // dahdi_speed gives: // Count: 1782120 / dahdi_speed is pointless. / // dahdi_test never somes back / DAHDI loaded. Device files exist. But nothing actually ticks. Still investigating DAHDI... more /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) I did the following on suggestions from the list. Verified that the packages: gcc g++ make libncurses5-dev flex bison patch linux-source linux-headers-$(uname -r) are indeed present on my system. service dahdi stop rm -rf /usr/include/dahdi rm -rf /lib/modules/2.6.18-92.el5/dahdi rm /etc/udev/rules.d/dahdi.rules remove my source tree for DAHDI. grabbed the linux-complete 2.0 again. extracted it. according to the readme in the complete package, I did the make all, make install, make config. then I rebooted. asterisk gives me the same error about DAHDI is misconfigured. ERROR[9878]: asterisk.c:3036 main: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have options: 1. You only have to compile DAHDI support into Asterisk if you need it. One option is to recompile without DAHDI support. 2. You only have to load DAHDI drivers if you want to take advantage of DAHDI services. One option is to unload DAHDI modules if you don't need them. 3. If you need DAHDI services, you must correctly configure DAHDI. the /proc/dahdi/1 shows its using the RTC Whats my next step. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IF else
On Wed, 19 Nov 2008, michel freiha wrote:
Hi all,
I have the following context in extensions.conf:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,Playback(AR_GetGiveToID)
exten = _X.,22,Wait(2)
exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,24,Wait(2)
exten = _X.,25,Playback(/tmp/asterisk-recording)
exten = _X.,26,Wait(2)
exten = _X.,27,Hangup
If the customer dial 111, it'll be router to the entry with priority 21,
else it'll go to priority 2...I would like to add a third condition that if
the user dial let's say 112 it'll go to the priority 28 let's say
1. Stop using numbers.
2. Start using labels.
3. Add comments.
exten = _X.,1,Gotoif($[${EXTEN} = 111]?exten111)
exten = _X.,n,Gotoif($[${EXTEN} = 112]?exten112)
exten = _X.,n,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
exten = _X.,n(exten111),Noop(Dialled 111)
exten = _X.,n,Playback(AR_GetGiveToID)
exten = _X.,n,Wait(2)
exten = _X.,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,n,Wait(2)
exten = _X.,n,Playback(/tmp/asterisk-recording)
exten = _X.,n,Wait(2)
exten = _X.,n,Hangup
exten = _X.,n(exten112),Noop(Dialed 112)
exten = _X.,n,Playback(AR_GetGiveToID)
exten = _X.,n,Wait(2)
exten = _X.,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,n,Wait(2)
exten = _X.,n,Playback(/tmp/asterisk-recording)
exten = _X.,n,Wait(2)
exten = _X.,n,Hangup
Gordon
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Re: [asterisk-users] Asterisk NOW - Where to start - FOUND, Thanks
http://brvmlaw.com/ Michael E. Ferguson, I.T. Director | Bio | V Card http://brvmlaw.com/fergusonm.vcf Berman Rennert Vogel Mandler, P.A. 100 SE 2nd Street, 29th Floor | Miami, Fl. 33131 (305.423.3408 Direct | (305.533.1582 Fax | * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] This message is for the named person's use only. It may contain confidential, proprietary or legally privileged information. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this message in error, please immediately delete it and all copies of it from your system, destroy any hard copies of it and notify the sender. You must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message if you are not the intended recipient. BERMAN RENNERT VOGEL MANDLER, P.A. reserve the right to monitor all e-mail communications through its networks. Any views expressed in this message are those of the individual sender, except where the message states otherwise and the sender is authorized to state them to be the views of any such entity. CIRCULAR 230 NOTICE: To comply with U.S. Treasury Department and IRS regulations, we are required to advise you that, unless expressly stated otherwise, any U.S. federal tax advice contained in this transmittal, is not intended or written to be used, and cannot be used, by any person for the purpose of (i) avoiding penalties under the U.S. Internal Revenue Code, or (ii) promoting, marketing or recommending to another party any transaction or matter addressed in this e-mail or attachment. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ferguson, Michael Sent: Wednesday, November 19, 2008 8:39 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk NOW - Where to start G'Day All, Greetings and best wishes. Many moons ago I had an Asterisk system running. Steve Totaro helped me quite a bit. Just now I installed Asterisk NOW 1.5 Beta, and am at the command prompt.I thought there was a GUI with Asterisk NOW. Anyway, where can I find the install/config documentation or how to launch the GUI, as I have look around on the site but cannot locate it. Thanks and Cheers!! siglogo.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Role of asterisk
On Wed, 2008-11-19 at 12:33 +0200, Valentin Bud wrote: When you have an asterisk box connected between the VoIP phones and an PSTN gateway what is the role of asterisk. Proxy server: stateful or stateless? Close, but not quite. Actually, Asterisk is what we call a back-to-back user agent. The most basic difference between a proxy and a back-to-back user agent is that with a proxy, a single call gets passed *through* the proxy and on to the destination. The proxy is not the destination of the call. With a back-to-back user agent, Asterisk is the destination of one call (in this case, the VoIP call), and then it creates a whole new call on the other side (to the PSTN in this case). It then acts as an endpoint to both calls, and sits in the middle and bridges the two calls, all while doing any necessary protocol or codec conversion between the two calls. Make sense? -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Role of asterisk
On Wed, 2008-11-19 at 12:33 +0200, Valentin Bud wrote: i think that asterisk is a stateful proxy server as well as registration server. To answer the second portion of your question (which I forgot to do in my earlier email)... yes, Asterisk can be a registration server as well. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with dahdi
Post your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf On Wed, Nov 19, 2008 at 3:16 PM, Jerry Geis [EMAIL PROTECTED] wrote: / // // dahdi_dummy loads as shown. // // When compiling asterisk 1.4.22 it compiles fine. // // when running I get the message: // ] ERROR[10981]: asterisk.c:3036 main: Asterisk has detected a problem // with your DAHDI configuration and will shutdown for your protection. // You have options: // 1. You only have to compile DAHDI support into Asterisk if you // need it. One option is to recompile without DAHDI support. // 2. You only have to load DAHDI drivers if you want to take // advantage of DAHDI services. One option is to unload DAHDI modules if // you don't need them. // 3. If you need DAHDI services, you must correctly configure DAHDI. // // // dahdi_speed gives: // Count: 1782120 / dahdi_speed is pointless. / // dahdi_test never somes back / DAHDI loaded. Device files exist. But nothing actually ticks. Still investigating DAHDI... more /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) I did the following on suggestions from the list. Verified that the packages: gcc g++ make libncurses5-dev flex bison patch linux-source linux-headers-$(uname -r) are indeed present on my system. service dahdi stop rm -rf /usr/include/dahdi rm -rf /lib/modules/2.6.18-92.el5/dahdi rm /etc/udev/rules.d/dahdi.rules remove my source tree for DAHDI. grabbed the linux-complete 2.0 again. extracted it. according to the readme in the complete package, I did the make all, make install, make config. then I rebooted. asterisk gives me the same error about DAHDI is misconfigured. ERROR[9878]: asterisk.c:3036 main: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have options: 1. You only have to compile DAHDI support into Asterisk if you need it. One option is to recompile without DAHDI support. 2. You only have to load DAHDI drivers if you want to take advantage of DAHDI services. One option is to unload DAHDI modules if you don't need them. 3. If you need DAHDI services, you must correctly configure DAHDI. the /proc/dahdi/1 shows its using the RTC Whats my next step. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring
is this for php?
- Original Message -
From: federico fetto [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 19, 2008 8:41 AM
Subject: Re: [asterisk-users] Monitoring
On Wed, 19 Nov 2008 15:17:50 +0200
Hakan C [EMAIL PROTECTED] wrote:
Hey Jon,
You are asking something too specific.
If you want to monitor your PRI, its not so difficult to script.
?
$checkPRI = exec(asterisk -rx 'pri show spans');
if (ereg('/^Down/', $checkPRI, $match) {
echo OMG, someone call the ambulance\r\n;
echo $match;
} else {
echo working...;
}
?
Or better (imo):
?
$checkPRI = exec(asterisk -rx 'pri show spans');
if (ereg('/^Up/', $checkPRI, $match) {
echo working...;
} else {
echo OMG, someone call the ambulance\r\n;
echo $match;
}
?
Bye
Federico Fetto
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Re: [asterisk-users] help with dahdi
On Wed, Nov 19, 2008 at 04:57:08PM +0200, Hakan C wrote: Post your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf Nither are needed for dahdi_dummy. http://bugs.digium.com/view.php?id=13930 (more infor requested) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IF else
On Wed, Nov 19, 2008 at 4:05 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Wed, 19 Nov 2008, michel freiha wrote:
Hi all,
I have the following context in extensions.conf:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,Playback(AR_GetGiveToID)
exten = _X.,22,Wait(2)
exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,24,Wait(2)
exten = _X.,25,Playback(/tmp/asterisk-recording)
exten = _X.,26,Wait(2)
exten = _X.,27,Hangup
If the customer dial 111, it'll be router to the entry with priority 21,
else it'll go to priority 2...I would like to add a third condition that if
the user dial let's say 112 it'll go to the priority 28 let's say
1. Stop using numbers.
2. Start using labels.
3. Add comments.
exten = _X.,1,Gotoif($[${EXTEN} = 111]?exten111)
exten = _X.,n,Gotoif($[${EXTEN} = 112]?exten112)
exten = _X.,n,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
exten = _X.,n(exten111),Noop(Dialled 111)
exten = _X.,n,Playback(AR_GetGiveToID)
exten = _X.,n,Wait(2)
exten = _X.,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,n,Wait(2)
exten = _X.,n,Playback(/tmp/asterisk-recording)
exten = _X.,n,Wait(2)
exten = _X.,n,Hangup
exten = _X.,n(exten112),Noop(Dialed 112)
exten = _X.,n,Playback(AR_GetGiveToID)
exten = _X.,n,Wait(2)
exten = _X.,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,n,Wait(2)
exten = _X.,n,Playback(/tmp/asterisk-recording)
exten = _X.,n,Wait(2)
exten = _X.,n,Hangup
1) Start using AEL (remove this context from extensions.conf and add
to extensions.ael):
context a2billing {
_X. = {
if(${EXTEN}=111) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else if(${EXTEN}=112) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else {
DeadAGI(a2billing.php);
Wait(2)
Hangup();
}
}
2) Start using extension masks (also works with AEL):
[a2billing]
exten = _111,1,Noop(Dialled 111)
exten = _111,n,Playback(AR_GetGiveToID)
exten = _111,n,Wait(2)
exten = _111,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _111,n,Wait(2)
exten = _111,n,Playback(/tmp/asterisk-recording)
exten = _111,n,Wait(2)
exten = _111,n,Hangup
exten = _112,1,Noop(Dialed 112)
exten = _112,n,Playback(AR_GetGiveToID)
exten = _112,n,Wait(2)
exten = _112,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _112,n,Wait(2)
exten = _112,n,Playback(/tmp/asterisk-recording)
exten = _112,n,Wait(2)
exten = _112,n,Hangup
exten = _X.,1,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Monitoring
Thanks I can work with this. -Jon - Original Message - From: Giorgio Ciccarelli To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 19, 2008 8:36 AM Subject: Re: [asterisk-users] Monitoring Hello Jon, you can see in the proc filesystem, in the same place where the zttool read. The command cat /proc/zaptel/1 | grep -i Span give you the status of the span 1 . You can looking for the word RED with a grep command: if it's present the span is KO. You can make a shell script and put it in crontab. Then, if the span is KO, you can use any applications to have to send you a alarm email. Giorgio Ciccarelli Jon Weisman wrote: Thanks Hakan, I was kind of hoping I wouldn't have to write anything. Anybody else got something I could just use? - Original Message - From: Hakan C To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 19, 2008 7:07 AM Subject: Re: [asterisk-users] Monitoring Hello Jon, Maybe you can think about SNMP support in Asterisk. Also you can develop custom applications in many languages or take a look to Nagios (http://www.nagios.org/) Try that command on your Asterisk box: asterisk -rx 'pri show spans', it returns PRI status. Good lucks On Wed, Nov 19, 2008 at 1:57 PM, Jon Weisman [EMAIL PROTECTED] wrote: Hello all - We are trying to implement some monitoring systems for our production asterisk boxes. We use whats up gold for all our other stuff. I'd like to be able to monitor the status of PRI's. For example if a PRI is in alarm, i'd like to get an e-mail notification. How are others accomplishing this? Thanks, Jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giorgio Ciccarelli Gruppo Capodarco - Area ICT Voip Ippofono Via Ostiense, 131L asc.B 00154 ROMA Cellulare Aziendale : 3454302411 Ai sensi e per effetti della legge sulla tutela della riservatezza personale (D.lgs n. 196/2003), questa @mail e' destinata unicamente alle persone sopra indicate e le informazioni in essa contenute sono da considerarsi strettamente riservate. E' proibito leggere, copiare, usare o diffondere il contenuto della presente @mail senza autorizzazione. Se avete ricevuto questo messaggio per errore, siete pregati di rispedire la stessa al mittente. Grazie -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring
Hi,
the command asterisk -rx 'pri show spans' on asterisk 1.2
doesn't work, work only on asterisk 1.4
=-O
federico fetto wrote:
On Wed, 19 Nov 2008 15:17:50 +0200
"Hakan C" [EMAIL PROTECTED] wrote:
Hey Jon,
You are asking something too specific.
If you want to monitor your PRI, its not so difficult to script.
?
$checkPRI = exec("asterisk -rx 'pri show spans'");
if (ereg('/^Down/', $checkPRI, $match) {
echo "OMG, someone call the ambulance\r\n";
echo $match;
} else {
echo "working...";
}
?
Or better (imo):
?
$checkPRI = exec("asterisk -rx 'pri show spans'");
if (ereg('/^Up/', $checkPRI, $match) {
echo "working...";
} else {
echo "OMG, someone call the ambulance\r\n";
echo $match;
}
?
Bye
Federico Fetto
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Re: [asterisk-users] help with dahdi
On Wed, Nov 19, 2008 at 04:57:08PM +0200, Hakan C wrote: / Post your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf / Nither are needed for dahdi_dummy. http://bugs.digium.com/view.php?id=13930 (more infor requested) ok - so I have a repeatable one here. how can I help to find a resolution. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with or without OpenSER
Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. Our long term goal is to use some sort of SBC for sip registrations, call routing, maybe even basic applications like voicemail and use Asterisk for media gateways, maybe transcoding, etc. Am I completely missing the mark as to whether freeswitch can do this sort of thing or is there a 'better' way to do it. Thanks! Look into FreeSwitch. http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ On Tue, Nov 18, 2008 at 7:29 AM, Yehavi Bourvine [EMAIL PROTECTED] wrote: Hello, I am running a small installation of asterisk and looking for future expansion of it to handle thousands of users. From what I read I see that usually large installation place OpenSER (or similar solution) in front of Asterisk in order to provide high call rate because OpenSER does only signalling while Asterisk does all. My question is: If Asterisk also does only signalling (i.e. the voice traffic goes directly between the phones and not via asterisk) is it still that slow? I preffer to have one software package rather than dealing with two. Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring
Hi Jon,
Imo is better if you tag all possible states
differently than ok with a warning/error msg.
Example: unknown $checkPRI output.
Bye
Federico
On Wed, 19 Nov 2008 10:10:30 -0500
Jon Weisman [EMAIL PROTECTED] wrote:
is this for php?
- Original Message -
From: federico fetto [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 19, 2008 8:41 AM
Subject: Re: [asterisk-users] Monitoring
On Wed, 19 Nov 2008 15:17:50 +0200
Hakan C [EMAIL PROTECTED] wrote:
Hey Jon,
You are asking something too specific.
If you want to monitor your PRI, its not so difficult to script.
?
$checkPRI = exec(asterisk -rx 'pri show spans');
if (ereg('/^Down/', $checkPRI, $match) {
echo OMG, someone call the ambulance\r\n;
echo $match;
} else {
echo working...;
}
?
Or better (imo):
?
$checkPRI = exec(asterisk -rx 'pri show spans');
if (ereg('/^Up/', $checkPRI, $match) {
echo working...;
} else {
echo OMG, someone call the ambulance\r\n;
echo $match;
}
?
Bye
Federico Fetto
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Re: [asterisk-users] Asterisk with or without OpenSER
You should ask on the proper FS venue. I typically Don't Believe the Hype about anything until I can lab it up and take it for a test drive and prove it out. Google can be a good tool too, but you have to wade through propaganda to get to other people's real world experiences. My only advice, since I do not monitor FS venues is to check it out, I think FS is a snowball gaining momentum. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Wed, Nov 19, 2008 at 10:28 AM, mail-lists [EMAIL PROTECTED] wrote: Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. Our long term goal is to use some sort of SBC for sip registrations, call routing, maybe even basic applications like voicemail and use Asterisk for media gateways, maybe transcoding, etc. Am I completely missing the mark as to whether freeswitch can do this sort of thing or is there a 'better' way to do it. Thanks! Look into FreeSwitch. http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ On Tue, Nov 18, 2008 at 7:29 AM, Yehavi Bourvine [EMAIL PROTECTED] wrote: Hello, I am running a small installation of asterisk and looking for future expansion of it to handle thousands of users. From what I read I see that usually large installation place OpenSER (or similar solution) in front of Asterisk in order to provide high call rate because OpenSER does only signalling while Asterisk does all. My question is: If Asterisk also does only signalling (i.e. the voice traffic goes directly between the phones and not via asterisk) is it still that slow? I preffer to have one software package rather than dealing with two. Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with or without OpenSER
mail-lists wrote: Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. What do you mean by functionality? Are you looking for low-level or high-level functionality? Also, XML is not a reasonable format for config files. I don't know what sipping-the-property-file-Kool-Aid J2EE droids decided that, but it's made me like UNIX a lot less than I did before now that they're proliferating. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IF else
On Wed, 19 Nov 2008, Atis Lezdins wrote:
1) Start using AEL (remove this context from extensions.conf and add
to extensions.ael):
context a2billing {
_X. = {
if(${EXTEN}=111) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else if(${EXTEN}=112) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else {
DeadAGI(a2billing.php);
Wait(2)
Hangup();
}
}
You're missing a couple of semi-colons.
2) Start using extension masks (also works with AEL):
[a2billing]
exten = _111,1,Noop(Dialled 111)
exten = _111,n,Playback(AR_GetGiveToID)
exten = _111,n,Wait(2)
exten = _111,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _111,n,Wait(2)
exten = _111,n,Playback(/tmp/asterisk-recording)
exten = _111,n,Wait(2)
exten = _111,n,Hangup
exten = _112,1,Noop(Dialed 112)
exten = _112,n,Playback(AR_GetGiveToID)
exten = _112,n,Wait(2)
exten = _112,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _112,n,Wait(2)
exten = _112,n,Playback(/tmp/asterisk-recording)
exten = _112,n,Wait(2)
exten = _112,n,Hangup
exten = _X.,1,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
And, just in case the 2 extensions really are supposed to do the exact
same thing, use extension pattern matching:
context a2billing
{
_11[12] =
{
playback(AR_GetGiveToID);
wait(2);
record(/tmp/asterisk-recording:ulaw,,5);
wait(2);
playback(/tmp/asterisk-recording);
wait(2);
hangup();
};
_x. =
{
deadagi(a2billing.php);
wait(2);
hangup();
};
};
(The above is my first attempt at AEL. It parses, but it hasn't actually
been tested.)
I would question the use of deadagi() in a non-h extension. Are signals
not being trapped correctly in a2billing.php?
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] P2P
Many phones can do direct dialing. I just tried it with the Grandstream GXP280 phones I am using to test an application I am developing. For this test I have the phones connected to each other via an Ethernet patch cable. Both phones have IP addresses on the same network. On either phone I select the option to do Direct IP call and enter the IP address of the phone I want to call. No asterisk or anything involved. Here is info from the manual: Making Calls using IP Addresses Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy. VoIP calls can be made between two phones if: Both phones have public IP addresses, or Both phones are on a same LAN/VPN using private or public IP addresses, or Both phones can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ) To make a direct IP call, please follow these steps: 1. Press MENU button to bring up MAIN MENU. 2. Select ³Direct IP Call² using the arrow-keys. 3. Press OK to select. 4. Input the 12-digit target IP address. (Please see example below). 5. Press OK key to initiate call. To make a quick IP call, please see next section. For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), input the following: 192*168*1*60#5062 - The ³ * ² key represent the dot³.² ; The ³#² key represent colon ³:². Press OK to dial out. Quick IP Call Mode The GXP also supports Quick IP call mode. This enables the phone to make direct IP-calls, using only the last few digits (last octet) of the target phone¹s IP-number. This is possible only if both phones are in under the same LAN/VPN. This simulates a PBX function using the CMSA/CD without a SIP server. Controlled static IP usage is recommended. Setting up the phone to make Quick IP calls To enable Quick IP calls, the phone has to be setup first. This is done through the web-setup function. In the ³Advanced Settings² page, set the Use Quick IP-call mode to YES. When #xxx is dialed, where x is 0-9 and xxx =255, a direct IP call to aaa.bbb.ccc.XXX is completed. ³aaa.bbb.ccc² is from the local IP address regardless of subnet mask. The numbers #xx or #x are also valid. The leading 0 is not required (but OK). For example: 192.168.0.2 calling 192.168.0.3 -- dial #3 follow by SEND or # 192.168.0.2 calling 192.168.0.23 -- dial #23 follow by SEND or # 192.168.0.2 calling 192.168.0.123 -- dial #123 follow by SEND or # 192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3 NOTE: If you have a SIP Server configured, a Direct IP-IP still works. If you are using STUN, the Direct IP-IP call will also use STUN. Configure the ³Use Random Port² to ³NO² when completing Direct IP calls. -- Jim Dickenson mailto:[EMAIL PROTECTED] CfMC http://www.cfmc.com/ From: Valentin Bud [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 19 Nov 2008 15:15:30 +0200 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] P2P Hi Mario, Hi Valetin, Valentin Bud wrote: Are the VoIP phone mobile on the internet or in fixed locations? If they are in fixed locations and they have to go through internet to reach the asterisk box, the way *i* would do it is with VPN tunnels. If they are in the same location (LAN) it is very simple, you just need the phones and an asterisk box with a network card as you said. You configure the phones to register with the asterisk and configure the dialplan and you are good to go. They are in the same network/lan. Can you recommend and hard phones for this task? Are there phones which can be used without asterisk in between them? I'm new in this VoIP / Asterisk business and the only hard phones i have used are Linksys SPA 901, 921, 922. Stay away from 901, they only bring problems. The 921 are very good and they even have an LCD. The 922 is something like 921 but they know PoE and the have a builtin switch so you can connect the phone to the wall plug and from the phone you connect the computer. The switch is 10/100. My wishlist for 922 would be: 1 Gig switch and the voice vlan that is used on the cisco switches so you can separate the voice traffic from the data traffic, all this if you use the builtin switch. There might be some phones that can handle calls between them without the need of a proxy (asterisk) but honestly i do not know. I repeat i am new in this business but into it :). all the best and a great day, v Thanks, Mario ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --
Re: [asterisk-users] IF else
On Wed, Nov 19, 2008 at 6:51 PM, Steve Edwards
[EMAIL PROTECTED] wrote:
On Wed, 19 Nov 2008, Atis Lezdins wrote:
1) Start using AEL (remove this context from extensions.conf and add
to extensions.ael):
context a2billing {
_X. = {
if(${EXTEN}=111) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else if(${EXTEN}=112) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else {
DeadAGI(a2billing.php);
Wait(2)
Hangup();
}
}
You're missing a couple of semi-colons.
Sorry, that was untested proof of options :)
2) Start using extension masks (also works with AEL):
[a2billing]
exten = _111,1,Noop(Dialled 111)
exten = _111,n,Playback(AR_GetGiveToID)
exten = _111,n,Wait(2)
exten = _111,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _111,n,Wait(2)
exten = _111,n,Playback(/tmp/asterisk-recording)
exten = _111,n,Wait(2)
exten = _111,n,Hangup
exten = _112,1,Noop(Dialed 112)
exten = _112,n,Playback(AR_GetGiveToID)
exten = _112,n,Wait(2)
exten = _112,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _112,n,Wait(2)
exten = _112,n,Playback(/tmp/asterisk-recording)
exten = _112,n,Wait(2)
exten = _112,n,Hangup
exten = _X.,1,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
And, just in case the 2 extensions really are supposed to do the exact
same thing, use extension pattern matching:
context a2billing
{
_11[12] =
{
playback(AR_GetGiveToID);
wait(2);
record(/tmp/asterisk-recording:ulaw,,5);
wait(2);
playback(/tmp/asterisk-recording);
wait(2);
hangup();
};
_x. =
{
deadagi(a2billing.php);
wait(2);
hangup();
};
};
(The above is my first attempt at AEL. It parses, but it hasn't actually
been tested.)
I would question the use of deadagi() in a non-h extension. Are signals
not being trapped correctly in a2billing.php?
AFAIK that's how a2billing is built, it's intentionally DeadAGI on
live channel. Ugly hack that gives warnings all the time in logs, but
it works and seems to provide correct billing info :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Asterisk not reading fast DTMFs, was: PBX - PRI - * - Telco not working
Mikel Lindsaar wrote: This must be how the Telco actually managed to router the call. Because it must go 'pri signaled digits first, inband second'. Because if you take the pri signal digits (which we assume are the first three) and put them at the start, you can see the number, all in the correct sequence. Thanks for this idea, I'm going to send it off to Digium and get it added to the ticket. Mikel It could also be possible that the NEC eventually sends the remaining digits via PRI signalling, but at that point Asterisk has already hit a pattern match with the inband + interleaved signaled digits so asterisk never sees the remaining signaled digits. If we had a DTMF= setting for zap channels like we do for sip, you could just turn off Asterisk's inband dtmf processing for the PRI to the NEC. -Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Howto grab back call transfered from SIP phone
Once in a while, someone mis-dials when transfering a call on their Snom SIP phone (using the Transfer button). Instead of sending them to, say, 1940; they mistype and enter 194 or 190 or somesuch. This ends up on the PSTN (for which three digit calls are valid); not what anyone wanted. On our old PBX (Network Alchemy Argent Office) there was a dialcode that grabbed back the last call that went through your extension - very useful when you realised what you'd done. Is there any way of programming this in Asterisk? I've googled to no avail :-( -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with or without OpenSER
On Wed, 19 Nov 2008, Alex Balashov wrote: mail-lists wrote: Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. What do you mean by functionality? Are you looking for low-level or high-level functionality? Also, XML is not a reasonable format for config files. I don't know what sipping-the-property-file-Kool-Aid J2EE droids decided that, but it's made me like UNIX a lot less than I did before now that they're proliferating. The downhill slide started someone someone thought it was a good idea to put curly brackets into config files... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with or without OpenSER
Gordon Henderson wrote: On Wed, 19 Nov 2008, Alex Balashov wrote: mail-lists wrote: Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. What do you mean by functionality? Are you looking for low-level or high-level functionality? Also, XML is not a reasonable format for config files. I don't know what sipping-the-property-file-Kool-Aid J2EE droids decided that, but it's made me like UNIX a lot less than I did before now that they're proliferating. The downhill slide started someone someone thought it was a good idea to put curly brackets into config files... I'll buy that. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with or without OpenSER
On Wed, Nov 19, 2008 at 12:33 PM, Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 19 Nov 2008, Alex Balashov wrote: mail-lists wrote: Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. What do you mean by functionality? Are you looking for low-level or high-level functionality? Also, XML is not a reasonable format for config files. I don't know what sipping-the-property-file-Kool-Aid J2EE droids decided that, but it's made me like UNIX a lot less than I did before now that they're proliferating. The downhill slide started someone someone thought it was a good idea to put curly brackets into config files... Gordon Personal preference and propaganda. If you like XML and are comfortable with it, then why is it not suitable? Thread quickly falls into this or that programming language is the best. Windows vs Linux, Mac vs PC... -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with or without OpenSER
Steve Totaro wrote: On Wed, Nov 19, 2008 at 12:33 PM, Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 19 Nov 2008, Alex Balashov wrote: mail-lists wrote: Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. What do you mean by functionality? Are you looking for low-level or high-level functionality? Also, XML is not a reasonable format for config files. I don't know what sipping-the-property-file-Kool-Aid J2EE droids decided that, but it's made me like UNIX a lot less than I did before now that they're proliferating. The downhill slide started someone someone thought it was a good idea to put curly brackets into config files... Gordon Personal preference and propaganda. If you like XML and are comfortable with it, then why is it not suitable? Not terse, high overhead to parse, difficult to read. There are objective reasons why it is ridiculous apart from personal preference and propaganda. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail - audio problem
Please help... The 1st voicemail message after a reload has audio to the caller. All subsequent calls have no audio to the caller even though the same voicemail application is being called? Asterisk Version 1.4.21.2 Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/voip-1fd034e0, 910|u) in new stack -- SIP/voip-1fd034e0 Playing 'vm-theperson' (language 'en') == Spawn extension (In, 08792200189, 2) exited non-zero on 'SIP/voip-1fd034e0' voicemail.conf [default] ; Define maximum number of messages per folder for a particular context. ;maxmsg=50 910 = 910,Ext910,[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about connecting with Mobile Base Station
On Nov 18, 2008, at 7:30 PM, mark morreny wrote: Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. Best Regards, Mark I've got a blog coming up on this shortly, but I'll comment that there is a BTS-like project that integrates with Asterisk. Check out the burning man installation example where they ran Asterisk and OpenBTS in the desert and enabled all the GSM cell phones in a 1km radius and uplinked them through Asterisk over a long-haul radio link: http://openbts.sourceforge.net/ They need programmers and funding! JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with or without OpenSER
On Wed, 19 Nov 2008, Steve Totaro wrote: On Wed, Nov 19, 2008 at 12:33 PM, Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 19 Nov 2008, Alex Balashov wrote: mail-lists wrote: Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. What do you mean by functionality? Are you looking for low-level or high-level functionality? Also, XML is not a reasonable format for config files. I don't know what sipping-the-property-file-Kool-Aid J2EE droids decided that, but it's made me like UNIX a lot less than I did before now that they're proliferating. The downhill slide started someone someone thought it was a good idea to put curly brackets into config files... Gordon Personal preference and propaganda. If you like XML and are comfortable with it, then why is it not suitable? Thread quickly falls into this or that programming language is the best. Windows vs Linux, Mac vs PC... Indeed, and I forgot the ;-) See: http://www.phespirit.info/montypython/four_yorkshiremen.htm Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Howto grab back call transfered from SIP phone
On our old PBX (Network Alchemy Argent Office) there was a dialcode that grabbed back the last call that went through your extension - very useful when you realised what you'd done. We tend to train our customers to always use attended transfer rather than blind transfer. Seems to solve the problem nicely. If that's not an option, how about writing a transfer macro that'll return the call to the originating extension if the transfer is unanswered within X seconds? Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400 (?) zap hangup
And if that ain't confusing I don't know what would be. I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago and ended up never using it. Passed it along to a friend who is having some problems with it. (He isn't on this list.) We've both tried searches using Google but haven't been able to find anything that helps. So this is more a question of what-the-heck-should-we-be-searching-for. :-) The TDM400 works taking inbound calls and gives a dial tone when the phone is picked up but as soon as a key is pressed the line (Asterisk says) hangs up. Asterisk is configured based on a working system but that one only has one module inbound (FXO?) The outbound settings are based on docs from voip-info.org. Does this ring a bell for anyone? No pun intended. Unfortunately the system is 35 miles away and I haven't got the logs handy so I can't provide more right now. Just hoping for a clue on search terms. TIA, Rod -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrading Asterisk and FreePBX from 1.2 to 1.4
I have a new customer that wants to upgrade their Asterisk installation from 1.2.27 to 1.4.22. They use FreePBX for administration. Since there are many syntax and command changes from those versions of Asterisk, is there an easy way to convert the FreePBX configuration so it will work with the newer Asterisk? -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 (?) zap hangup
On Wed, Nov 19, 2008 at 11:07:57AM -0800, Roderick A. Anderson wrote: And if that ain't confusing I don't know what would be. I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago and ended up never using it. Passed it along to a friend who is having some problems with it. (He isn't on this list.) We've both tried searches using Google but haven't been able to find anything that helps. So this is more a question of what-the-heck-should-we-be-searching-for. :-) The TDM400 works taking inbound calls and gives a dial tone when the phone is picked up but as soon as a key is pressed the line (Asterisk says) hangs up. Dialplan issue? What do you have in the s context there? Asterisk is configured based on a working system but that one only has one module inbound (FXO?) The outbound settings are based on docs from voip-info.org. Does this ring a bell for anyone? No pun intended. Unfortunately the system is 35 miles away and I haven't got the logs handy so I can't provide more right now. Just hoping for a clue on search terms. 35 miles away is no excuse for not having logs :-) Specifically, the debug-level logs should be able to give you the exact reason for the hangup (among many lines of useless information). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to handle include files?
On Wed, Nov 19, 2008 at 01:14:55PM -0600, Doug wrote: Hi folks, I am building a new box. Want it to look pretty much like an older Asterisk 1.2, Debian box that is in production. The new box will used as a test box before we implement changes to the production box. New box: # cat /etc/issue; uname -a Debian GNU/Linux 4.0 \n \l Linux ServerName 2.6.18-6-686 #1 SMP Mon Oct 13 16:13:09 UTC 2008 i686 GNU/Linux I've got Asterisk compiled and running: # asterisk -rv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.2.30.2, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. The problem lies when I try to compile rxfax and txfax. The compiler jumps out of the /usr/src/asterisk/asterisk/asterisk-1.2.30.2/apps/ directory: /bin/sh: curl-config: command not found cc -fPIC -c -o app_dial.o app_dial.c app_dial.c:37:22: error: asterisk.h: No such file or directory app_dial.c:39: error: expected declaration specifiers or â...â before string constant asterisk.h is located: # find / -name asterisk.h /usr/src/asterisk/asterisk/asterisk-1.2.30.2/include/asterisk.h I am finding that other Asterisk-related include files are located: /usr/include/asterisk/ but, they have a recent time stamp. I prefer a time stamp that indicated the last real modification date. [Use package management rather than gueswork?] Researching on the Web, some people suggest copying all the include files to: /usr/include/asterisk/ This is indeed normally installed by 'make install' of Asterisk. Others suggest making a symbolic link that translates: /usr/include/asterisk/ to: /usr/src/asterisk/asterisk/asterisk-1.2.30.2/include/ Why do you actually want to keep the build directory around? (Note that Asterisk modules don't link at build time with and Asterisk component (e.g.: library), and hence the sterisk-devel only includes only the header files) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best way to handle include files?
Hi folks, I am building a new box. Want it to look pretty much like an older Asterisk 1.2, Debian box that is in production. The new box will used as a test box before we implement changes to the production box. New box: # cat /etc/issue; uname -a Debian GNU/Linux 4.0 \n \l Linux ServerName 2.6.18-6-686 #1 SMP Mon Oct 13 16:13:09 UTC 2008 i686 GNU/Linux I've got Asterisk compiled and running: # asterisk -rv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.2.30.2, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. The problem lies when I try to compile rxfax and txfax. The compiler jumps out of the /usr/src/asterisk/asterisk/asterisk-1.2.30.2/apps/ directory: /bin/sh: curl-config: command not found cc -fPIC -c -o app_dial.o app_dial.c app_dial.c:37:22: error: asterisk.h: No such file or directory app_dial.c:39: error: expected declaration specifiers or â...â before string constant asterisk.h is located: # find / -name asterisk.h /usr/src/asterisk/asterisk/asterisk-1.2.30.2/include/asterisk.h I am finding that other Asterisk-related include files are located: /usr/include/asterisk/ but, they have a recent time stamp. I prefer a time stamp that indicated the last real modification date. Researching on the Web, some people suggest copying all the include files to: /usr/include/asterisk/ Others suggest making a symbolic link that translates: /usr/include/asterisk/ to: /usr/src/asterisk/asterisk/asterisk-1.2.30.2/include/ Does anyone have some suggestions on the best way to handle include files so that rxfax and txfax, Asterisk and its related components can be compiled? Also, what if the box was upgraded in the future to 1.2.32? What would be the best overall solution? Thanks for your help! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Role of asterisk
Hello Mr. Smith, Thank you very much for you time and explanations. I just started to take this VoIP business serious and as i mentioned in my previous email, I took a SIP book. If you know any kind of books that are suitable for a beginner please let me know. I started to read SIP books because i have an ackward problem with an asterisk box and i really have to know the protocol to understand what happens there. thanks once again and a great day, v On Wed, Nov 19, 2008 at 4:55 PM, Jared Smith [EMAIL PROTECTED] wrote: On Wed, 2008-11-19 at 12:33 +0200, Valentin Bud wrote: i think that asterisk is a stateful proxy server as well as registration server. To answer the second portion of your question (which I forgot to do in my earlier email)... yes, Asterisk can be a registration server as well. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] puzzle
Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC performance
What you might want to do it try OSLEC Gordon, Digium hasn't responded to me with my key to install HPEC after waiting several days, and tonight I need to get the card installed as my number port takes place and that location will be w/o phones. I am using Asterisk 1.6 and DAHDI and from what I see its not trivial to build OSLEC support into DAHDI, has that changed? Thanks for the reco! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 (?) zap hangup
Tzafrir Cohen wrote: On Wed, Nov 19, 2008 at 11:07:57AM -0800, Roderick A. Anderson wrote: And if that ain't confusing I don't know what would be. I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago and ended up never using it. Passed it along to a friend who is having some problems with it. (He isn't on this list.) We've both tried searches using Google but haven't been able to find anything that helps. So this is more a question of what-the-heck-should-we-be-searching-for. :-) The TDM400 works taking inbound calls and gives a dial tone when the phone is picked up but as soon as a key is pressed the line (Asterisk says) hangs up. Dialplan issue? What do you have in the s context there? I'll have to look. It is a bit of a mess. Bits and pieces and the system is also used for testing a multi-tenant setup. The Asterisk configuration started on BSD using * 1.2 but because of support (or lack thereof) and some needed/desired features was moved to CentOS 5, and * 1.4. I based my system on the same setup but have taken time to strip the cruft out and clean up the poorly formatted files that also lacked any comments/documentation. Talk about a learning experience! Asterisk is configured based on a working system but that one only has one module inbound (FXO?) The outbound settings are based on docs from voip-info.org. Does this ring a bell for anyone? No pun intended. Unfortunately the system is 35 miles away and I haven't got the logs handy so I can't provide more right now. Just hoping for a clue on search terms. 35 miles away is no excuse for not having logs :-) Logs I can get just didn't have them handy when I first tried to send the message (a whole other story of too many email accounts and mailing lists.). Plus I'd like to watch as it was happening. Specifically, the debug-level logs should be able to give you the exact reason for the hangup (among many lines of useless information). Yup. That's why I want to watch it happening. Just hoping (wishing?) for serendipity! Looks like it will be the hard way. :-( Thanks, Rod -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 (?) zap hangup
On Wed, 2008-11-19 at 11:07 -0800, Roderick A. Anderson wrote: The TDM400 works taking inbound calls and gives a dial tone when the phone is picked up but as soon as a key is pressed the line (Asterisk says) hangs up. Asterisk is configured based on a working system but that one only has one module inbound (FXO?) The outbound settings are based on docs from voip-info.org. It sounds to me like you've got the two channels pointed at a context that doesn't exist, or there aren't extensions in that context that match the number being dialed. Look in zapata.conf (or chan_dahdi.conf if you're using DAHDI) and look for the context= line immediately above the channel= line for each channel. That tells Asterisk where in the dialplan to look when calls come into Asterisk from those channels. Next, look at extensions.conf and see if those contexts exist. For example, if your FXS port (the port connected to an analog phone) is channel 1, and you're trying to dial extension 500 from the analog phone, make sure the context contains an extension 500. You can also check from the Asterisk CLI by typing dialplan show [EMAIL PROTECTED], where 500 is the extension you're trying to dial and context is the name of the context where calls are being sent. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
Try flushing all of your iptables and see if that helps. See if there's anything in your dmesg that might indicate what's up. Jeff LaCoursiere wrote: Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
Yes, the second 'ps' below showed the parent to be '1' (init), which means its real parent died already. Any attempt to flush the iptables hangs :( j On Wed, 19 Nov 2008, Danny Nicholas wrote: Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
Your could try this History|grep modprobe Rmmod XXX where xxx is the parameter from the history|grep modprobe. This of course assumes that the command is in your last 1000 commands. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle Yes, the second 'ps' below showed the parent to be '1' (init), which means its real parent died already. Any attempt to flush the iptables hangs :( j On Wed, 19 Nov 2008, Danny Nicholas wrote: Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with or without OpenSER
Alex Balashov wrote: mail-lists wrote: Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. What do you mean by functionality? Are you looking for low-level or high-level functionality? I guess I mean FS has more high level functionality like conference rooms and voicemail modules which might allow us to offload some of this from *. OpenSER has some of this as well I think. I'm not sure FS lets you interact directly with the SIP stack like OpenSER/SER does though (I might be completely wrong about this) Also, XML is not a reasonable format for config files. I don't know what sipping-the-property-file-Kool-Aid J2EE droids decided that, but it's made me like UNIX a lot less than I did before now that they're proliferating. I like XML. I know there's a lot of extra grammar but it keeps things straight in my head. I don't have a a great deal of experience with various config options but in the past I've much preferred XML based phone configs to others. To each their own I suppose. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Howto grab back call transfered from SIP phone
On Wed, Nov 19, 2008 at 2:01 PM, Chris Bagnall [EMAIL PROTECTED] wrote: On our old PBX (Network Alchemy Argent Office) there was a dialcode that grabbed back the last call that went through your extension - very useful when you realised what you'd done. We tend to train our customers to always use attended transfer rather than blind transfer. Seems to solve the problem nicely. If that's not an option, how about writing a transfer macro that'll return the call to the originating extension if the transfer is unanswered within X seconds? Regards, Chris Look into app_bridge. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with or without OpenSER
mail-lists wrote: Alex Balashov wrote: mail-lists wrote: Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. What do you mean by functionality? Are you looking for low-level or high-level functionality? I guess I mean FS has more high level functionality like conference rooms and voicemail modules which might allow us to offload some of this from *. OpenSER has some of this as well I think. I'm not sure FS lets you interact directly with the SIP stack like OpenSER/SER does though (I might be completely wrong about this) No, OpenSER doesn't have any of this functionality. OpenSER is not a user agent of any kind and has no features relevant to the user experience as such. It only serves the low-level part of this formula. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Howto grab back call transfered from SIP phone
On Wed, Nov 19, 2008 at 3:28 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Wed, Nov 19, 2008 at 2:01 PM, Chris Bagnall [EMAIL PROTECTED] wrote: On our old PBX (Network Alchemy Argent Office) there was a dialcode that grabbed back the last call that went through your extension - very useful when you realised what you'd done. We tend to train our customers to always use attended transfer rather than blind transfer. Seems to solve the problem nicely. If that's not an option, how about writing a transfer macro that'll return the call to the originating extension if the transfer is unanswered within X seconds? Regards, Chris Look into app_bridge. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Extra, Extra, Read all About it: http://bugs.digium.com/view.php?id=5841 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 (?) zap hangup
Jared Smith wrote: On Wed, 2008-11-19 at 11:07 -0800, Roderick A. Anderson wrote: The TDM400 works taking inbound calls and gives a dial tone when the phone is picked up but as soon as a key is pressed the line (Asterisk says) hangs up. Asterisk is configured based on a working system but that one only has one module inbound (FXO?) The outbound settings are based on docs from voip-info.org. It sounds to me like you've got the two channels pointed at a context that doesn't exist, or there aren't extensions in that context that match the number being dialed. Look in zapata.conf (or chan_dahdi.conf if you're using DAHDI) and look for the context= line immediately above the channel= line for each channel. That tells Asterisk where in the dialplan to look when calls come into Asterisk from those channels. Next, look at extensions.conf and see if those contexts exist. For example, if your FXS port (the port connected to an analog phone) is channel 1, and you're trying to dial extension 500 from the analog phone, make sure the context contains an extension 500. You can also check from the Asterisk CLI by typing dialplan show [EMAIL PROTECTED], where 500 is the extension you're trying to dial and context is the name of the context where calls are being sent. Thanks for the pointers Jared. They will help. Rod -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC performance
On Wed, Nov 19, 2008 at 12:57:53PM -0700, Joseph L. Casale wrote: What you might want to do it try OSLEC Gordon, Digium hasn't responded to me with my key to install HPEC after waiting several days, and tonight I need to get the card installed as my number port takes place and that location will be w/o phones. I am using Asterisk 1.6 and DAHDI and from what I see its not trivial to build OSLEC support into DAHDI, has that changed? Not trivial but not as voodoo as before: http://docs.tzafrir.org.il/dahdi-linux/#_oslec -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 call files Disposition=NO ANSWER
On Wed, 2008-11-19 at 10:00 +1000, David Klaverstyn wrote: Hi Guys, Since moving to Asterisk 1.6, whenever I am using call files the call is always logged with a disposition of NO ANSWER even though the call is connected and answered. The duration displays the correct time. Can anyone explain as to why when using call files the disposition is not correct? It just so happens that I've JUST generated a patch for a fairly similar problem (see http://bugs.digium.com/view.php?id=12694 ) The main difference is that, they are seeing the problem where the CDRs are OK with BUSY, and ANSWER; they were getting FAIL instead of NO ANSWER. You are seeing somewhat the opposite... Report here or in the bug tracker, the contents of your call file, the corresponding referenced parts of your dialplan, and maybe, since this code is freshly in my brain, I might be able to debug it quickly; or let you know the error of your ways. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail - audio problem
On Wed, Nov 19, 2008 at 1:07 PM, Shaun Wingrin [EMAIL PROTECTED] wrote: Please help... The 1st voicemail message after a reload has audio to the caller. All subsequent calls have no audio to the caller even though the same voicemail application is being called? make sure you have ztdummy loaded. Not sure why, but I ran into a problem similar to what you're describing with 1.4.21.2 (even though I have a wcte11xp module loaded) and modprobing ztdummy fixed it. Asterisk Version 1.4.21.2 [snip] HTH, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
On Wed, Nov 19, 2008 at 07:57:33PM +, Jeff LaCoursiere wrote: Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state modprobe -r is basically rmmod . rmmod and insmod and nowdays mostly wrappers to kernel code. So while an strace of that process might give some more information about it, I believe that the kernel-level backtrace would be more interesting. For that, try either the 'p' or 't' sysrq commands. 'p' gives a stack trace of the current process. 't': of all the processes. You can give a sysrq command either through the console (on x86: alt-sysrq-key) or: echo key /proc/sysrq-trigger The output goes to the kernel logs, e.g. in dmesg . ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 This will probably apply when the process will leave whatever busy context it is in. ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? BTW: what kernel? What ditsribution? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
A good idea! The modprobe command is actually in the ps below - it is part of the /etc/init.d/iptables script, and apparently was trying to remove the ipt_state module. The result, however: [EMAIL PROTECTED] init.d]# rmmod ipt_state ERROR: Module ipt_state does not exist in /proc/modules (sigh). In fact /proc/modules is empty. [EMAIL PROTECTED] init.d]# ls -ltr /proc/modules -r--r--r-- 1 root root 0 Nov 19 14:46 /proc/modules j On Wed, 19 Nov 2008, Danny Nicholas wrote: Your could try this History|grep modprobe Rmmod XXX where xxx is the parameter from the history|grep modprobe. This of course assumes that the command is in your last 1000 commands. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle Yes, the second 'ps' below showed the parent to be '1' (init), which means its real parent died already. Any attempt to flush the iptables hangs :( j On Wed, 19 Nov 2008, Danny Nicholas wrote: Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to handle include files?
Thanks, Tzafrir, for your reply! At 13:25 11/19/2008, Tzafrir Cohen wrote: On Wed, Nov 19, 2008 at 01:14:55PM -0600, Doug wrote: Hi folks, I am building a new box. Want it to look pretty much like an older Asterisk 1.2, Debian box that is in production. The new box will used as a test box before we implement changes to the production box. New box: # cat /etc/issue; uname -a Debian GNU/Linux 4.0 \n \l Linux ServerName 2.6.18-6-686 #1 SMP Mon Oct 13 16:13:09 UTC 2008 i686 GNU/Linux I've got Asterisk compiled and running: # asterisk -rv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.2.30.2, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. The problem lies when I try to compile rxfax and txfax. The compiler jumps out of the /usr/src/asterisk/asterisk/asterisk-1.2.30.2/apps/ directory: /bin/sh: curl-config: command not found cc -fPIC -c -o app_dial.o app_dial.c app_dial.c:37:22: error: asterisk.h: No such file or directory app_dial.c:39: error: expected declaration specifiers or â...â before string constant asterisk.h is located: # find / -name asterisk.h /usr/src/asterisk/asterisk/asterisk-1.2.30.2/include/asterisk.h I am finding that other Asterisk-related include files are located: /usr/include/asterisk/ but, they have a recent time stamp. I prefer a time stamp that indicated the last real modification date. [Use package management rather than gueswork?] Do you mean: # apt-get update Get:1 http://ftp.uwsg.indiana.edu etch Release.gpg [386B] Hit http://ftp.uwsg.indiana.edu etch Release Get:2 http://security.debian.org etch/updates Release.gpg [189B] Get:3 http://security.debian.org etch/updates Release [37.6kB] Ign http://ftp.uwsg.indiana.edu etch/main Packages/DiffIndex Ign http://ftp.uwsg.indiana.edu etch/non-free Packages/DiffIndex Ign http://ftp.uwsg.indiana.edu etch/main Sources/DiffIndex Ign http://ftp.uwsg.indiana.edu etch/non-free Sources/DiffIndex Hit http://ftp.uwsg.indiana.edu etch/main Packages Hit http://ftp.uwsg.indiana.edu etch/non-free Packages Hit http://ftp.uwsg.indiana.edu etch/main Sources Hit http://ftp.uwsg.indiana.edu etch/non-free Sources Ign http://security.debian.org etch/updates/main Packages/DiffIndex Ign http://security.debian.org etch/updates/contrib Packages/DiffIndex Ign http://security.debian.org etch/updates/non-free Packages/DiffIndex Ign http://security.debian.org etch/updates/main Sources/DiffIndex Ign http://security.debian.org etch/updates/contrib Sources/DiffIndex Ign http://security.debian.org etch/updates/non-free Sources/DiffIndex Get:4 http://security.debian.org etch/updates/main Packages [291kB] Hit http://security.debian.org etch/updates/contrib Packages Hit http://security.debian.org etch/updates/non-free Packages Get:5 http://security.debian.org etch/updates/main Sources [45.9kB] Hit http://security.debian.org etch/updates/contrib Sources Hit http://security.debian.org etch/updates/non-free Sources Fetched 375kB in 1s (309kB/s) Reading package lists... Done http://www.google.com/search?q=asterisk+%22package+management%22 Researching on the Web, some people suggest copying all the include files to: /usr/include/asterisk/ This is indeed normally installed by 'make install' of Asterisk. Right. Why do the .h files have the install date instead of the last modified date? Others suggest making a symbolic link that translates: /usr/include/asterisk/ to: /usr/src/asterisk/asterisk/asterisk-1.2.30.2/include/ Why do you actually want to keep the build directory around? Well, we've got plenty of disk space, and it gives a historical record of upgrades. Why wouldn't we want to keep them around? This seems to imply that the include files should be copied into: /usr/include/asterisk/ Is this correct? (Note that Asterisk modules don't link at build time with and Asterisk component (e.g.: library), and hence the sterisk-devel only includes only the header files) I am confused. Are you saying that when compiling or recompiling the Asterisk modules don't link? I am not exactly sure what you are saying. Again, what is the best way to handle include files so that rxfax and txfax will compile, and will allow for future upgrades of Asterisk versions? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___
[asterisk-users] VoiceMail - audio problem
Dear David, Thanks for the reply. I have lsmod | grep ztdummy ztdummy38856 0 zaptel231496 3 ztdummy but still the issue persists?! Any ideas really apreciated. - Original Message - From: David A. Bandel [EMAIL PROTECTED] To: Shaun Wingrin [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 19, 2008 10:36 PM Subject: Re: [asterisk-users] VoiceMail - audio problem On Wed, Nov 19, 2008 at 1:07 PM, Shaun Wingrin [EMAIL PROTECTED] wrote: Please help... The 1st voicemail message after a reload has audio to the caller. All subsequent calls have no audio to the caller even though the same voicemail application is being called? make sure you have ztdummy loaded. Not sure why, but I ran into a problem similar to what you're describing with 1.4.21.2 (even though I have a wcte11xp module loaded) and modprobing ztdummy fixed it. Asterisk Version 1.4.21.2 [snip] HTH, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
On Wed, 19 Nov 2008, Tzafrir Cohen wrote: On Wed, Nov 19, 2008 at 07:57:33PM +, Jeff LaCoursiere wrote: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state modprobe -r is basically rmmod . rmmod and insmod and nowdays mostly wrappers to kernel code. So while an strace of that process might give some more information about it, I believe that the kernel-level backtrace would be more interesting. For that, try either the 'p' or 't' sysrq commands. 'p' gives a stack trace of the current process. 't': of all the processes. You can give a sysrq command either through the console (on x86: alt-sysrq-key) or: echo key /proc/sysrq-trigger No access to the console, sadly, so I tried the trigger method: [EMAIL PROTECTED] init.d]# echo p /proc/sysrq-trigger which resulted in a single line in /var/log/messages: Nov 19 14:51:10 ast kernel: SysRq : Show Regs I waited a few minutes, then tried the 't': [EMAIL PROTECTED] init.d]# echo t /proc/sysrq-trigger which seemed to hang, so I killed it about thirty seconds later, and now my /var/log/messages has 20,000 extra lines :):) I grepped for the PID and found this: Nov 19 14:52:40 ast kernel: modprobe R running 2988 17744 1 31140 28078 (NOTLB) The next line started with 'sshd', so I guess there was no trace with this? BTW: what kernel? What ditsribution? Keep in mind it has been running almost 1000 days ;) [EMAIL PROTECTED] init.d]# uname -a Linux ast.jbtelenet.com 2.6.9-22.0.2.ELsmp #1 SMP Thu Jan 5 17:13:01 EST 2006 i686 i686 i386 GNU/Linux I believe it is Redhat 9. Its a colo... Thanks for the interesting debug pointers! j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
On Wed, Nov 19, 2008 at 09:06:47PM +, Jeff LaCoursiere wrote: I grepped for the PID and found this: Nov 19 14:52:40 ast kernel: modprobe R running 2988 17744 1 31140 28078 (NOTLB) The next line started with 'sshd', so I guess there was no trace with this? Right :-( BTW: what kernel? What ditsribution? Keep in mind it has been running almost 1000 days ;) [EMAIL PROTECTED] init.d]# uname -a Linux ast.jbtelenet.com 2.6.9-22.0.2.ELsmp #1 SMP Thu Jan 5 17:13:01 EST 2006 i686 i686 i386 GNU/Linux I believe it is Redhat 9. Its a colo... RHEL 4.2 (or compatible, e.g. Centos 4.2) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
/proc/modules is a pipe You can see what is in there by type cat /proc/modules|more -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle A good idea! The modprobe command is actually in the ps below - it is part of the /etc/init.d/iptables script, and apparently was trying to remove the ipt_state module. The result, however: [EMAIL PROTECTED] init.d]# rmmod ipt_state ERROR: Module ipt_state does not exist in /proc/modules (sigh). In fact /proc/modules is empty. [EMAIL PROTECTED] init.d]# ls -ltr /proc/modules -r--r--r-- 1 root root 0 Nov 19 14:46 /proc/modules j On Wed, 19 Nov 2008, Danny Nicholas wrote: Your could try this History|grep modprobe Rmmod XXX where xxx is the parameter from the history|grep modprobe. This of course assumes that the command is in your last 1000 commands. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle Yes, the second 'ps' below showed the parent to be '1' (init), which means its real parent died already. Any attempt to flush the iptables hangs :( j On Wed, 19 Nov 2008, Danny Nicholas wrote: Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 call files Disposition=NO ANSWER
On Wed, 2008-11-19 at 13:34 -0700, Steve Murphy wrote: On Wed, 2008-11-19 at 10:00 +1000, David Klaverstyn wrote: Hi Guys, Since moving to Asterisk 1.6, whenever I am using call files the call is always logged with a disposition of NO ANSWER even though the call is connected and answered. The duration displays the correct time. Can anyone explain as to why when using call files the disposition is not correct? It just so happens that I've JUST generated a patch for a fairly similar problem (see http://bugs.digium.com/view.php?id=12694 ) The main difference is that, they are seeing the problem where the CDRs are OK with BUSY, and ANSWER; they were getting FAIL instead of NO ANSWER. You are seeing somewhat the opposite... Report here or in the bug tracker, the contents of your call file, the corresponding referenced parts of your dialplan, and maybe, since this code is freshly in my brain, I might be able to debug it quickly; or let you know the error of your ways. Hmmm. This is http://bugs.digium.com/view.php?id=13665 ! I'm working on it now. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC performance
Not trivial but not as voodoo as before: http://docs.tzafrir.org.il/dahdi-linux/#_oslec Tzafrir, Appreciate this pointer, I am intending on setting this up on a CentOS 5 x86 box. The drastically different stock running kernel compared to the files I need from your doc won't be an issue? Also, in searching the net, I see some issues where people complain DAHDI is not as stable as Zap, is this true or no longer the case? Thank you for all the help! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
Hmm, I am more of a BSD guy I guess. I would expect a pipe to show a 'p' in a long ls. This is interesting though: [EMAIL PROTECTED] init.d]# cat /proc/modules | head ip_conntrack 45573 0 - Unloading 0xf8945000 [EMAIL PROTECTED] init.d]# rmmod -f ip_conntrack ERROR: Removing 'ip_conntrack': Device or resource busy (sigh) I am pretty sure ip_conntrack is part of the iptables stuff... j On Wed, 19 Nov 2008, Danny Nicholas wrote: /proc/modules is a pipe You can see what is in there by type cat /proc/modules|more -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle A good idea! The modprobe command is actually in the ps below - it is part of the /etc/init.d/iptables script, and apparently was trying to remove the ipt_state module. The result, however: [EMAIL PROTECTED] init.d]# rmmod ipt_state ERROR: Module ipt_state does not exist in /proc/modules (sigh). In fact /proc/modules is empty. [EMAIL PROTECTED] init.d]# ls -ltr /proc/modules -r--r--r-- 1 root root 0 Nov 19 14:46 /proc/modules j On Wed, 19 Nov 2008, Danny Nicholas wrote: Your could try this History|grep modprobe Rmmod XXX where xxx is the parameter from the history|grep modprobe. This of course assumes that the command is in your last 1000 commands. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle Yes, the second 'ps' below showed the parent to be '1' (init), which means its real parent died already. Any attempt to flush the iptables hangs :( j On Wed, 19 Nov 2008, Danny Nicholas wrote: Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC performance
Not trivial but not as voodoo as before: http://docs.tzafrir.org.il/dahdi-linux/#_oslec Tzafrir, I pulled down linux-2.6.28-rc5.tar.bz2 and followed the doc, now when compiling I get the following: WARNING: oslec_create [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined! WARNING: oslec_free [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined! WARNING: oslec_update [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined! Any ideas? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi_test drops after restarting Sangoma driver
Hi, Does anybody have an idea as to why dahdi_test results drop to unacceptable levels after doing a wanrouter stop/start using a Sangoma card? See below that it drops from 99.99% to 98.55%: [EMAIL PROTECTED] dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.999512% 99.992874% --- Results after 2 passes --- Best: 100.000 -- Worst: 99.993 -- Average: 99.996193, Difference: 99.996683 [EMAIL PROTECTED] bin]# service wanrouter stop Shutting down asterisk:[ OK ] Stopping Asterisk... Shutting down wanpipe4 interface: w4g1 Shutting down wanpipe3 interface: w3g1 Shutting down wanpipe2 interface: w2g1 Shutting down wanpipe1 interface: w1g1 Shutting down device: wanpipe4 Shutting down device: wanpipe3 Shutting down device: wanpipe2 Shutting down device: wanpipe1 No devices running, Unloading Modules [EMAIL PROTECTED] bin]# service wanrouter start Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 -- Loading ec image OCT6116-256S.ima... Starting up device: wanpipe2 Starting up device: wanpipe3 Starting up device: wanpipe4 Configuring interfaces: w1g1 done. Configuring interfaces: w2g1 done. Configuring interfaces: w3g1 done. Configuring interfaces: w4g1 done. [EMAIL PROTECTED] dahdi_test Opened pseudo dahdi interface, measuring accuracy... 98.553322% 98.650970% --- Results after 2 passes --- Best: 98.651 -- Worst: 98.553 -- Average: 98.602146, Difference: 98.602149 It has to do something with starting the driver from the command line. If I start from boot or via a cron job, the value goes up to 99.99%. I'm stumped. Andres ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
YUM update? service iptables stop service iptables start? On Wed, Nov 19, 2008 at 4:46 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: Hmm, I am more of a BSD guy I guess. I would expect a pipe to show a 'p' in a long ls. This is interesting though: [EMAIL PROTECTED] init.d]# cat /proc/modules | head ip_conntrack 45573 0 - Unloading 0xf8945000 [EMAIL PROTECTED] init.d]# rmmod -f ip_conntrack ERROR: Removing 'ip_conntrack': Device or resource busy (sigh) I am pretty sure ip_conntrack is part of the iptables stuff... j On Wed, 19 Nov 2008, Danny Nicholas wrote: /proc/modules is a pipe You can see what is in there by type cat /proc/modules|more -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle A good idea! The modprobe command is actually in the ps below - it is part of the /etc/init.d/iptables script, and apparently was trying to remove the ipt_state module. The result, however: [EMAIL PROTECTED] init.d]# rmmod ipt_state ERROR: Module ipt_state does not exist in /proc/modules (sigh). In fact /proc/modules is empty. [EMAIL PROTECTED] init.d]# ls -ltr /proc/modules -r--r--r-- 1 root root 0 Nov 19 14:46 /proc/modules j On Wed, 19 Nov 2008, Danny Nicholas wrote: Your could try this History|grep modprobe Rmmod XXX where xxx is the parameter from the history|grep modprobe. This of course assumes that the command is in your last 1000 commands. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle Yes, the second 'ps' below showed the parent to be '1' (init), which means its real parent died already. Any attempt to flush the iptables hangs :( j On Wed, 19 Nov 2008, Danny Nicholas wrote: Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To
Re: [asterisk-users] puzzle
Its not Centos - there is no 'yum'. service iptables stop is what produced the hanging process in the first place - I think my big problem here is that a kernel module is broken, and there is no way to stop it, and there seems to be no way to unload it (in fact it is hung trying to do just that). Thanks for the suggestions, though! j On Wed, 19 Nov 2008, Steve Totaro wrote: YUM update? service iptables stop service iptables start? On Wed, Nov 19, 2008 at 4:46 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: Hmm, I am more of a BSD guy I guess. I would expect a pipe to show a 'p' in a long ls. This is interesting though: [EMAIL PROTECTED] init.d]# cat /proc/modules | head ip_conntrack 45573 0 - Unloading 0xf8945000 [EMAIL PROTECTED] init.d]# rmmod -f ip_conntrack ERROR: Removing 'ip_conntrack': Device or resource busy (sigh) I am pretty sure ip_conntrack is part of the iptables stuff... j On Wed, 19 Nov 2008, Danny Nicholas wrote: /proc/modules is a pipe You can see what is in there by type cat /proc/modules|more -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle A good idea! The modprobe command is actually in the ps below - it is part of the /etc/init.d/iptables script, and apparently was trying to remove the ipt_state module. The result, however: [EMAIL PROTECTED] init.d]# rmmod ipt_state ERROR: Module ipt_state does not exist in /proc/modules (sigh). In fact /proc/modules is empty. [EMAIL PROTECTED] init.d]# ls -ltr /proc/modules -r--r--r-- 1 root root 0 Nov 19 14:46 /proc/modules j On Wed, 19 Nov 2008, Danny Nicholas wrote: Your could try this History|grep modprobe Rmmod XXX where xxx is the parameter from the history|grep modprobe. This of course assumes that the command is in your last 1000 commands. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle Yes, the second 'ps' below showed the parent to be '1' (init), which means its real parent died already. Any attempt to flush the iptables hangs :( j On Wed, 19 Nov 2008, Danny Nicholas wrote: Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas other than rebooting? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
Well then use whatever package manager you have. Apt-get I assume. Maybe that might help. What do you get with #ls -ltr /etc/init.d? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Wed, Nov 19, 2008 at 7:19 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: Its not Centos - there is no 'yum'. service iptables stop is what produced the hanging process in the first place - I think my big problem here is that a kernel module is broken, and there is no way to stop it, and there seems to be no way to unload it (in fact it is hung trying to do just that). Thanks for the suggestions, though! j On Wed, 19 Nov 2008, Steve Totaro wrote: YUM update? service iptables stop service iptables start? On Wed, Nov 19, 2008 at 4:46 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: Hmm, I am more of a BSD guy I guess. I would expect a pipe to show a 'p' in a long ls. This is interesting though: [EMAIL PROTECTED] init.d]# cat /proc/modules | head ip_conntrack 45573 0 - Unloading 0xf8945000 [EMAIL PROTECTED] init.d]# rmmod -f ip_conntrack ERROR: Removing 'ip_conntrack': Device or resource busy (sigh) I am pretty sure ip_conntrack is part of the iptables stuff... j On Wed, 19 Nov 2008, Danny Nicholas wrote: /proc/modules is a pipe You can see what is in there by type cat /proc/modules|more -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle A good idea! The modprobe command is actually in the ps below - it is part of the /etc/init.d/iptables script, and apparently was trying to remove the ipt_state module. The result, however: [EMAIL PROTECTED] init.d]# rmmod ipt_state ERROR: Module ipt_state does not exist in /proc/modules (sigh). In fact /proc/modules is empty. [EMAIL PROTECTED] init.d]# ls -ltr /proc/modules -r--r--r-- 1 root root 0 Nov 19 14:46 /proc/modules j On Wed, 19 Nov 2008, Danny Nicholas wrote: Your could try this History|grep modprobe Rmmod XXX where xxx is the parameter from the history|grep modprobe. This of course assumes that the command is in your last 1000 commands. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle Yes, the second 'ps' below showed the parent to be '1' (init), which means its real parent died already. Any attempt to flush the iptables hangs :( j On Wed, 19 Nov 2008, Danny Nicholas wrote: Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23224:41 modprobe -r ipt_state ast% You may also notice that I tried renice to bump it all the way to +19 and still it consumes 100% of the CPU. The result for asterisk is that I hear bits of robot noise during conversations, which is annoying as hell but not neccessarily show stopping. But for another 19 days?? Argg! I assume that because it is 'modprobe' it has tickled some kernel bug that is merrily spinning away and won't respond to interrupts. I even tried to stop it with gdb and strace, both of which also hung and had to be killed with -9. It seems to be related to me screwing with the iptables a few weeks ago. Any ideas
Re: [asterisk-users] Meetme talker optimization always on even when no o option present.
Bill wrote: After loading 1.6.0.1, I notice that I always have the VOX effect on Meetme conferences whether I have the o option set in the dial plan or not. Is anyone else seeing this? Can you describe the effect? I am seeing odd behavior when I have PSTN calls in a conference, oddly most noticeable if the calling party is on a blackberry, but it also impacts other cell phones and land lines. Although I'm now running 1.6.0.1, I'm also seeing this on a system still running 1.6.0beta9. My calls route through a Cisco voice gateway and one Hint is that Asterisk tells me to turn off Comfort Noise for that peer (not possible as far as I can tell) All callers are G711, with very low latency and QOS between the gateway and Asterisk. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
Hi Steve, [EMAIL PROTECTED] ~]# ls -ltr /etc/init.d lrwxrwxrwx 1 root root 11 Nov 29 2007 /etc/init.d - rc.d/init.d [EMAIL PROTECTED] ~]# Although I agree that updating the kernel et all would be a good idea, the whole point is to keep the machine running for 19 more days without the rogue process interfering with my voice quality. If I cannot unload the module or otherwise interrupt the process which is currently spinning in kernel space, no upgrade will be possible. I am quite sure that rebooting will fix this problem, but the puzzle was to fix it without doing so... Cheers, j On Wed, 19 Nov 2008, Steve Totaro wrote: Well then use whatever package manager you have. Apt-get I assume. Maybe that might help. What do you get with #ls -ltr /etc/init.d? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Wed, Nov 19, 2008 at 7:19 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: Its not Centos - there is no 'yum'. service iptables stop is what produced the hanging process in the first place - I think my big problem here is that a kernel module is broken, and there is no way to stop it, and there seems to be no way to unload it (in fact it is hung trying to do just that). Thanks for the suggestions, though! j On Wed, 19 Nov 2008, Steve Totaro wrote: YUM update? service iptables stop service iptables start? On Wed, Nov 19, 2008 at 4:46 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: Hmm, I am more of a BSD guy I guess. I would expect a pipe to show a 'p' in a long ls. This is interesting though: [EMAIL PROTECTED] init.d]# cat /proc/modules | head ip_conntrack 45573 0 - Unloading 0xf8945000 [EMAIL PROTECTED] init.d]# rmmod -f ip_conntrack ERROR: Removing 'ip_conntrack': Device or resource busy (sigh) I am pretty sure ip_conntrack is part of the iptables stuff... j On Wed, 19 Nov 2008, Danny Nicholas wrote: /proc/modules is a pipe You can see what is in there by type cat /proc/modules|more -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle A good idea! The modprobe command is actually in the ps below - it is part of the /etc/init.d/iptables script, and apparently was trying to remove the ipt_state module. The result, however: [EMAIL PROTECTED] init.d]# rmmod ipt_state ERROR: Module ipt_state does not exist in /proc/modules (sigh). In fact /proc/modules is empty. [EMAIL PROTECTED] init.d]# ls -ltr /proc/modules -r--r--r-- 1 root root 0 Nov 19 14:46 /proc/modules j On Wed, 19 Nov 2008, Danny Nicholas wrote: Your could try this History|grep modprobe Rmmod XXX where xxx is the parameter from the history|grep modprobe. This of course assumes that the command is in your last 1000 commands. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle Yes, the second 'ps' below showed the parent to be '1' (init), which means its real parent died already. Any attempt to flush the iptables hangs :( j On Wed, 19 Nov 2008, Danny Nicholas wrote: Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill 17744 ast% sudo kill 17744 ast% sudo kill -9 17744 ast% sudo kill -9 17744 ast% !ps ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?
Re: [asterisk-users] puzzle
No. You can't restart the iptables scripts of any distro and expect them to unstick a conntrack module, even if they explicitly reload those modules from the script (as the user himself tried to do and failed) rather than simply installing iptables rules and expecting them to be loaded on demand. Steve Totaro wrote: Well then use whatever package manager you have. Apt-get I assume. Maybe that might help. What do you get with #ls -ltr /etc/init.d? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] puzzle
I was not implying that you upgrade anything but iptables. What is the output of ls /etc/init.d/ On Wed, Nov 19, 2008 at 8:02 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: Hi Steve, [EMAIL PROTECTED] ~]# ls -ltr /etc/init.d lrwxrwxrwx 1 root root 11 Nov 29 2007 /etc/init.d - rc.d/init.d [EMAIL PROTECTED] ~]# Although I agree that updating the kernel et all would be a good idea, the whole point is to keep the machine running for 19 more days without the rogue process interfering with my voice quality. If I cannot unload the module or otherwise interrupt the process which is currently spinning in kernel space, no upgrade will be possible. I am quite sure that rebooting will fix this problem, but the puzzle was to fix it without doing so... Cheers, j On Wed, 19 Nov 2008, Steve Totaro wrote: Well then use whatever package manager you have. Apt-get I assume. Maybe that might help. What do you get with #ls -ltr /etc/init.d? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Wed, Nov 19, 2008 at 7:19 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: Its not Centos - there is no 'yum'. service iptables stop is what produced the hanging process in the first place - I think my big problem here is that a kernel module is broken, and there is no way to stop it, and there seems to be no way to unload it (in fact it is hung trying to do just that). Thanks for the suggestions, though! j On Wed, 19 Nov 2008, Steve Totaro wrote: YUM update? service iptables stop service iptables start? On Wed, Nov 19, 2008 at 4:46 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: Hmm, I am more of a BSD guy I guess. I would expect a pipe to show a 'p' in a long ls. This is interesting though: [EMAIL PROTECTED] init.d]# cat /proc/modules | head ip_conntrack 45573 0 - Unloading 0xf8945000 [EMAIL PROTECTED] init.d]# rmmod -f ip_conntrack ERROR: Removing 'ip_conntrack': Device or resource busy (sigh) I am pretty sure ip_conntrack is part of the iptables stuff... j On Wed, 19 Nov 2008, Danny Nicholas wrote: /proc/modules is a pipe You can see what is in there by type cat /proc/modules|more -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle A good idea! The modprobe command is actually in the ps below - it is part of the /etc/init.d/iptables script, and apparently was trying to remove the ipt_state module. The result, however: [EMAIL PROTECTED] init.d]# rmmod ipt_state ERROR: Module ipt_state does not exist in /proc/modules (sigh). In fact /proc/modules is empty. [EMAIL PROTECTED] init.d]# ls -ltr /proc/modules -r--r--r-- 1 root root 0 Nov 19 14:46 /proc/modules j On Wed, 19 Nov 2008, Danny Nicholas wrote: Your could try this History|grep modprobe Rmmod XXX where xxx is the parameter from the history|grep modprobe. This of course assumes that the command is in your last 1000 commands. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] puzzle Yes, the second 'ps' below showed the parent to be '1' (init), which means its real parent died already. Any attempt to flush the iptables hangs :( j On Wed, 19 Nov 2008, Danny Nicholas wrote: Have you done a ps -elf to see if the process has a parent that is re-launching or preserving it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Wednesday, November 19, 2008 1:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] puzzle Sorry again for the only marginal relation to asterisk, but the issue does affect the voice performance I am experiencing, so I am soothing my guilt with that. Bet you don't see this every day: ast% uptime 13:48:08 up 981 days, 18:29, 1 user, load average: 1.08, 1.02, 1.01 ast% I *REALLY* want this machine to see 1000 days uptime, if for nothing other than bragging rights. Its been through mysql and asterisk upgrades, a horrible hacking nightmare that very nearly made me reboot, and several power outages where the batteries lasted JUST long enough to keep her up. After all of this, I find I may have to reboot after all. Because there is a [EMAIL PROTECTED] process running, consuming 100% CPU (note the load average), and I cannot seem to kill it: ast% ps auxw | grep modprobe root 17744 99.9 0.0 2688 412 ?RN Nov03 23223:01 modprobe -r ipt_state ast% ps ealx | grep modprobe | grep -v grep 4 0 17744 1 39 19 2688 412 - RN ?23223:38 modprobe -r ipt_state ast% sudo kill
Re: [asterisk-users] Upgrading Asterisk and FreePBX from 1.2 to 1.4
Carlos Chavez wrote: I have a new customer that wants to upgrade their Asterisk installation from 1.2.27 to 1.4.22. They use FreePBX for administration. Since there are many syntax and command changes from those versions of Asterisk, is there an easy way to convert the FreePBX configuration so it will work with the newer Asterisk? Unless you have a lot of custom dialplan components in there, the only thing you need to be sure of is that you are running FreePBX 2.3 (I believe - possibly 2.2) or later. If you are running a very old version of FreePBX, then you will need to upgrade it /before/ you upgrade to Asterisk 1.4. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
