Re: [asterisk-users] Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)
HI! Quoting Michael Nausch mich...@nausch.org: You installed the module, but did you load it in modules.conf? No, 'cause the modul should be autoloaded, as on Asterisk 1.6 it has done. I know a little bit more, after testing this/last night! If I start asterisk 1.8 with service asterisk start or /etc/init.d/asterisk start, I can't load chan_misdn.so If I run asterisk 1.8 as root via asterisk -vvvc I can access my ISDN-card and I be able to dial out to my PSTN provider! ;) Example: *CLI == Using SIP RTP CoS mark 5 [Dec 1 10:49:47] ERROR[16779]: chan_sip.c:27876 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Executing [089216750...@default:1] Dial(SIP/14-, mISDN/g:Mnet/089216750916) in new stack -- Called g:Mnet/089216750916 -- mISDN/1-u1 is proceeding passing it to SIP/14- -- mISDN/1-u1 is ringing -- mISDN/1-u1 answered SIP/14- == Spawn extension (default, 089216750916, 1) exited non-zero on 'SIP/14-' P[ 0] received 1k Unhandled Bchannel Messages: prim 20081 len 0 from addr 52010101, dinfo 0 on this port. *CLI O.K., but what's the difference between asterisk-1.6 and asterisk-1.8, or why won't asterisk-1.8's startscript produce this error? I have: # cat /etc/sysconfig/asterisk AST_USER=asterisk AST_GROUP=asterisk # grep 97 /etc/passwd asterisk:x:97:97:Asterisk_System_User:/home/asterisk:/sbin/nologin # grep 97 /etc/group asterisk:x:97: If I run asterisk with service asterisk start i have two processes running, asterisk and asterisk_safe: # ps aux | grep asterisk root 16946 0.0 0.0 4624 544 pts/2S11:07 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk asterisk 16956 0.4 11.4 139472 118060 pts/2 Sl 11:07 0:01 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c If I start asterisk as root I have only one asterisk process running: # ps aux | grep asterisk root 17079 1.0 11.4 160508 118180 pts/2 Sl+ 11:14 0:00 asterisk -vvvc I'm a little bit confused - I think is better to go to bed and sleep a few hours. ;) n8! Django -- Bonnie Clyde der Postmaster-Szene! approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org pgpQeMoPB4a7q.pgp Description: Digitale PGP-Unterschrift -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Broadsoft-like BLF List URI ?
Hello, I've seen several references in IP phones manuals to Broadsoft's BLF-List URI feature (also referred to as List-Oriented BLF). With this mechanism, a server is able to update the BLFs an IP Phone is supervising without asking the IP phone to reboot, as for a reason I don't know, most phones send BLF-related SUBSCRIBEs during boot time. Is this feature compatible with Asterisk design ? Is there a way to update BLFs without rebooting ? (I could successfully update BLF but I could force the IP phone to subscribe withour rebooting). Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)
On 1 Dec 2010, at 10:18, Michael Nausch wrote: If I start asterisk 1.8 with service asterisk start or /etc/init.d/asterisk start, I can't load chan_misdn.so If I run asterisk 1.8 as root via asterisk -vvvc I can access my ISDN-card and I be able to dial out to my PSTN provider! ;) File permissions? If you run with init.d script it may be running under 'asterisk' user not 'root'. If files are not readable by asterisk it wont load it. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi on Realtime.
Good morning list. I wonder if I can put files and chan_dahdi dahdi_channels in real time. Not the generals but the channels. Thanks, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] solved! Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)
Hello I fixed my problem. I changed user and group in /etc/mISDN.conf: devnode user=asterisk group=asterisk mode=644mISDN/devnode now it works again! Thanx 4 help! ;) Ingrid -- Bonnie Clyde der Postmaster-Szene! approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org HI! Quoting Michael Nausch mich...@nausch.org: You installed the module, but did you load it in modules.conf? No, 'cause the modul should be autoloaded, as on Asterisk 1.6 it has done. I know a little bit more, after testing this/last night! If I start asterisk 1.8 with service asterisk start or /etc/init.d/asterisk start, I can't load chan_misdn.so If I run asterisk 1.8 as root via asterisk -vvvc I can access my ISDN-card and I be able to dial out to my PSTN provider! ;) Example: *CLI == Using SIP RTP CoS mark 5 [Dec 1 10:49:47] ERROR[16779]: chan_sip.c:27876 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Executing [089216750...@default:1] Dial(SIP/14-, mISDN/g:Mnet/089216750916) in new stack -- Called g:Mnet/089216750916 -- mISDN/1-u1 is proceeding passing it to SIP/14- -- mISDN/1-u1 is ringing -- mISDN/1-u1 answered SIP/14- == Spawn extension (default, 089216750916, 1) exited non-zero on 'SIP/14-' P[ 0] received 1k Unhandled Bchannel Messages: prim 20081 len 0 from addr 52010101, dinfo 0 on this port. *CLI O.K., but what's the difference between asterisk-1.6 and asterisk-1.8, or why won't asterisk-1.8's startscript produce this error? I have: # cat /etc/sysconfig/asterisk AST_USER=asterisk AST_GROUP=asterisk # grep 97 /etc/passwd asterisk:x:97:97:Asterisk_System_User:/home/asterisk:/sbin/nologin # grep 97 /etc/group asterisk:x:97: If I run asterisk with service asterisk start i have two processes running, asterisk and asterisk_safe: # ps aux | grep asterisk root 16946 0.0 0.0 4624 544 pts/2S11:07 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk asterisk 16956 0.4 11.4 139472 118060 pts/2 Sl 11:07 0:01 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c If I start asterisk as root I have only one asterisk process running: # ps aux | grep asterisk root 17079 1.0 11.4 160508 118180 pts/2 Sl+ 11:14 0:00 asterisk -vvvc I'm a little bit confused - I think is better to go to bed and sleep a few hours. ;) n8! Django -- Bonnie Clyde der Postmaster-Szene! approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org pgpQxNeMacvVf.pgp Description: Digitale PGP-Unterschrift -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME krytographische Unterschrift -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reasons of OriginateResponse
Good morning everyone. I wonder where I can find a list of the reasons the event OriginateResponse. I found this list [1]. But in my Asterisk has other reasons too. [1] 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Thanks in advanced, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unit of measurement dahdi_monitor
**Thanks a lot, this was been very helpful 2010/11/26 Moises Silva moises.si...@gmail.com On Thu, Nov 25, 2010 at 11:54 AM, Gustavo Santos gust...@voip.ufrj.brwrote: I am studying about echo cancellation in asterisk and I want to use the numeric information from dahdi_monitor verbose for my research. Unfortunately, I couldn't find anything about the unit of measurement used in this tool. Which unit is used to measure the signal level? dahdi_monitor uses the sample values in L16 format. They are in orders of magnitud of G.711. See tables 5 and 6 of the G.711 spec. In the end, the reference value is the dBm (google that). Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on smartphone?
Thanks everyone for the feedback. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing
anyone have a idea on this.. On 11/22/2010 10:50 AM, Nikhil wrote: Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing
On Wed, Dec 1, 2010 at 6:56 AM, Nikhil d.nik...@cem-solutions.net wrote: anyone have a idea on this.. On 11/22/2010 10:50 AM, Nikhil wrote: Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I can tell you that back in 2005 I set up a complete residential/business VOIP ITSP that was using just the Asterisk CDRs to track billing. As long as you setup your billing, I see now reason wh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing
On Wed, Dec 1, 2010 at 5:56 AM, Nikhil d.nik...@cem-solutions.net wrote: anyone have a idea on this.. On 11/22/2010 10:50 AM, Nikhil wrote: Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil Nikhil-- Yes, there is a problem with CDR's in asterisk. There is a bug filed against the problem, but no practical solution for it. The cure: use the CEL interface instead, and generate your own CDR's (or whatever else you could bill from [it doesn't *have* to be CDRs]) The cause of the problem: An interesting combination of 3 operations being performed on channels, namely masquerading, and forming local channels; add to that the hardwired 'roles' that channels are given (channel, and peer), and the final knockout: CDRs are stored on channels. The above 3 operations affect CDR's because parking and transfers can change the role that a channel plays (chan to peer or vice-versa); Transfers and parking involve the masquerading, and sometimes local channels-- both these operations involve duplicating a channel. CDR's are meant for the channel playing the channel role, and CDRs on channels playing the peer role are tossed out. Transfers turn the above into a complex mush in which the results vary depending on who transfers who, who is calling, and who is called, etc. Because the CDR's are stored on the channels themselves, they pass thru many filters and operations that vary based on the roles they play and the operations performed, which can change in subtle ways from release to release, from one bugfix to another, even. So, the best way to get around all this is to get the CDRs out of the channels, And to do that, you either use CEL, or you build your own event tracking mechanism, using whatever means available (I've seen folks use the manager event reports with their own logic in perl, or php, or whatever to do the parsing and storage). Then, you either turn the events into your own billing mechanism, or you generate CDR's to fit into your currently existing billing mechanism. Doing this right and making it dependable is not an overnight sort of thing. I proposed a CDR generating backend for CEL (which I haven't completed yet). I even started it, but got layed off before I could finish it. I've generated a spec for CEL-CDR generation, involving something I call simple CDRs.This doc has been evolving with time, and needs to be updated. I previously described complex CDR's in the spec that provided more fine-grained event logging in CDR format, but I've convinced myself that the complex stuff can also be done via the simple method, and so I'm about half way thru the spec, expunging the complex stuff. All my examples have to be changed -- If you are interested in looking at my spec, you can: svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs and look at the pdf there in that directory. murf Steve Murphy ParseTree Corp. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reasons of OriginateResponse
On 10-12-01 07:22 AM, Rodrigo Lang wrote: I wonder where I can find a list of the reasons the event OriginateResponse. I found this list [1]. But in my Asterisk has other reasons too. You'll have more control if you use local channels together with Originate. You'll then have access to dialplan variables. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with TE122 on HP DL120G6 - can't disable USB
Shaun, thanks for the reply and the hints! In article 4cf3e7d4.6080...@digium.com, Shaun Ruffell sruff...@digium.com wrote: On 11/29/2010 11:11 AM, Tony Mountifield wrote: I have recently built a single-T1 Asterisk box using an HP DL120G6 with a Digium TE122 card. I was finding that I was getting missed interrupts on the TE122, causing the driver to report that it was increasing latency. It kept doing this until the T1 did not work reliably. DAHDI does add idle buffers which can allow the max latency to be caped at something low. This change went in revision 7517 [1]. You would still have data problems in the channel but you wouldn't have to worry about the framer getting confused. Some other things you might try: 1) Is there an option for legacy keyboard emulation in your BIOS that you could disable? It could be that there is a long running System Management Interrupt running to see if it should make the USB keyboard look like a PS/2 keyboard for DOS, etc.. Couldn't find anything like that. It looks like the kbd and mouse are just implemented as USB on this hardware. 2) Do you have the latest BIOS for the DL120G6? The box was brand new recently, although I assume the BIOS doesn't play any part once Linux is booted and running. 3) Update your kernel to the 2.6.32 stable series in case the problem really is in the USB stack. That would be a bit tricky at the moment as it's using CentOS 4. I'll be interested to explore CentOS 6 when it is released. 4) Use /proc/irq/IRQ num/smp_affinity to force the USB interrupts onto CPU0 and the TE122 interrupts onto CPU1 (assuming the DL120G6 is dual core). This seems to have helped, and the box now appears to run reliably. I needed first to disable the irqbalance daemon, and then I made an init script that would bind the TE122 interrupt to core 3 and all the other interrupts to cores 0-2 (it's a quad-core CPU). Thanks again for your help. Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing
On Wed, Dec 1, 2010 at 7:41 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Wed, Dec 1, 2010 at 6:56 AM, Nikhil d.nik...@cem-solutions.net wrote: anyone have a idea on this.. On 11/22/2010 10:50 AM, Nikhil wrote: Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil I can tell you that back in 2005 I set up a complete residential/business VOIP ITSP that was using just the Asterisk CDRs to track billing. As long as you setup your billing, I see now reason wh Oh, and by the way, Sherwood is correct; you can ignore everything I wrote previously, IF you don't do call transfers, and you don't park calls. And even if you do a small amount of that, as long as no one forwards an incoming call to some international destination, you'll probably be OK with CDR's. If you don't mind losing a little chunk of the conversation here or there, the current CDR's should be sufficient for you, and you don't have to go thru any bother. Just keep in mind that clever people can/will take advantage of the fact that everything after an incoming call is transferred is lost to billing (as an example). murf Steve Murphy ParseTree Corp. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor not recording in version 1.8
Greetings.
Just updated from 1.4.22 to 1.8. Minor changes in dialplan and things work
ok. Except for one thing.
I have a call to MixMonitor. This is implementing a dictaphone kind of app.
With forwarding recordings to email and storing them on the server.
The process works so that we dial into Asterisk and answer the phone,
initiate MixMontior and WaitExten until recording finishes.
Problem is that in 1.8 the MixMonitor does not begin recording, ever (when
applied as shown below). I've tried MixMonitor on the same server with
bridged channels and this is no problem and works as expected.
Question is, is there a way to force MixMonitor to work on 1.8 as it used to
on 1.4.22?
Dial plan (AEL) is as follows (excerpts):
// BEGIN OF SAMPLE
incoming {
555 = {
jump 0...@dicta;
}
}
dicta {
= {
Answer(1000); // Slight initial pause to allow audio to
balance
Playback(beep);
// ***
// THIS IS WHERE the problem lies.
// This call does NOT start recording at this time!
// It used to work, in 1.4.22. But in 1.8 it does not.
// ***
MixMonitor(myfilename.alaw,,mv myfilename.alaw
myfinishedfilename.alaw);
jump 0...@dicta-while-recording;
}
}
dicta-while-recording {
0001 = {
WaitExten(400); // This is effectively the maximum length of
a recording!
}
}
// END OF SAMPLE
Any help is greatly appreciated.
Best regards,
Baldvin
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[asterisk-users] Dahdi 2.4.0 and unplugged spans
Hi, I'm facing an issue with which loading wctdm24xxp module fails. Here is relevant dmesg's output : [ 13.455729] dahdi: Telephony Interface Registered on major 196 [ 13.455729] dahdi: Version: 2.4.0 [ 13.510847] ACPI: PCI Interrupt :01:0b.0[A] - GSI 22 (level, low) - IRQ 22 [ 15.527788] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 17.527787] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 19.527787] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 21.527785] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 23.527784] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 25.527785] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 27.527785] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 29.527785] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 31.527785] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 33.527942] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 33.527995] wctdm24xxp :01:0b.0: The firmware may be corrupted. Please completely power off your system, power on, and then reload th e driver with the 'forceload' module parameter set to 1 to attempt recovery. [ 33.544741] ACPI: PCI interrupt for device :01:0b.0 disabled [ 33.544757] wctdm24xxp: probe of :01:0b.0 failed with error -5 [ 34.157817] dahdi: Registered tone zone 2 (France) [ 38.873158] warning: `ntpd' uses 32-bit capabilities (legacy support in use) [ 47.462040] dahdi: Detected time shift. My setup is : Asterisk box with HA8+B400M ---ISDN/BRI Patton SN4638 SIP--- Asterisk Patton SN4638 is configured to behave as telco node (ie NT/PtmP). To narrow my research scope, I'm wondering if the followin sentence is true : if you load a dahdi module with a modprobe command (like modprobe wctdm24xxp) disconnect any cable of any kind connected to the corresponding PCI board, should this command succeed or fail In other words, do you need to connect board to a public network when configuring it ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing
On Wed, Dec 1, 2010 at 9:17 AM, Steve Murphy m...@parsetree.com wrote: On Wed, Dec 1, 2010 at 7:41 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Wed, Dec 1, 2010 at 6:56 AM, Nikhil d.nik...@cem-solutions.net wrote: anyone have a idea on this.. On 11/22/2010 10:50 AM, Nikhil wrote: Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil I can tell you that back in 2005 I set up a complete residential/business VOIP ITSP that was using just the Asterisk CDRs to track billing. As long as you setup your billing, I see now reason wh Oh, and by the way, Sherwood is correct; you can ignore everything I wrote previously, IF you don't do call transfers, and you don't park calls. And even if you do a small amount of that, as long as no one forwards an incoming call to some international destination, you'll probably be OK with CDR's. If you don't mind losing a little chunk of the conversation here or there, the current CDR's should be sufficient for you, and you don't have to go thru any bother. Just keep in mind that clever people can/will take advantage of the fact that everything after an incoming call is transferred is lost to billing (as an example). murf Steve Murphy ParseTree Corp. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Good call Steve! Your comment is spot on, as you said, transfers screw it all up. My prior client did not allow transfers, since it was just a simple VOIP provider service, without transfer capabilities for customers :) Good to see you around! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with MySQL Cluster
Awesome. Didn't notice that, but that is my fault for not reading the changelog or the updated sample configs. I will try this out. Thanks all for the comments. On Wed, Dec 1, 2010 at 10:09 AM, Tilghman Lesher tles...@digium.com wrote: On Tuesday 30 November 2010 18:34:17 Duane Larson wrote: I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant High-Availability. I was wondering if it's possible for Asterisk to also use multiple database servers for Realtime? Currently with Realtime I am only able to point to a single IP address for a database. If that database server goes down that Asterisk is pointed to then Asterisk won't be able to do anything. Any options within Asterisk 1.8 to make it more fault tolerant when it comes to Realtime and databases? Yes, if you refer to configs/extconfig.conf.sample, within the Asterisk 1.8 tree, you'll see that realtime supports multiple lines per realtime family, scored by consecutive priorities. 1 is the default, but you can have as many as you'd like. Additionally, for res_config_odbc, there is a setting in res_odbc.conf called negative_connection_cache, which is the length of time that Asterisk remembers that a connection is down before it will once again attempt to connect. The intention, of course, is that once the primary comes back up, you'll want the Asterisk server to revert back to using it. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi 2.4.0 and unplugged spans
On 12/01/2010 09:56 AM, Olivier wrote: I'm facing an issue with which loading wctdm24xxp module fails. Here is relevant dmesg's output : [ 33.527942] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 33.527995] wctdm24xxp :01:0b.0: The firmware may be corrupted. Please completely power off your system, power on, and then reload th e driver with the 'forceload' module parameter set to 1 to attempt recovery. [ 33.544741] ACPI: PCI interrupt for device :01:0b.0 disabled [ 33.544757] wctdm24xxp: probe of :01:0b.0 failed with error -5 The essential thing here is that your Hx8 board doesn't appear to be responding. I've seen certain systems where revision 9397 [1] works around cases like what you're seeing. If that doesn't help your best bet is to contact Digium technical support for help with troubleshooting. [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9397 To narrow my research scope, I'm wondering if the followin sentence is true : if you load a dahdi module with a modprobe command (like modprobe wctdm24xxp) disconnect any cable of any kind connected to the corresponding PCI board, should this command succeed or fail In other words, do you need to connect board to a public network when configuring it ? No, the current alarm state of any ports doesn't matter when you configure the board. The configuration of modules on a given Hx8 basecard will not affect whether the driver is able to load or not. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi 2.4.0 and unplugged spans
On 12/01/2010 11:00 AM, Shaun Ruffell wrote: On 12/01/2010 09:56 AM, Olivier wrote: I'm facing an issue with which loading wctdm24xxp module fails. [ 33.527942] wctdm24xxp :01:0b.0: Timeout waiting for receive frame. [ 33.527995] wctdm24xxp :01:0b.0: The firmware may be corrupted. Please completely power off your system, power on, and then reload th e driver with the 'forceload' module parameter set to 1 to attempt recovery. [ 33.544741] ACPI: PCI interrupt for device :01:0b.0 disabled [ 33.544757] wctdm24xxp: probe of :01:0b.0 failed with error -5 The essential thing here is that your Hx8 board doesn't appear to be responding. I've seen certain systems where revision 9397 [1] works around cases like what you're seeing. If that doesn't help your best bet is to contact Digium technical support for help with troubleshooting. [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9397 I forgot to add, please update to the current trunk and retry as opposed to checking out revision 9397 specifically. ]# svn co http://svn.asterisk.org/svn/dahdi/linux/trunk dahdi-linux-trunk ]# cd dahdi-linux-trunk ]# make install -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Queue_log and CDR.
Good afternoon list. I am facing a problem with the CDR and Queue_log tables (MySQL). The ABANDON events is being saved correctly in queue_log, but in the table CDR is not saving the registry of such abandoned calls. Apparently the CDR table is functioning normally, I have several records of links in it. From what I noticed, is only the events abandonment that are malfunctioning. With this SELECT [1] I can pick up the records on other servers without problems. With this another SELECT [2], I get the events Abaddon normally. With this other [3] I can get all the channels that joined the queue with no problems. Both tables are recording normally and Asterisk has no errors in the logs. Only happens with the event Abandon. [1] SELECT from_unixtime(ql.time),ql.callid,ql.event FROM cdr c,queue_log ql WHERE c.uniqueid = ql.callid AND ql.event = 'ABANDON'; [2] SELECT * FROM queue_log WHERE event = 'ABANDON'; [3] SELECT from_unixtime(ql.time),ql.callid,ql.event FROM cdr c,queue_log ql WHERE c.uniqueid = ql.callid AND ql.event = 'ENTERQUEUE'; Thanks in advanced, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issues with 1.8 and BlindTransfer
I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use the # to initiate the transfer everything works. But our customers are use to using the transfer key on the phone. I found several bugs out there on the bug tracker but do not see if there is any work around. Any ideas or help would be appreciated. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / Asterisk on Solaris
Hi RR, As far as I am aware the version of Zaptel on SolarisVoIP is out of date. Aditionally the versions of the packages compiled at SolarisVoIP are only available, as far as I am aware, for the Solaris platform and not the OpenSolaris platform, there may be subtle differences between the two that may be causing your build error. If you have a look at SolarisVoIP there are pre-built packages for SPARC/X86 hardware which you do not need to build yourself. In saying all of the above, your millage may vary with zaptel running in a VM as the timing is virtualized (via usb) and is not, as far as I know, very well supported within a VM. Thanks Bruce From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR Sent: 01 December 2010 00:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zaptel / Asterisk on Solaris Hello nice people :) I have been struggling with trying to get Zaptel from http://www.slimey.org/zaptel-solaris.tar.gz on a Solaris 5.11 VM I obtained from the OpenSolaris Website. I have tried installing all the necessary packages, yet I keep getting errors no matter if I try using the gcc available at sunfreeware.comhttp://sunfreeware.com OR the blastwave CSWgcc packages and GNU 'gmake' (as suggested somewhere on the Internet). I have tried sending emails to the people at SolarisVoIP.com and To Simon, from Slimey.org who built/created this Zaptel Solaris Port, but it's been over 2 weeks and I've not heard anything from anyone. This is EXTREMELY critical for me to work...can anyone kind generous gentleman please help? Thank you so much \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct operation of timout parameter for dial application
Hi All,
Does anyone have any thoughts on the question below, or do you think it may be
a question for the dev list?
Thanks
Bruce
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce McAlister
Sent: 30 November 2010 13:15
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Correct operation of timout parameter for dial
application
Hi All,
I'd just like to verify what the correct operation of the timeout parameter is
for the dial application. I'm not sure if I've encountered a bug or a
configuration issue.
When a sip phone is not responding to invites on an outbound call, the dial
application still waits the duration of timeout before continuing with dialplan
execution. I was under the impression that app_dial would timeout on the
signalling prior to the timeout parameter specified in the dial parameter.
For example, consider the following dialplan:
exten = 111,1),Dial(SIP/phone1,30,tg)
exten = 111,n,NoOp(DialStatus=${DIALSTATUS})
exten = 111,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = NOANSWER]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = BUSY]?busy)
exten = 111,n,GotoIf($[${DIALSTATUS} = CONGESTION]?busy)
exten = 111,n(unavail), Goto(voice-mail,vmu-phone1,1)
exten = 111,n(busy), Goto(voice-mail,vmb-phone1,1)
Under normal operation the originating caller is passed through to voicemail.
However, if/when the device is not responding to invites, for whatever reason,
the dial application waits 30 seconds before setting the DIALSTATUS to
NOANSWER. Is this expected behaviour? In previous versions of asterisk,
specifically (v1.2/v1.4) when the device did not respond to invites the dial
application exited prior to the value specified by timeout.
Can anyone clarify this issue for me please? Is this expected behaviour?
We are currently running v1.6.2.13
Thanks
Bruce
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Re: [asterisk-users] Abandon events in cdr
Sorry, of course cdr.conf not queues.conf. marcus Am 01.12.2010 19:16 schrieb marcus rothe sync...@googlemail.com: Hi Rodrigo, have you got enabled the appropriate line in queues. Conf? Regards Marcus Thanks very much, I include the line unansweredy=yes in the cdr.conf and solve the problem. Thanks again! -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] codec_g729a implicated in file descriptor buildup
Hello, I wonder if anyone else has noticed this. I see a pair of calls to pipe() within the codec_g729a, and suddenly, I have a leaked file descriptor that remains until asterisk dies. Now, maybe no-one sees this, mainly because I have no g729 licenses on the machines where this happens. And conversely, I haven't yet studied servers that do have licenses. Why have codec_g729a.so loaded if I don't have licenses? Well, I can install licenses on the run as needed this way, and not worry about having to install anything, or muck things up if there is a mistake. I can mod the phones and get g729 used without restarting asterisk or loading modules. On completely quiet machines, with no calls in or out, I get one descriptor per day, maybe of a daily reload or something. I haven't gotten that far in my investigations yet. Since the module isn't compiled with debug info, the best stack trace I can get is: #0 0x4d5544a0 in pipe () from /lib/libc.so.6 #1 0xb69384ce in __cxa_finalize () from /usr/lib/asterisk/modules/codec_g729a.so #2 0xae7fdae4 in ?? () #3 0xae7fcae4 in ?? () #4 0x1000 in ?? () #5 0x in ?? () The version of the g279 module is: Digium G.729A Module Version 1.6.2.0_3.1.4 (optimized for generic_32) Just now, on a very low-volume asterisk server I am monitoring, two calls just got processed. The g729a codec did a pair of pipe() calls, and voila! I have one more open file descriptor as reported by lsof. Some of my servers (which are busy, but nowhere near capacity!) will build up 100 such leaked descriptors per day, and unless I jack up the maximum number of file descriptors, those servers will have to be restarted about every 10 days, or they will eventually stop accepting calls (or making them, for that matter). Not nice. So, since there is no list of problems fixed with the current g729a module distribution, (at least, no in the README in the dist, is this a problem that is known? Is this a new problem? Should I call support? Anybody else see this? murf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec_g729a implicated in file descriptor buildup
On 12/01/2010 01:05 PM, Steve Murphy wrote: Hello, I wonder if anyone else has noticed this. I see a pair of calls to pipe() within the codec_g729a, and suddenly, I have a leaked file descriptor that remains until asterisk dies. Now, maybe no-one sees this, mainly because I have no g729 licenses on the machines where this happens. And conversely, I haven't yet studied servers that do have licenses. Why have codec_g729a.so loaded if I don't have licenses? Well, I can install licenses on the run as needed this way, and not worry about having to install anything, or muck things up if there is a mistake. I can mod the phones and get g729 used without restarting asterisk or loading modules. On completely quiet machines, with no calls in or out, I get one descriptor per day, maybe of a daily reload or something. I haven't gotten that far in my investigations yet. Since the module isn't compiled with debug info, the best stack trace I can get is: #0 0x4d5544a0 in pipe () from /lib/libc.so.6 #1 0xb69384ce in __cxa_finalize () from /usr/lib/asterisk/modules/codec_g729a.so #2 0xae7fdae4 in ?? () #3 0xae7fcae4 in ?? () #4 0x1000 in ?? () #5 0x in ?? () The version of the g279 module is: Digium G.729A Module Version 1.6.2.0_3.1.4 (optimized for generic_32) Just now, on a very low-volume asterisk server I am monitoring, two calls just got processed. The g729a codec did a pair of pipe() calls, and voila! I have one more open file descriptor as reported by lsof. Some of my servers (which are busy, but nowhere near capacity!) will build up 100 such leaked descriptors per day, and unless I jack up the maximum number of file descriptors, those servers will have to be restarted about every 10 days, or they will eventually stop accepting calls (or making them, for that matter). Not nice. So, since there is no list of problems fixed with the current g729a module distribution, (at least, no in the README in the dist, is this a problem that is known? Is this a new problem? Should I call support? Anybody else see this? This problem may be in the license file checking code... I've just taken a quick look at it, and there may be at least one code path that leaks a pair of pipe file descriptors. I'll enter an internal issue to get this addressed ASAP. Thanks for the report. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with 1.8 and BlindTransfer
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote: I am having issues with Blind Transfer on asterisk 1.8 What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. Does the remote party (being transferred) initially hear hold music, then the line go silent after completing the transfer? Does the Grandstream show the call still on hold, but you are unable to pick it up? Are you using Realtime and/or Direct media? It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use the # to initiate the transfer everything works. But our customers are use to using the transfer key on the phone. I found several bugs out there on the bug tracker but do not see if there is any work around. Any ideas or help would be appreciated. I have been chasing a deadlock (issue #18403) on blind transfers with 1.8SVN and have not found a work-around yet. While I can deadlock every time (Polycom and Cisco handsets), at least one other has reported different results with the Bria Softphone and Grandstream handsets. You could try a softphone and see if you get the same results as the physical phones. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / Asterisk on Solaris
Hi Bruce, Thanks for responding to my message. Doesn't seem like anyone is or is interested in running Asterisk on Solaris or if they are then they're being very secretive / quiet about it as I need a bit of help. Yes, I do know that SolarisVoIP people do have pre-built packages out there that I can simply install without having to deal with compiling them which I have actually done for Asterisk. Alas, the version of Asterisk they have in that pkg is v1.2.7.1. Also, don't know if you've recently looked at the page at SolarisVoIP but they do have a package for OpenSolaris v5.11 which is a fairly recent Solaris version however it beats me why it comes with Asterisk v1.2.7.1 instead of 1.4 or 1.6 even. Anyway, so yeah like I said, I have that installed, but I can't start it as it fails trying to look for Zaptel stuff. Zaptel package isn't installing though ...crashes midway complaining that: *Operating environment requirement not met. This package requires Solaris 7 or better. checkinstall script suspends* huh? I'm running 5.11, which according to some rigorous mathematical calculations, I concluded IS better than v5.7. Unfortunately, I've been away from the development world so long that I can't remember where to go about hacking a package and extract the scripts etc to change the logic or fix whatever is causing it to believe that my OS isn't meeting the min. req. Lastly, w.r.t to running it within a VM, yes, I do understand the timing problems etc, but this exercise is just to document how to compile Asterisk/Zaptel under Solaris 10/11 so when I do get a real Solaris machine, I have already sorted out all the issues with installing/compiling etc Thanks \R On Wed, Dec 1, 2010 at 1:10 PM, Bruce McAlister bruce.mcalis...@blueface.ie wrote: Hi RR, As far as I am aware the version of Zaptel on SolarisVoIP is out of date. Aditionally the versions of the packages compiled at SolarisVoIP are only available, as far as I am aware, for the Solaris platform and not the OpenSolaris platform, there may be subtle differences between the two that may be causing your build error. If you have a look at SolarisVoIP there are pre-built packages for SPARC/X86 hardware which you do not need to build yourself. In saying all of the above, your millage may vary with zaptel running in a VM as the timing is virtualized (via usb) and is not, as far as I know, very well supported within a VM. Thanks Bruce *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *RR *Sent:* 01 December 2010 00:55 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Zaptel / Asterisk on Solaris Hello nice people :) I have been struggling with trying to get Zaptel from http://www.slimey.org/zaptel-solaris.tar.gz on a Solaris 5.11 VM I obtained from the OpenSolaris Website. I have tried installing all the necessary packages, yet I keep getting errors no matter if I try using the gcc available at sunfreeware.com OR the blastwave CSWgcc packages and GNU 'gmake' (as suggested somewhere on the Internet). I have tried sending emails to the people at SolarisVoIP.com and To Simon, from Slimey.org who built/created this Zaptel Solaris Port, but it's been over 2 weeks and I've not heard anything from anyone. This is EXTREMELY critical for me to work...can anyone kind generous gentleman please help? Thank you so much \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi on Realtime.
On Wed, 2010-12-01 at 10:15 -0200, Rodrigo Lang wrote: Good morning list. I wonder if I can put files and chan_dahdi dahdi_channels in real time. Not the generals but the channels. There is no specific Realtime database for chan_dahdi (that I know if). You can store the configuration using Realtime Static using the new chan_dahdi.conf notation without any problems. The only problem with Realtime Static is that you cannot use the text file, you need to load everything from the database. Another possibility would be to use an #exec from chan_dahdi.conf to extract the channel configuration from the database. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI 2.4.0 produces HDLC errors with echo canceler
On Mon, Nov 29, 2010 at 10:02 AM, Shaun Ruffell sruff...@digium.com wrote: On 11/27/2010 11:03 AM, James Lamanna wrote: Hi, After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC errors on my console: [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 These errors prevented calls from being made and received on my PRI spans. This seems similar to bug 15498: https://issues.asterisk.org/view.php?id=15498 Which says this was fixed in 2.2...so maybe it got back into 2.4? I can get rid of the errors by disabling the mg2 echo canceler in /etc/dadhi/system.conf. Do you have a hardware echocan module installed on your card? If so, it's strange indeed that the error goes away when you disable mg2. What is the complete output of your /etc/dahdi/system.conf? Nope, No h/w echo canceler on this card. Here's system.conf: span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 # Global data loadzone= us defaultzone = us -- James -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing
Thanks,Now I understand the problem,Now I am trying to change CDR to fix these issues. Thanks Nikhil On 12/01/2010 08:31 PM, Steve Murphy wrote: On Wed, Dec 1, 2010 at 5:56 AM, Nikhil d.nik...@cem-solutions.net mailto:d.nik...@cem-solutions.net wrote: anyone have a idea on this.. On 11/22/2010 10:50 AM, Nikhil wrote: Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil Nikhil-- Yes, there is a problem with CDR's in asterisk. There is a bug filed against the problem, but no practical solution for it. The cure: use the CEL interface instead, and generate your own CDR's (or whatever else you could bill from [it doesn't *have* to be CDRs]) The cause of the problem: An interesting combination of 3 operations being performed on channels, namely masquerading, and forming local channels; add to that the hardwired 'roles' that channels are given (channel, and peer), and the final knockout: CDRs are stored on channels. The above 3 operations affect CDR's because parking and transfers can change the role that a channel plays (chan to peer or vice-versa); Transfers and parking involve the masquerading, and sometimes local channels-- both these operations involve duplicating a channel. CDR's are meant for the channel playing the channel role, and CDRs on channels playing the peer role are tossed out. Transfers turn the above into a complex mush in which the results vary depending on who transfers who, who is calling, and who is called, etc. Because the CDR's are stored on the channels themselves, they pass thru many filters and operations that vary based on the roles they play and the operations performed, which can change in subtle ways from release to release, from one bugfix to another, even. So, the best way to get around all this is to get the CDRs out of the channels, And to do that, you either use CEL, or you build your own event tracking mechanism, using whatever means available (I've seen folks use the manager event reports with their own logic in perl, or php, or whatever to do the parsing and storage). Then, you either turn the events into your own billing mechanism, or you generate CDR's to fit into your currently existing billing mechanism. Doing this right and making it dependable is not an overnight sort of thing. I proposed a CDR generating backend for CEL (which I haven't completed yet). I even started it, but got layed off before I could finish it. I've generated a spec for CEL-CDR generation, involving something I call simple CDRs.This doc has been evolving with time, and needs to be updated. I previously described complex CDR's in the spec that provided more fine-grained event logging in CDR format, but I've convinced myself that the complex stuff can also be done via the simple method, and so I'm about half way thru the spec, expunging the complex stuff. All my examples have to be changed -- If you are interested in looking at my spec, you can: svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs and look at the pdf there in that directory. murf Steve Murphy ParseTree Corp. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
