Re: [asterisk-users] Problem with Portech MV-372
I've seen that behaviour on he MV-374. One possible solution (workaround) is to prevent the gateway from registering itself and to declare each of the channels explicitly in sip.conf via its associated IP + port. -- exvito ___ -- 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] Agent with External Number as Extension
On Wed, Nov 25, 2009 at 11:41 PM, Shaun Clark shaun_cl...@hotmail.com wrote: Can you add an agent dynamically to a queue with an external number, i.e. cell phone as an extension? If so how? Thanks! Maybe adding the channel Local/PSTN-number@context-that-dials-PSTN to the queue as a member will be good enough for you ? -- exvito ___ -- 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] Problems with dahdi on asterisk 1.6.1.9 with TE122
Oliver, Without any experience with Asterisk 1.6.1.x here it goes: - Is your signalling=pri_net correct or should it be pri_cpe ? - I'm having myself some issues with DAHDI 2.2.0.2 + TE121 which, I reckon, is equal to your TE122 apart from the bus interface (PCIe vs PCI) -- see the thread about DAHDI and odd idle system load for references to two issues on the bugtracker (this comes mostly as a FYI as I woudn't say your issue is related to it). Cheers, -- exvito ___ -- 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] TE121 - Idle system load at ~0.3 - Bad DAHDI 2.2.0.2 behaviour ?!
Shaun, Thanks for your feedback. See my inline comments. On Fri, Nov 13, 2009 at 7:18 PM, Shaun Ruffell sruff...@digium.com wrote: It appears there may be a regression in dahdi-linux 2.2.0 with regards to the wcte12xp driver and the VPMADT032 module (as discussed https://issues.asterisk.org/view.php?id=15724). Would you be willing to try at least revision 7584 of http://svn.asterisk.org/svn/dahdi/linux/branches/2.2 and report your results on that issue? We will, either on 15724 or on 15798, both, I'd say, closely related to what we're experiencing. First, however, we will test 2.1.0.4 for at least 10 days to effectively confirm are experiencing the regression you refer. We defined 10 stable days because we've been experiencing 1 - 3 failures per week with 2.2.0.2. So, we will move to 2.1.0.4 later today and, if there are no incidents, we will move to SVN revision 7584 or later on Nov 27th. The idle load you're seeing can be a little misleading, but essentially, once you load the drivers for both the wctdm24xxp and wcte12xp, there is a fixed cost associated with continuously moving the TDM data to/from the card. The load imposed by the drivers would only go up after this point if a) software echocan is enabled, or b) you're conferencing many calls in the kernel. Otherwiseit's fixed. I agree the load can be misleading however, consider: - The system is really idle - zero calls, zero activity, no other software. - 0.3 to 0.4 load on a modern CPU is a lot of processing, is there that much data to move around ?! :) We tested some variations and here is what we found: (recall, AEX410 with no VPM is physically installed) DAHDI 2.2.0.2 - Removed TE121 - Idle load is 0 - TE121 without VPM - Idle load is 0.3 - 0.4 - TE121 with VPM - Idle load is 0.3 - 0.4 DAHDI 2.1.0.4 - TE121 without VPM - Idle load is 0.01 - 0.02 - TE121 with VPM - Idle load is 0.01 - 0.05 DAHDI 2.2.0.1 - TE121 with VPM - Idle load is 0.26 - 0.32 DAHDI 2.2.0 - TE121 with VPM - Idle load is 0.31 - 0.36 DAHDI SVN-branch-2.2-r7584 - TE121 with VPM - Idle load is 0.69 - 0.82 Under all cases, we stopped asterisk, unloaded DAHDI, rebuilt DAHDI (+Asterisk if needed), loaded DAHDI, started Asterisk, waited 120s, tested one inbound call + one outbound call. In short: - DAHDI 2.1, Idle load is 0 - DAHDI 2.2, Idle load is 0.3 - 0.4 - DAHDI 2.2. SVN, Idle load is roughly twice as in 2.2 releases So, while your explanation regarding system load makes sense, I find it odd that 2.2, which apparently is not working ok for us in this case, is showing such behaviour when 2.1 does not and neither Zaptel 1.2/1.4 used many other systems I've installed. What would you say the explanation for these observations is ? Would you say that there is a correlation between the load and the suspected not so good behaviour we're getting from DAHDI 2.2.0.2 on our case ? Or is it just a coincidence ? Cheers, -- exvito ___ -- 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] BLF with SPA941?
Although I've never tested such feature on those devices, I know that it was only enabled in a recent firmware (6.1.3a/6.1.5a ?). Are you running it ? -- exvito ___ -- 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] TE121 - Idle system load at ~0.3 - Bad DAHDI 2.2.0.2 behaviour ?!
Hi Asterisk Users, We've been experiencing some tough time regarding a new Asterisk installation connected to the PSTN via an ISDN PRI with a Digium TE121 with the optional VPMADT032 echo cancellation module. For now, I'll focus on something very specific which is summarized on this email's subject. However, here are some general facts for the context: - System pbxfri went into production about a month ago. - System pbxfrv is HW+SW copy+paste of pbxfri not in production yet. - Had several incidents where the PSTN connection was not operational (calls had bad quality/echo or PRI trunk could not be used for either inbound or outbound) - Most of the incidents (maybe all of them, haven't verified thourougly) are asso- ciated to hundreds/thousands of HDLC Abort / Bad FCS messages in the asterisk log. - DAHDI + Asterisk + libpri never seemed to recover from those conditions. We manually had to stop Asterisk, unload+load DAHDI, start Asterisk. - Had at least on kernel panic on DAHDI load. - We have logs + traces and are working with the telco so as to try to fully diagnose what's going on here. For now we'd like to focus on the following (but if you think we should start somewhere else, please, by all means, fire away!): - Lots of info out there (google) seems to associate the HDLC Abort / Bad FCS with a system hardware issue - whatever it is: interrupts, badly behaved NICs, disk array controllers, etc. Question #1: What do these messages actually mean ? Can they be associated to a bad link/telco switch configuration ? - We've noticed that the system load at idle is about 0.3 when DAHDI is loaded. If we unload DAHDI, system load at idle goes to appoximately 0, as expected. Question #2: This looks like a very odd behaviour. We've installed several other systems (different HW/SW versions, however) without seeing such behaviour. Is this expected or could this be related with the HDLC Aborts / Bad FCS and general failures we've been experiencing ? System info (same for both): HW: HP Proliant ML310 G5 TE121 + VPMADT032 AEX410 + 4x FXS + without DSP OS: CentOS 5.3, kernel 2.6.18-164.el5 DAHDI:2.2.0.2 libpri: 1.4.10.2 Asterisk: 1.4.26.2 Here is a session transcript for pbxfrv (not in production) showing the odd DAHDI / system load behaviour. It starts with DAHDI unloaded: # uname -a Linux pbxfrv.replaced 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686 i386 GNU/Linux # cat /proc/cmdline ro root=/dev/vg0/lv00 console=tty0 console=ttyS1,115200 # cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 25288985 25275219 25290489 25274409IO-APIC-edge timer 1: 3 0 0 0IO-APIC-edge i8042 3: 24819 20503 24395 19262IO-APIC-edge serial 8: 14 16 13 11IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 12: 3 0 1 0IO-APIC-edge i8042 74: 0 0 0 0 IO-APIC-level ehci_hcd:usb1, uhci_hcd:usb2, uhci_hcd:usb3, uhci_hcd:usb4, uhci_hcd:usb5 82: 21 24 21 30 IO-APIC-level uhci_hcd:usb6 90: 17 16 14 16 IO-APIC-level ata_piix, ata_piix 106: 77476 0 0 0 PCI-MSI eth0 169:1912615190964619113021910566 IO-APIC-level ioc0 NMI: 0 0 0 0 LOC: 101129266 101132004 101132444 101128234 ERR: 0 MIS: 0 # uptime 17:52:15 up 1 day, 4:07, 1 user, load average: 0.00, 0.07, 0.06 # dmesg ... ACPI: PCI interrupt for device :05:08.0 disabled Freed a Wildcard ACPI: PCI interrupt for device :08:08.0 disabled Freed a Wildcard TE12xP. dahdi: Telephony Interface Unloaded # /etc/init.d/dahdi start Loading DAHDI hardware modules: wcte12xp:[ OK ] wctdm24xxp: [ OK ] Running dahdi_cfg: [ OK ] # dmesg ... dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.0.2 PCI: Enabling device :08:08.0 (0150 - 0153) ACPI: PCI Interrupt :08:08.0[A] - GSI 19 (level, low) - IRQ 185 wcte12xp: VPM present and operational (Firmware version 117) wcte12xp: Setting up global serial parameters for E1 wcte12xp: Found a Wildcard TE121 PCI: Enabling device :05:08.0 (0150 - 0153) wcte12xp0: Missed interrupt. Increasing latency to 4 ms in order to compensate. ACPI: PCI Interrupt :05:08.0[A] - GSI 18 (level, low) - IRQ 177 Port 1: Installed -- AUTO FXS/DPO Port 2: Installed -- AUTO FXS/DPO Port 3: Installed -- AUTO FXS/DPO Port 4: Installed -- AUTO FXS/DPO VPM100: Not Present
Re: [asterisk-users] callfile to auto-answering extension
2009/9/27 Leif Neland le...@neland.dk: Can I, via a callfile, or command-line parameters to Asterisk start a dialplan-script? eg asterisk -someflag execute callalert then in dialplan [callalert] exten = s,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = s,2,dial(SIP/36) exten = s,3,Playback(firealert) exten = s,4,Hangup ...sure, use Local channels. You can use Local/ext@context as the originating channel in a call file or AMI/CLI originate command. -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Know for how long an agent is talking?
On Sun, Sep 27, 2009 at 12:07 AM, Gabriel Ortiz Lour ortiz.ad...@gmail.com wrote: Hi, Is there a way to know for how long an agent is talking on the queue call? (without keeping a timer myself... just asking asterisk) Identify the channel at the CLI and then get its details via core show channel channel-spec. Asterisk will gladly give you lots of details regarding that channel, including the channel uptime. Does this answer your question or did I misunderstood it ? -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where are phone registrations kept?
I've been willing to give such a solution a try but the lack of time has prevented it to date... Are you using realtime for your SIP peers/users ? Would the failover behaviour improve under such scenario ? (just a thought) -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SUN and PRI ?
The system specs mention PCIe expansion slots, so your only option is the TE420B. -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local music on hold -- mohinterpret=passthrough assymetrical ?
On Fri, Apr 3, 2009 at 11:11 AM, Richard Brady rnbr...@gmail.com wrote: Exvito Did you ever make any progress on this? ...no, sorry. Never got to the perfect solution. (and in all due honesty, I can't recall the exact setup we ended up deploying) -- exvito ___ -- 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 -f and restart now
# rasterisk Connected to Asterisk 1.4.23 currently running on debian (pid = 17191) samuel*CLI restart now samuel*CLI Disconnected from Asterisk server Executing last minute cleanups Asterisk ending (0). # rasterisk Connected to Asterisk SVN-branch-1.4-r178373 currently running on debian (pid = 17191) This is interesting: Asterisk is running a new version, but still the same PID... man 3 exec -- exvito ___ -- 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] call file bug?
On Tue, Feb 17, 2009 at 8:04 PM, Ray Chen ray1...@techie.com wrote: I have a problem of using call file to make an auto dial out call through FXO channel. I defined the channel in the call file as Channel: DAHDI/1/8775203463 When I put the call file under the /var/spool/asterisk/outgoing dir it did not call out but came to the context I defined in extensions.conf as if the callee had answered the call. If I make a call from an extension to DAHDI/1/8775203463 it'll success. . If I change the channel to SIP/8000 and put the call file under /var/spool/asterisk/outgoing it is also success - it calls the extension 8000 and the controle goes to the context after the extension 8000 answers the call. I am using asterisk 1.4.23.1. Is it a bug introduced in this release? It's probably the result of FXO lines having very little signalling. IOW, asterisk picks up the FXO line and dials the number... By then it has no way of knowing wheather the other party answered or not. That's probably why your getting your other leg too early in the process. Maybe you could try to Wait() for a few seconds on your dialplan. -- exvito ___ -- 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] Contact lookup
On Thu, Feb 5, 2009 at 7:22 AM, Geoff Lane ge...@gjctech.co.uk wrote: The nice thing about that is that if I use MySQL I can run the management application on another machine, and so don't need to run a web server on the Asterisk box. However, I wonder whether the overhead necessary to run MySQL on the Asterisk box is more than that required to run Apache to provide a web interface to astdb. I'm not running either at present, which is probably as well since my Asterisk machine is low-spec by todays standards. Regarding system resource usage it is, of course, to you to run the DB engine along with asterisk or on some other system. :-) -- exvito ___ -- 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] Incoming fax detection on mISDN hfcmulti B410P card
App nvfaxdetect() works fine for that purpose on both Zap and mISDN. See http://www.voip-info.org/wiki-NVFaxDetect -- exvito ___ -- 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] early dial: asterisk and ATA
On Tue, Feb 3, 2009 at 5:04 PM, Vieri rentor...@yahoo.com wrote: I did but apparently, there's nothing in the guides that lets me do this. It's something about supporting 484 responses that Grandstream GXW4008 seems to do and Linksys SPA8000 doesn't (or at least it's not documented). In other words, the SPA8000's L1-L8 Dial Plan parameter only allows for matches to be performed entirely on the device and not via 484 ADDRESS INCOMPLETE responses with Asterisk's dial patterns. Sorry, I failed to fully understand your question. I'm not sure if the SPAs will dispatch partial numbers and manage 484 responses like the GS gear seems to do. -- exvito ___ -- 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 on originate call
On Wed, Feb 4, 2009 at 7:40 PM, Jerry Geis ge...@pagestation.com wrote: Seems like the first call to Channel is being MADE successfully. Then it goes to do Context and Exten: I get failed... [smvoice-dialout] exten = smvoice_single_mediaport,1,agi(smvoice) exten = smvoice_single_mediaport,n,Hangup I can't identify nothing specific, apart from the fact that you're running non-standard Asterisk applications. My humble suggestions: 1. Increase log verbosity and check logs 2. Break the problem into something simpler (example: Grab one extension and Play a file into it) When you get to a working setup, build up from there, one step at a time... -- exvito ___ -- 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] Contact lookup
For a simple (but flexible) case I would consider ODBC + func_odbc.
Here is the idea (in case you aren't aware of how it goes...)
- Make a DB available (your choice as long as it is accessible via ODBC)
- Create table in it with your contacts (say columns number and
name, maybe more)
- Setup an ODBC connection for asterisk so that it can connect to that DB
(res_odbc.conf)
- Setup an ODBC func.This is basically an SQL query which will be
mapped into a dialplan function. (func_odbc.conf) It is essentially
something that states my function ODBC_LOOKUP(arg) will give me
the results of SELECT name FROM contactsTable WHERE number=${arg}
into the dialplan.
- Then use it in the dialplan
exten = _x.,n,Set(CALLERID(name)=${ODBC_LOOKUP(${EXTEN})})
There! Your dialplan is almost directly executing SQL queries. :)
Check both the sample asterisk configs + Asterisk TFOT, chapter 12.
It may be a bit more work than using the Ast DB or other means, but it
has the advantage of allowing the easy setup of any kind of frontend for
contact management.
Note: Check for the correctness of my filenames/syntax... They're shown
just to fill in the idea with something resembing the reality!
My 2c,
--
exvito
___
-- 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] siemens hipath 4000
Any suggestions? Jerry Are you sure asterisk is to behave as signalling=pri_cpe or should it be pri_net ? -- exvito ___ -- 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] Music On Hold
On Mon, Feb 2, 2009 at 8:39 AM, Idris AVCI idris.a...@vodatech.com.tr wrote: In my situation AMI is not an option. When somebdy puts a call on hold, on asterisk console I can see messages like Started music on hold, class 'default', on SIP/ and Started music on hold, class 'default', on SIP/. I guess the only way in my scenerio is to modify res_musiconhold.so. ...and for each of those console messages an AMI event is fired; it should be relatively simple to attach script execution to those events. Also, I confess that I'm curious as to what environment you're running where using AMI is not an option but hacking res_musiconhold.so is... Good luck, anyway. -- exvito ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
(my 2c, Portugal Based) - Most really small installations are PtMP (that's the default you get when ordering a BRI) - You also get 3 MSNs and an NT. - You can order a TA instead of the NT. - You can order PTP + optional DDIs in blocks of 10, but you need to be explicit. - Larger installations go PTP (up to 4/5 BRIs... from there up, a PRI is cost effective which, as far as I've managed to get from the local telcos, can only be ordered with 15 channels or more) My experience points to 60/40 in PtMP / PTP installations. While I understand Tzafrir's question (why go PtMP from Asterisk to PBX), I think Olivier's statement is very valuable: because going NT PtMP allows you insert/remove Asterisk between PSTN and legacy PBX without any reconfiguration (beyond physical cables!). Cheers, -- exvito ___ -- 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] dialstatus through a call file
On Tue, Jan 27, 2009 at 10:21 PM, Pascal Bruno tipas...@gmail.com wrote:
Is it possible to retrieve the DIALSTATUS variable when placing call through
a call file. This variable is set when using the Dial() application from
the dialplan, but I am using a call file for my current application and need
to get the dialstatus.
Your call file will initiate actions defined in the dialplan and
certainly after
the triggered Dial the DIALSTATUS will be available to the dialplan.
Now the question is: where do you want to retreive the DIALSTATUS to ?
If back to the OS environment (a file ?) you will need to have your dialplan
do it for you, maybe via System(echo ${DIALSTATUS} /tmp/file) or
something...
(NOTE: i'm not sure of the syntax of the application... check it
with core show
application System on the CLI)
--
exvito
___
-- 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] Music On Hold
On Fri, Jan 30, 2009 at 3:23 PM, Danny Nicholas da...@debsinc.com wrote: The dialplan AFAIK doesn't cover HOLD handling. If you can spare the overhead, you can make a daemon to watch hints and run a script whenever the hint for a line goes to hold and changes from hold to inuse. Just run asterisk –rx core show hints and asterisk –rx core show channels and integrate the 2 outputs. For your purpose, you can probably just use the first command. You should instaed use the AMI and create an event based solution instead of relying on polling via asterisk -rx !... Check out: http://www.voip-info.org/wiki-Asterisk+manager+API http://www.voip-info.org/wiki/view/asterisk+manager+events Cheers, -- exvito ___ -- 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] early dial: asterisk and ATA
On Thu, Jan 29, 2009 at 6:15 PM, Vieri rentor...@yahoo.com wrote: I'm trying to do the same in the SPA8000 units but without any luck. If anyone is doing something similar with this device then I'd appreciate it if you could share your relevant config options (dial pattern, etc.). Not sure about the SPA8000, but the SPA devices I know (phones + 2102 ATA) all have a per line Dial Plan paramenter that will allow you to acheive that behaviour. Check link sys ATA / Phone Admin Guides. -- exvito ___ -- 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] Managing codecs
Assuming you are using SIP phones and IIRC, you can hint at the codec to be used by setting the SIP_CODEC variable in the dialplan; before Dial()'ing, of course ! :-) I think this is still an area where asterisk needs improvement... Dynamic codec (re) negotiation. Anyone care to correct me ? Cheers, -- exvito ___ -- 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] Using centos and kickstart to build a minimum installation
Anyone done anything similar before that would care to share ? Here is a snippet of our standard ks file for automated installs: %packages --nobase @core sendmail sendmail-cf ntp vixie-cron crontabs at logrotate telnet bind-utils lsof wget which unzip man bc nc sharutils # Up till now, just regular unix tools -- useful in a running system, not really # required for running / building asterisk... # The packages needed to build asterisk follow gcc ncurses-devel libtermcap-devel kernel-devel gcc-c++ openssl-devel newt-devel zlib-devel make unixODBC-devel libtool-ltdl-devel sox # sox is obviously not needed and unixODBC-devel + libtool-ltdl-devel allowed us # to have the ODBC funcs in asterisk... I'm not really sure why c++ is there, I wouldn't # say it's needed. # Then, for mISDN we still have: # needed For mISDN flex usbutils Cheers, -- exvito ___ -- 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] mISDN BRI Asterisk 1.4
While I don't know the OpenVOX B200P specifics, some interface cards need you to change physical jumpers in order to acheive NT vs TE, mode. Could that be the case ? -- exvito ___ -- 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] Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5
On Mon, Jan 5, 2009 at 8:20 AM, Nick Wolf new...@gmail.com wrote: besides this, I paste my zaptel.conf : span=1,1,6,ccs,hdb3 span=2,1,6,ccs,hdb3 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=fr defaultzone=fr when I put 6 in LINE BUILD OUT (1,1,6,ccs,hdb3) value I got better call quality than when it was 5 (1,1,5,ccs,hdb3). IIRC, the second argument in the span lines indicates the timing sync with 1 meaning that this span is master. I'd say it makes no sense to have both of them be masters... I have no current docs / system at hand; give it a check and then, maybe try to have one as master and the other as slave (0 instead of 1, again, IIRC). Cheers, -- exvito ___ -- 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] queue log parser
On Mon, Jan 5, 2009 at 10:12 PM, David fire ddf...@gmail.com wrote: if you don't know any parser maybe you can send me a link or a pdf whit info on how to parse the log. ...check queuelog.txt under the doc/ directory on the asterisk source distribution (apparently, under 1.6 it is queuelog.tex... no more txt ?!) Format for each entry is along the lines of timestamp | uniquecallid | queuename | event-data-as-per-queuelog.txt ...timestamp is in seconds since the epoch (as obtained with: date +%s). Also, check the wiki + TFOT... Both should be more than good enough starting points. Of course, this can also be useful: http://tinyurl.com/9k8r97 Good luck, :-) -- exvito ___ -- 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] Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5
On Sun, Jan 4, 2009 at 8:37 AM, Nick Wolf new...@gmail.com wrote: I forgot to describe the audio problem, well, I experience micro cuts in the voice, this does not happen during the whole call, it happens during 2 seconds then audio becomes normal, then back again 2 or 3 seconds then goes away. I'd start by checking for interrupt conflicts / sharing on the system. cat /proc/interrupts... also check your zap timing accuracy (was it zaptest ?) It looks as if there is some periodic IO (disk?) activity that is leading to bad audio. (also: check your logs !) :-) Then I'd add that the first impressions with the ML110 G5 were not very good regarding IO activity and the PCI bus... Not really sure what it was, as it was in the context of a few quick tests we did during the summer, but I think we had to change the BIOS defaults for the disk controller or something because we were getting really sloow IO from them. Question: Any particular reason to run 1.2 instead of something more recent ? -- exvito ___ -- 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] IAX2 softphones keep ringing....
On Sat, Dec 20, 2008 at 5:54 PM, Jerome Deyle jde...@gmail.com wrote: Running AsteriskNow, with FreePBX front end. Have two users who use softphones on notebooks in the field. Problem is that if the softphone receives a call, but the user is not available to pick it up, Asterisk will send the call to voice mail as normal, but the softphone will continue to ring.. I've tested the Virbiage and x-Lite softphones, same issue. Is there an IAX setting in Asterisk that applies here? Jerome What version of asterisk are you running ? IIRC I experienced similar behaviours with 1.4.22... I'm now running 1.4.19 (as far as this version, because of other issues, fixed in 1.4.22 -- but IAX non-hangups broke 1.4.22 for me) -- exvito ___ -- 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] 1.4.22 vs 1.4.21.2 - IAX2 regression ?
On Tue, Nov 4, 2008 at 3:12 PM, Igor Zamocky [EMAIL PROTECTED] wrote: http://bugs.digium.com/view.php?id=13645 Thanks Igor, we'll keep an eye on it. -- exvito ___ -- 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] 1.4.22 vs 1.4.21.2 - IAX2 regression ?
Hi list,
I just experienced an odd behaviour in 1.4.22 vs 1.4.21.2.
To cut a long story short, IAX2 is not tx-ing hangup...
Scenario is composed of two asterisk systems A and B.
A receives calls from IAX users X, Y, Z, etc, does some
validation and forwards them to B, also over IAX.
When B hangs up, it transmits IAX hangup which A receives
who, in turn, does not transmit the IAX hangup to its user
X, Y or Z. So X, Y or Z still think the call is up...
All of this is verified with iax debug... A receives the hangup but
never hangs up the other side if running 1.4.22. Everything is ok
if running 1.4.21.2.
Could this be something we're doing wrong ? What steps would
you suggest for further diagnostic?
Thanks in advance for any feedback.
System A runs 1.4.22 / 1.4.21.2
System A iax.conf
[userX]
type=user
transfer=no
host=dynamic
secret=whatever
context=the-context
disallow=all
allow=alaw
allow=ulaw
[systemB]
type=peer
qualify=200
transfer=no
host=ip-here
disallow=all
allow=gsm
System A extensions.conf:
[the-context]
exten = _.,1,Wait(1)
exten = _.,n,Set(CALL_UUID=${EXTEN})
exten = _.,n,Set(RESULT_STRING=${ODBC_CALL_DATA_4_UUID(${CALL_UUID})})
exten = _.,n,Set(ARRAY(NAME,ACCT,IAXUSER,NUM)=${RESULT_STRING})
exten = _.,n,Set(DONT_CARE=${ODBC_REMOVE_CALL_4_UUID(${CALL_UUID})})
exten = _.,n,Set(CALLERID(name)=${NAME})
exten = _.,n,Set(CDR(accountcode)=${ACCT})
exten = _.,n,Dial(IAX2/[EMAIL PROTECTED]/${NUM})
exten = _.,n,Hangup()
(note: behaviour is also failing in 1.4.22 if, instead of Dialing
system B, we just wait+hangup directly here!)
System B runs asterisk 1.2.30.1
System B iax.conf:
[one-systemA-user]
type=user
context=one-context
notransfer=yes
disallow=all
allow=gsm
System B extensions.conf:
[one-context]
exten = _N,1,Dial(.../${EXTEN})
exten = _N,n,Hangup()
--
exvito
___
-- 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] Decent Voip Phones for enterprise
...as others have mentioned, yes they do have the ability to be centrally managed, provisioned, configured. Also, from the latest firmware, 6.x.x: - Ability to use line buttons as quick dials - Ability to query centralized LDAP for directory (I haven't tested this one yet) So, the phones you have may be able to address all your needs, without forcing you to replace them with something else. -- exvito ___ -- 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] OT - How to test tftp for phones provisioning
So let's say, you've got : a perfectly running tftp server somewhere on your LAN, it holds foo.txt file in its /srv/tftp directory. Which command could you type in for a LAN workstation to receive this foo.txt ? tftp is the client, do you have it installed ?... example: # tftp hostname tftp get /srv/tftp/foo.txt tftp ^D # cat foo.txt ... Things to check: is /srv/tftp the tftp directory or is it the os filesystem directory where the tftp root resides ? Also, the tftp daemon in CentOS is started by xinetd and can be invoked with extra -v flags so as to increase logging verbosity. Check your dist. This may help... Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] different gains per channel?
I need to have different gain settings on each channel. Is this easy to achieve? txgain, rxgain and many other parameters are defined on a per-channel basis in zapata.conf, they're not global. Each channel definition channel = x assumes previous definitions of such parameters. Example: txgain=3.0 rxgain=0.0 channel = 1 channel = 2 Both channel 1 and 2 will have the same gains (both previously defined). txgain=3.0 rxgain=0.0 channel = 1 txgain=-4.5 rxgain=0.0 channel = 2 Now you get different gains for channels 1 and 2. Again, as defined before the channel = x definition. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?
I recently observed a similar behaviour under 1.4.21. The member was a SIP phone which had its calls forwarded to another SIP phone via its built-in configuration... (fyi: linksys spa922) For some reason, asterisk could not manage this scenario. I still have to test it better to understand if this is supposed to work or not. Could that be your case ? (not very probable, I know...) -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.20.1 hang -- extra info + gdb hangs
Here is an update, 1. Reviewed 'core show locks' with the help of russellb @ #asterisk-devs last friday 2. Recommended recompilling asterisk with DONT_OPTIMIZE and getting a stack trace with: # gdb /usr/sbin/asterisk $(pidof asterisk) (gdb) set pagination off (gdb) thread apply all bt We did reinstall asterisk with the new compile flags back then and just experienced another hang now (weekend, monday and tuesday were very low activity days). Unfortunatelly, gdb seems to hang on startup, after what seems to be a thread list. It never gets to the reading symbols from... steps. As such, no gdb prompt - no stack trace ! :-/ ps shows gdb process as defunct and, as such, it responds to no signals; asterisk seems to not respond to signals as well... (maybe that's why gdb hangs... I really do not know how gdb works in regards to attaching itself to a running process) Again we have a 'core show locks' + 'core show threads' output from asterisk which we have no skills to read... Lastly, asterisk log displays 12x... [Jun 11 09:41:07] ERROR[4837] chan_sip.c: SIP transaction failed: [EMAIL PROTECTED] [Jun 11 09:41:07] ERROR[4837] chan_sip.c: We could NOT get the channel lock for SIP/000e08de4cbe-097555c8! ...then... [Jun 11 09:41:19] WARNING[4837] chan_sip.c: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Request) ...and finally about 1200 of these: [Jun 11 09:42:59] WARNING[4842] chan_iax2.c: Max retries exceeded to host 192.168.166.40 on IAX2/private-13779 (type = 6, subclass = 11, ts=40022, seqno=10) ...with several combinations of: - the number inside WARNING[xxx] - 13 different - the host IP: 192.168.166.40 and 192.168.170.40 - the iax channel - 12 different Till today, our gut feelings were: 1. The TC400B installation / usage change (idea: asterisk responds to no signals because it is waiting in kernel space, maybe something's wrong with zaptel, wctc4xxp, our HW ?) 2. The activation of a voicemail account with MWI We now have an extra possibility: - This system exchanges IAX calls with several other systems - The hanging one is running asterisk 1.4.20.1, but all the others are running 1.4.19 - The changelog from 1.4.19 - 1.4.20.1 includes several chan_iax fixes -- could the absense of such fixes in this system's iax peers be leading it to hang ? Possibility: 3. Upgrade all peers to 1.4.20.1 Again, if anyone can chime in with their contribution, thanks in advance. Question of the day: why on earth does gdb hang ?! (our guess: because asterisk does not respond to signals... now why ?!) Cheers, -- exvito ___ -- 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] 1.4.20.1 hang -- extra info + gdb hangs
On Wed, Jun 11, 2008 at 12:33 PM, Steve Totaro [EMAIL PROTECTED] wrote: Try switching from IAX to SIP. Steve, thanks for your suggestion... As you may understand that is not an easy decision to take and implement: we're peering with about 20 other systems within a private network where routing/firewalls/QoS etc has been setup considering IAX -- it can be done, of course, but we already have better suspects... :-) After a brief discussion over at IRC, we are seriously suspecting either the TC400B or its driver, wctc4xxp (recall we're running latest asterisk+zaptel). In pursuing some stability and trying to prove that that is the source, we've changed our g729 trunks to gsm and blacklisted wctc4xxp so as to ensure no TC400B is used at all. If this brings us back to a stable system we can almost say for certain that the issue is where we suspect it is: bad HW, bad driver or, as a last resort, bad kernel... (again, fyi, latest centos 5.1) If things go as we expect, we'll then give asterisk-1.4-transcoder + zaptel-1.4-transcoder branches a run while re-enabling wctc4xxp + g729 over the IAX trunks -- hopefully they'll allow the usage of the TC400B along with some stability -- also they'll provide feedback to the developers which is obviously useful in the short and long run (let's have it shorter, shall we?) :) As always, we're open to suggestions and/or further questions. We'll keep posting our experience. Cheers, -- exvito ___ -- 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] 1.4.20.1 hang -- three times in 1.5 days (TC400B at fault ?)
On Fri, Jun 6, 2008 at 1:01 PM, Ex Vito [EMAIL PROTECTED] wrote: In Our Heads -- - we're suspecting that the presence of the TC400B is making asterisk behave in different ways that lead to what we're now calling a hang (that is the apparent change in the system since it started mis-behaving) - as such we're considering removing the TC400B to see if the system stabilizes however removing it may remove the possibility of further diagnosing this issue and trying fixes - of course, we're trying to manage customer expectations and satisfaction at the same time ...other possibility: - instead of removing the TC400B, change the IAX trunk codec to GSM instead of G.729... this would prevent the TC400B usage and may lead to different (as in stable) behaviour More troubleshooting ideas ? -- exvito ___ -- 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] 1.4.20.1 hang -- three times in 1.5 days (TC400B at fault ?)
On Fri, Jun 6, 2008 at 3:16 PM, Shaun Ruffell [EMAIL PROTECTED] wrote: I'm soon going to petition for this interface to be merged into the trunk, so if you would like to try the branches out now and need any help, please contact me directly. Thanks for you feedback Shaun. I've had a quick feedback from russellb @ #asterisk-dev and we'll try next to get a full stack trace when the hang condition occurs. We've already rebuilt with the DONT_OPTIMIZE and had a lucky time-slot to restart asterisk. So, now we're hoping it fails again (ironic, isn't it?) so we can move forward in the diagnostic. Of course, future possibilities of changing codecs, removing the TC400B or others are open (such as: I guess we enabled the 1st voicemail account as test on the same day that we installed the TC400B -- could it be the change ?) We're still open to peer feedback, of course. Post back later. Cheers, -- exvito ___ -- 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] 1.4.20.1 hang -- three times in 1.5 days (TC400B at fault ?)
On Fri, Jun 6, 2008 at 5:01 PM, Andres [EMAIL PROTECTED] wrote: Of course, future possibilities of changing codecs, removing the TC400B or others are open (such as: I guess we enabled the 1st voicemail account as test on the same day that we installed the TC400B -- could it be the change ?) Do you have MWI enabled? We are suspecting a similar SIP deadlock on a system that may be caused by it. Although our version is 1.4.17. There is some mention of it on: http://bugs.digium.com/view.php?id=10953 Yes, on the single test mailbox that is configured. And yes, we are already considering disabling it as a future troubleshooting step... BTW, our voicemail account is realtime ODBC -- exvito ___ -- 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] Server recommendation help
On Tue, May 20, 2008 at 2:14 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, May 20, 2008 at 8:55 AM, Cavanna, Richard [EMAIL PROTECTED] wrote: ... Cards I have installed: Digium TE205P - 5v TDM410 I hear rave reviews about Supermicro but no personal experience. I like the HP DL380. Me to, but beware: current DL380 G5 do not have internal molex power connectors for the TDM410 -- needed for FXS ports. -- exvito ___ -- 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] dundi network - redundancy / fault tolerance ?
On Thu, May 8, 2008 at 3:59 AM, Russell Bryant [EMAIL PROTECTED] wrote: Ex Vito wrote: Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Which documentation are you referring to? You may have misunderstood something, or there may be some false information floating around the internet (*GASP*). Went back and reviewd the docs (essentially: Asterisk TFOT 2nd ed, wiki, the excellent docs by JR Richardson and dundi.com)... ...in short: nowhere is such statement written. I presume we self-inflicted such idea from the best practices mentioned in dundi.com and from the special attention that should be taken when creating looping topologies regarding TTLs. As I said before, don't worry about loops. Set your TTL to handle a worst case path for a query in your DUNDi topology. Great. That's now clear, thanks. - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? ... #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. I'm not necessarily up on my graph theory, either, but I would probably go with something like #1. After internal discussion and reviewing the final example in the DUNDi protocol draft, while agreeing that the differences are actually small, we are also targetting #1... Again, thanks for your quick feedback, Russel. Cheers, -- exvito ___ -- 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] Realtime status feature - user feedback needed
On Thu, May 8, 2008 at 1:23 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Tilghman Lesher a écrit : Your question leads to this question: why don't you create a proxy application that listens on AMI and populates a database outside of Asterisk, then do all your queries to that database? That would provide exactly the same functionality, but it would not require a single change to the Asterisk codebase. You could even contribute that application back as something in the contrib/scripts subdirectory. I second that, If there is already a way to do things, why adding another one, especialy if it's for caching reasons. While we cannot say that asterisk fall into the KISS rule, it's not a reason to let it grow. Agreed. There should be ONE to do it, it should be SIMPLE and as RELIABLE as possible, without interfereing (bad spelling?) with asterisk's operations: the proxy into AMI looks like the way to acheive the required funcionality... After all, that's exactly the purpose of AMI ! Let's keep the codebase as small as possible, let's make asterisk as solid and reliable as possible. Let's not reinvent wheels! -- exvito ___ -- 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] dundi network - redundancy / fault tolerance ?
Hi list, I'm planning a private DUNDi network for a cross-country distributed PBX. Initially it will be composed of about 10 systems, growing to about 20. Current requirements point to a topology of two interconnected DUNDi hubs, each peering with half the PBXs... This would lead to two interconnected / inter-peered stars. Example: - Consider PBXs A to H - C and E will be hubs and peer with each other - A, B and D peer with C - F, G and H peer with E This leads to a maximum three hop lookup and will make good use of current network topology / bandwidths. Of course, should any of the hubs be unavailable and the lookup capability is severely compromised. Now, how to move on to acheive some kind of fault tolerance ? According to the docs we've studied, DUNDi does not like loops (which we assume one can limit with low enough TTLs). Our doubts are: - Should one use the order peer parameter to specify alternate lookup paths / peers ? Is that its purpose ? If not, what is it used for ? - Alternatively, should one create loops in the DUNDi topology and limit them via TTL ? - If both options are possible, which would be the trade-offs between them ? (Not clear at all to us!) - Assuming any of the above is possible as a means to acheive redundancy, which of the following topologies would your prefer ? (hmmm, maybe I need to refresh my graph theory...) ;-) #1 - Peer each PBX with both hubs #2 - Duplicate both hubs and peer each PBX with its hub and its hub dup For better understanding, take a look at: #1 - http://www.2photosharing.com/images/qhpnzycd7j7kf26j2f.png #2 - http://www.2photosharing.com/images/npzbwvgnr4t079laou0.png Thanks in advance for review and feedback. Cheers, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Fri, Apr 18, 2008 at 10:12 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: Matthew, ...is there any specific test you'd like us to perform on this revision ? (considering that currently we have no PSTN line to attach to... we can cross-connect the spans and generate traffic or, cross-connect with another lab system) Not really from me specifically. You already tested what I wanted to be tested, and that was to see if I could fix the load time issue and softlockup warning. Ok. So, since the bug we logged was closed and these tests weren't registered along with it, when can one expect to have your new code available in a zaptel release ? In the next one or maybe later because the branch you're working on has lots of different things to merge ? Thanks in advance, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Mon, Apr 21, 2008 at 4:38 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: ...when can one expect to have your new code available in a zaptel release ? In the next one or maybe later because the branch you're working on has lots of different things to merge ? It should be in the next release. Great. Thanks for your feedback. We will be waiting for it... -- exvito ___ -- 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] Switch recommendation?
We've been very happy with the SRW224Ps we've deployed. (noisy as hell... good for either the datacentre / computer room or for installation in a noise-cancelling cabinet... but then again, are there any PoE switches that aren't ?) -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Fri, Apr 18, 2008 at 4:15 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: I just realized where this is coming from. I was attempting to patch this from a different angle, but as soon as you mentioned the drastic difference in load time I realized what had happened. I'm going to make another update to my stack reduction branch to see if I can fix this. I'll let you know when it's done. Great. We'll be right here... Since the bug has been closed, we post the timing results we did within this context. Recall that, if it helps, we can provide Digium remote system access. System currently has TE220B + TE122, we are willing to redo without the TE122. Timed ztcfg under stock kernel 2.6.18-53.1.14.el5 with 4K stacks: == 1.4.9.2 == # time ztcfg real 0m4.716s user 0m0.000s sys 0m0.029s == 1.4.10 == # time ztcfg real 0m22.778s user 0m0.000s sys 0m0.044s == SVN-mattf-zaptel-1.4-stackcleanup-r4163 == # time ztcfg real 0m22.775s user 0m0.000s sys 0m0.045s Cheers, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Fri, Apr 18, 2008 at 8:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: I just updated the branch. Wait about 5-10 minutes in case for the changes to get mirrored, and then try updating and doing it again. Looks better, no more soft lockup and ztcfg time is comparable to 1.4.9.2's: # time ztcfg real0m4.719s user0m0.000s sys 0m0.044s # dmesg | tail ... Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-mattf-zaptel-1.4-stackcleanup-r4177 Zaptel Echo Canceller: MG2 PCI: Enabling device :12:01.0 (0150 - 0153) ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 138 wcte12xp: Setting up global serial parameters for T1 wcte12xp: Found a Wildcard TE122 Found TE2XXP at base address fdff, remapped to f885 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x36dbd400 Reg 1: 0x36dbd000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x010200ff Reg 9: 0x00fd0001 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12xp: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 SPAN 2: Primary Sync Source timing source auto card 0! VPM400: Not Present wcte12xp: Setting yellow alarm VPM450: echo cancellation for 64 channels wcte12xp0: Missed interrupt. Increasing latency to 4 ms in order to compensate. VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 3: Secondary Sync Source Completed startup! -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Fri, Apr 18, 2008 at 9:36 PM, Ex Vito [EMAIL PROTECTED] wrote: On Fri, Apr 18, 2008 at 8:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: I just updated the branch. Wait about 5-10 minutes in case for the changes to get mirrored, and then try updating and doing it again. Looks better, no more soft lockup and ztcfg time is comparable to 1.4.9.2's: Matthew, ...is there any specific test you'd like us to perform on this revision ? (considering that currently we have no PSTN line to attach to... we can cross-connect the spans and generate traffic or, cross-connect with another lab system) -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 7:18 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: Tested with no 4K stack kernel and stackcleanup svn branch zaptel version. Correct, the kernel no longer complains about the soft hangup. However the system still hangs (console inoperative, etc) while ztcfg'ing... That is normal while the firmware is loading. It should go away after the firmware has loaded. Ok. So here is our reasoning according to collected info. Please correct us where appropriate: 1. The system is supposed to hang while the firmware loads into the DSPs under any zaptel version 2. zaptel 1.4.10 leads to a soft hangup detected, zaptel 1.4.9.2 does not (assuming softhangup detection active in kernel) 3. zaptel 1.4.10 takes much longer ztcfg'ing than 1.4.9.2, that's why the soft hangup is detected under zaptel 1.4.10 (difficult to time, but let's say 1.4.10 takes 10s, 1.4.9.2 takes 3s) Now, back to the original question: - Should this be considered a regression ? - Next steps: a) file a bug and move this analysis to the bug tracker b) don't file bug and move analysis to the dev list c) don't file bug, keep on working on the users list I recommend 1.4.10 by default. However, from what you said it would appear that you are having problems with 1.4.10 so you might stay with 1.4.10 if you are not having any issues with it. Did you mean this instead ? ...so you might stay with 1.4.9.2 if you are not having any issues with it. We think so. Again, thanks for clarifying. Cheers, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Thu, Apr 17, 2008 at 2:36 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Apr 17, 2008 at 02:20:57PM +0100, Ex Vito wrote: - Should this be considered a regression ? Yes, it is a regression, and thus a bug. Mattf has already offered you to work with him on resolving this. FYI, Submitted bug 0012468 as per Tzafrir suggestion. (http://bugs.digium.com/view.php?id=12468) -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Tue, Apr 15, 2008 at 7:07 PM, Shaun Ruffell [EMAIL PROTECTED] wrote: Your stack trace appears to possibly be stack corruption. Could you try either this branch: http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Just tried it... Behaviour looks equivalent. Drivers load ok, ztcfg leads to BUG: soft lockup detected on CPU#1... dmesg snippet is: Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-mattf-zaptel-1.4-stackcleanup-r4163M Zaptel Echo Canceller: MG2 PCI: Enabling device :12:01.0 (0150 - 0153) ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 138 wcte12xp: Setting up global serial parameters for T1 wcte12xp: Found a Wildcard TE122 Found TE2XXP at base address fdff, remapped to f89c4000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x37407400 Reg 1: 0x37407000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x000200ff Reg 9: 0x00f5 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12xp: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 2: Primary Sync Source VPM400: Not Present wcte12xp: Setting yellow alarm VPM450: echo cancellation for 64 channels BUG: soft lockup detected on CPU#1! [c044d448] softlockup_tick+0x96/0xa4 [c042ddc8] update_process_times+0x39/0x5c [c04196f7] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [f8f6b1e7] init_vpm450m+0x32d/0x34a [wct4xxp] [f8f52b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8f56ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042624e] release_console_sem+0x1b0/0x1b8 [c042680e] printk+0x18/0x8e [f8966fe4] t1_configure_t1+0xc10/0xc18 [wcte12xp] [f89945ef] zt_rbs_sethook+0x102/0x13b [zaptel] [f899bf39] zt_ioctl+0x273/0x14be [zaptel] [c045] chrdev_open+0x11e/0x132 [c0477657] chrdev_open+0x0/0x132 [c046e9e6] __dentry_open+0xea/0x1ab [c0604451] schedule+0x90d/0x9ba [c047f46b] do_ioctl+0x47/0x5d [c047f6cb] vfs_ioctl+0x24a/0x25c [c0470daa] __fput+0x13f/0x167 [c047f725] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === wcte12xp0: Missed interrupt. Increasing latency to 4 ms in order to compensate. VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 SPAN 3: Secondary Sync Source timing source auto card 0! Completed startup! wcte12xp: Clearing yellow alarm Or with a kernel that does not have 4K stacks enabled? You can check if your installed kernel does with the following command. $ cat /boot/config-`uname -r` | grep 4K # CONFIG_4KSTACKS is not set ...as mentioned previously, current kernel has CONFIG_4KSTACKS set. I'll now go ahead and rebuild a kernel with 4K stacks disabled. I'll post back later. -- exvito ___ -- 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] Problem with B410P
Could be this... http://www.misdn.org/index.php/FAQ_chan_mISDN#Why_does_the_L1_goes_DOWN_on_my_PMP_Isdn_Link.3F_Or_why_do_i_get_No_free_chan_even_after_group_call_from_chan_misdn_if_dialing_out_on_my_PMP_Link.3F Hmmm... that's a long link. It is the Why does the L1 goes DOWN on my PMP Isdn Link FAQ in the chan_misdn FAQ at http://www.misdn.org/index.php/FAQ_chan_mISDN In short, the telcos shut PMP links down to save on power costs, you need to bring them up before initiating an outbound call. Cheers, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: The softlockup indicator should be benign. It gets called when loaded the firmware for the part since the firmware image is so large and it takes a long time to load. However, I might have a fix for you. Can you try my stack reduction branch at: https://origsvn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup If that does not work, please contact me directly and I will work with you to get a resolution. Matt, Thanks for your feedback. We've already tested the following branch as per Shaun's suggestion, without getting a different behaviour (see today's earlier email to the list): http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Question: - The url you suggest is very similar, are we talking about a different stackcleanup branch ? We are now in the middle of rebuilding a non 4K stack page kernel so as to give it a try with 1.4.10, the branch Shaun suggested, 1.4.9.2 and the branch you mention, if it is in fact different from Shaun's. We wait your confirmation and will post non 4K stack kernel results later today. Cheers, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 4:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: One thing also I would like to see is your kernel .config file. Another thing that would for sure remove that warning is to disable the kernel softlockup detector which is giving a false lockup warning in this case. I belive it's under the KERNEL HACKING configuration menu if you are using menuconfig. Up till now we're running stock CentOS kernel: 2.6.18-53.1.14.el5 The .config is publicly available but we can fwd it to you should you prefer. The kernel we're now building (it is taking quite a while... but it also has been quite a few years since we've built custom kernels... since the 2.0.3x days ?) is based on the stock CentOS kernel with only the 4K stacks option disabled. Please confirm if the SVN branch you suggested is the same or different from the one Shaun suggested yesterday which we already tested. Thanks, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Question: - The url you suggest is very similar, are we talking about a different stackcleanup branch ? Try: http://svn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Try the seocnd one (svn.digium.com), actually. All point to the same place. But origsvn does not allow annonymous access and /view is the viewcvs/viewsvn web interface. So Matt's suggestion is the same as Shaun's... Which we already tested with no different results, correct ? -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 4:46 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: It's the same. Sorry, I sent you that email before I saw his message. I just got an idea for a clever way to make the softlockup detector not complain. I'll let you know when I have a patch to try. ...sure. Thanks. (we're still waiting for the kernel build to finish...) -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
update with no 4K stack kernel: - The kernel was build from stock centos 5 kernel 2.6.18-53.1.14.el5 - The only .config change was to disable the CONFIG_4KSTACKS Tested zaptel-1.4.10, 1.4.9.2 and the stackcleanup svn branch as suggested by Shaun and Mathew. Short: Results are about the same (stack traces are different). 1.4.10 and the stackcleanup lead to soft hangups, 1.4.9.2 does not. 1.4.10 dmesg snippet: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154 wcte12xp: Setting up global serial parameters for T1 wcte12xp: Found a Wildcard TE122 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 162 Found TE2XXP at base address fdff, remapped to f893e000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x3613a400 Reg 1: 0x3613a000 Reg 2: 0x Reg 3: 0x Reg 4: 0x3101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff0031 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12xp: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 2: Primary Sync Source VPM400: Not Present wcte12xp: Setting yellow alarm VPM450: echo cancellation for 64 channels wcte12xp: Clearing yellow alarm BUG: soft lockup detected on CPU#1! [c044d480] softlockup_tick+0x96/0xa4 [c042de00] update_process_times+0x39/0x5c [c04196ef] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [c0605c30] _spin_unlock_irqrestore+0x8/0x9 [f8e82d57] Oct6100UserDriverWriteBurstApi+0x1d/0x27 [wct4xxp] [f8e95de0] Oct6100ApiLoadImage+0x1b5/0x289 [wct4xxp] [f8e9afc4] Oct6100ChipOpen+0x166/0x25e [wct4xxp] [f8e83050] init_vpm450m+0x196/0x306 [wct4xxp] [f8e6ab11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8e6eee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042624e] release_console_sem+0x1b0/0x1b8 [c042680e] printk+0x18/0x8e [f8af6fe4] t1_configure_t1+0xc10/0xc18 [wcte12xp] [f8ac65ef] zt_rbs_sethook+0x102/0x13b [zaptel] [f8acdf6a] zt_ioctl+0x273/0x144f [zaptel] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c0483cb3] __d_lookup+0x98/0xdb [c047b32c] do_lookup+0x53/0x166 [c047d9ec] do_path_lookup+0x20e/0x25e [c0471053] get_empty_filp+0x99/0x15e [c047b5a5] permission+0xa2/0xb5 [c04e1a36] kobject_get+0xf/0x13 [c046ea1e] __dentry_open+0xea/0x1ab [c046eb43] nameidata_to_filp+0x19/0x28 [c046eb7d] do_filp_open+0x2b/0x31 [c047f4a7] do_ioctl+0x47/0x5d [c047f707] vfs_ioctl+0x24a/0x25c [c0470de6] __fput+0x13f/0x167 [c047f761] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 3: Secondary Sync Source Completed startup! 1.4.9.2 dmesg snippet: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.9.2 Zaptel Echo Canceller: MG2 PCI: Enabling device :12:01.0 (0150 - 0153) ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154 wcte12x[p]: Setting up global serial parameters for T1 wcte12x[p]: Found a Wildcard TE122 Found TE2XXP at base address fdff, remapped to f893e000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x3571b400 Reg 1: 0x3571b000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x010200ff Reg 9: 0x00fd0001 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12x[p]: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 SPAN 2: Primary Sync Source timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 64 channels VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 SPAN 3: Secondary Sync Source Completed startup! timing source auto card 0! 1.4-stackcleanup-r4163 dmesg snippet: Zapata Telephony Interface Registered on
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 6:51 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: One thing you can also do is pass the nosoftlockup kernel parameter into the kernel from the bootloader. That should disable the softlockup detector. Tested with no 4K stack kernel and stackcleanup svn branch zaptel version. Correct, the kernel no longer complains about the soft hangup. However the system still hangs (console inoperative, etc) while ztcfg'ing... Can you answer my previous questions ? - If going live would you recommend zaptel 1.4.9.2 or 1.4.10 ? - Does the current behaviour from 1.4.10 prevent firmware uploading ? (or, stated differently: can you explain what is happening that makes the system hang for a few seconds ?) Thanks, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Your stack trace appears to possibly be stack corruption. Could you try either this branch: http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Or with a kernel that does not have 4K stacks enabled? You can check if your installed kernel does with the following command. $ cat /boot/config-`uname -r` | grep 4K # CONFIG_4KSTACKS is not set ...thanks for your feedback Shaun. I am currently nearing other troubleshooting issues regarding a TC400B (which will probably lead me to get in touch with Digium install support). So I have no schedule today to test your suggestions; maybe tomorrow / thursday. They are noted, however. :) Cheers, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Or with a kernel that does not have 4K stacks enabled? You can check if your installed kernel does with the following command. $ cat /boot/config-`uname -r` | grep 4K # CONFIG_4KSTACKS is not set Opps, forgot to feedback: yes this kernel seems to have CONFIG_4KSTACKS enabled. -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Tue, Apr 15, 2008 at 8:37 PM, Al Baker [EMAIL PROTECTED] wrote: exvito - I know it is a pain in the cahoonkus - but would you consider sharing the OTHER Digium board issues you are having , the recommended steps you were given by Digium to troubleshoot them, and the results ? I think this real-wold experience wold be invaluable to the list. THX in Advance for sharing ! ...sure, here it goes, without all the infinite detail we went through in the process. Short version: same DL380 G5 system, Centos 5, kernel 2.6.18-53.1.14.el5, zaptel 1.4.10, zaptel 1.4.9.2, almost all possible combinations in PCI slots, USB / 2nd NIC / ILO enabling / disabling. TC400B module loading fails (wctc4xxp) (actually it loaded fine once or twice and asterisk recognized its presence, but failed in subsequent reboots without any reconfiguration!) If asterisk 1.4.19 is started under these conditions, we get a kernel panic -- did not get a dump / log of it but we have a console picture that we can share (~460KiB). But at some point we get: ... [address] apic_timer_interrupt+0x1f/0x24 [address] zt_tc_open+0x59/0xc3 [zttranscode] [address] zt_open+0x86/0x22a [zaptel] [address] chrdev_open+0x11e/0x123 ... We also tried the same card under all the other variations in a different system -- a proliant ML110 G4 -- we obtained the same behaviour: once or twice it loaded most of the time it failed with the same error. dmesg snippet is: ... Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 Zaptel Transcoder support loaded Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) Zaptel DTE (G.729a / G.723.1) Transcoder support LOADED (firm ver = 6.12) wctc4xxp: probe of :0a:01.0 failed with error -5 ... Both when the card is the only one installed on the system and when in the presence of TE220B and / or TE122. We contacted Digium support, who suggested we RMA this card, they believe the card is faulty. We seem to agree, as the behavior does not seem to make much sense (although this is our first experience with such a card) There it is, in the hope that it helps some one in the future. We will post back results when the new card arrives. Cheers, -- exvito ___ -- 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] Zap Codec
On Tue, Apr 15, 2008 at 11:37 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In the sip peer definition, disallow=all allow=g729 allow=ulaw SHOULD work. Asterisk can't transcode g729, so it should fall on ulaw for the ZAP calls. But, when your polycoms talk with each other, as long as all necessary REINVITEs happen, they should use the 729 codec I believe. Remember however, that many options to the Dial application, like t,w,m,k (or so) REQURE asterisk to remain in the media path. moj AFAICT, I say that in this case this will not work... Very unfortunatelly. It's related to the way the current asterisk versions behave regarding codec negotiation / renegotiation. Your sip.conf entry will have the phone-asterisk leg be g729 and the other leg, to the PSTN, will be a/u-law. When bridging, asterisk is not clever enough (yet!) to renegotiate the SIP leg back to a/u-law and either a) it transcodes or b) the call fails if no transcoder is available... I've given this issue some testing with no sucessful results in the recent past... (check last two/three months list archives) Asterisk really needs a revamped media renegotiation algorithm ! Will we get one in 1.6 ?!... I guess not. Again, unfortunatelly, as this is a very core, very important issue. (feel free to correct me and give me the good news !!!) Cheers, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Hi list, After a lot of testing + troubleshooting, I guess I'm observing what I am now calling a regression with zaptel 1.4.10 (is it?) As such I call for peer feedback, before either asking Digium install support or filing a bug. Thanks in advance! System: HP Proliant DL380 G5 with 2x PCI-X + 1x PCIe riser card OS: Centos 5 Kernel: 2.6.18-53.1.14.el5 (also tested under 2.6.18-53.el5) HW: Digium TE220B, the one with HW echo cancellation (configured as 2x E1 via jumpers) Context: Pre-site installation of system, no E1 conectivity (loopbacks tested) /etc/zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=25-39,41-55 dchan=40 span=2,2,0,ccs,hdb3,crc4 bchan=56-70,72-86 dchan=71 Under zaptel 1.4.10, when ztcfg runs this gets logged in the kernel buffer: About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: echo cancellation for 64 channels BUG: soft lockup detected on CPU#0! [c044d448] softlockup_tick+0x96/0xa4 [c042ddc8] update_process_times+0x39/0x5c [c04196f7] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [f89bc1e7] init_vpm450m+0x32d/0x34a [wct4xxp] [f89a3b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f89a7ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042621c] release_console_sem+0x17e/0x1b8 [c0407406] do_IRQ+0xa5/0xae [f8994311] t4_dacs+0x211/0x24b [wct4xxp] [f8a01f6a] zt_ioctl+0x273/0x144f [zaptel] [c0457600] mempool_alloc+0x28/0xc9 [c04ddd33] cfq_resort_rr_list+0x23/0x8b [c04deb6c] cfq_add_crq_rb+0xba/0xc3 [c04dec72] cfq_insert_request+0x42/0x498 [c04d5175] elv_insert+0x10a/0x1ad [c04d908b] __make_request+0x31d/0x366 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04dde27] __cfq_slice_expired+0x8c/0xa5 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04d505d] elv_next_request+0x15c/0x16a [f88bc101] start_io+0x77/0xdc [cciss] [f88bf63e] do_cciss_request+0x32c/0x337 [cciss] [f88ccff0] __split_bio+0x408/0x418 [dm_mod] [f88cd6a6] dm_request+0xce/0xd4 [dm_mod] [c04d6a81] generic_make_request+0x248/0x258 [c04d8734] submit_bio+0xbf/0xc5 [c04548e2] find_get_page+0x18/0x38 [c04719ad] __find_get_block_slow+0xfb/0x105 [c0471cea] __find_get_block+0x15c/0x166 [c0471cea] __find_get_block+0x15c/0x166 [c0471d24] __getblk+0x30/0x270 [f885a485] journal_cancel_revoke+0x8a/0x96 [jbd] [f885a472] journal_cancel_revoke+0x77/0x96 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c041f871] __wake_up+0x2a/0x3d [f8856679] journal_stop+0x1b0/0x1ba [jbd] [c042a209] current_fs_time+0x4a/0x55 [c048626d] touch_atime+0x60/0x8f [c04552ee] do_generic_mapping_read+0x421/0x468 [c045478b] file_read_actor+0x0/0xd1 [c04548e2] find_get_page+0x18/0x38 [c0457319] filemap_nopage+0x192/0x315 [c046048f] __handle_mm_fault+0x85e/0x87b [c047f46b] do_ioctl+0x47/0x5d [c047f6cb] vfs_ioctl+0x24a/0x25c [c047f725] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 2: Secondary Sync Source Completed startup! Soft lockup ?! Hmmm... I'm ignorant on this, but it smells fishy ! For completeness sake, driver was previously loaded ok: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 98 Found TE2XXP at base address fdff, remapped to f8854000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x375a2400 Reg 1: 0x375a2000 Reg 2: 0x Reg 3: 0x Reg 4: 0x3101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff2031 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) After trying lot's of things (disable ILO, disable USBs, try different kernel, different TE220B, etc), I figured that this soft hangup does not show under zaptel 1.4.9.2... In all due honesty, I haven't got the faintest idea what kind of impact this could have. Side testing zaptel 1.4.10 on a simpler system, an HP Proliant ML110 (nearly a PC), the error does not show up as well. I checked the zaptel 1.4.10 ChangeLog and there are some changes which I'd suspect: 2008-04-01 16:39 + [r4122] sruffell [EMAIL PROTECTED]: * kernel/wct4xxp/base.c: Work around for host bridges that generate fast back to back transactions which the current version of the quad span cards do not advertise support for.
Re: [asterisk-users] RTP Payload Problem
If you are running 1.4, check rtp-packetization.txt under doc/ directory from source distribution. Cheers, -- exvito ___ -- 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] LCR in Asterisk
On Wed, Feb 13, 2008 at 6:49 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: Macros are deprecated. Gosubs are the way forward, and yes, they have local variables. Simply define them once as Set(LOCAL(foo)=bar) and foo will be gone when the innermost stack is removed (either by Return or StackPop). I was keeping up with the list traffic when I found this... Hmmm, maybe to my surpirse. Are macros deprecated in 1.4 or 1.6 ? What, if any, is the replacement for the M() option in the Dial() application ? -- exvito ___ -- 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] How to find out the IP of the calling party?
On Thu, Mar 13, 2008 at 3:47 AM, Gonzalo Servat [EMAIL PROTECTED] wrote: I can't find any channel variable that gives me this info. Gonzalo, With SIP callers you can get the address from the SIPURI channel variable. IAX does not seem to have an equivalent var... The best I could find is the output of iax2 show channels on the CLI wich seems to include the peer IP for running channels. Good luck, -- exvito ___ -- 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] How to find out the IP of the calling party?
Improvement: also check the funcions SIPCHANINFO and IAXPEER... With this and the SIPURI channel variable you should be able to have all the info. Cheers, -- exvito ___ -- 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] RedFone foneBRIDGE2 2e1 - anyone used it oranother TDMoE bridge?
On Thu, Mar 13, 2008 at 9:07 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Which is something not unlike the Junghanns ISDNGuard. ...or the beroNet bero*fos (https://shop.beronet.com/product_info.php/cPath/56/products_id/159) Not affiliated, just a satisfied customer. -- exvito ___ -- 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] dialstatus and cancelled calls
...as long as the destination does not answer you'll get a NO ANSWER disposition. Note, however, that answering can be one of: - Dial a phone and the user answers the phone - Connecting the caller to voicemail, for example, after Dial timed out - Playing an IVR / sound / music - And more... Anything that connects the caller! So, if in your case you want to know if a user answered the phone, then, yes, you will have to add the DIALSTATUS value to the CDR, probably in the CDR's userfield. -- exvito ___ -- 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] Local music on hold -- mohinterpret=passthrough assymetrical ?
Hi list, I'm planning and testing a distributed asterisk deployment throughout several sites; each will be connected to the PSTN and all of them among themselves via IAX trunks. Phones will be SIP. I guess I already solved (worked-around, actually) asterisk's codec negotiation limitations regarding local G.711 utilization vs. remote G.729 while minimizing transcoding -- a bit of dial plan tweaking via the setting of SIP_CODEC variable seems to do the trick. But I digress... (with patch in issue 4825 things would be much nicer!) Now I'm still trying to improve bandwith usage with local music on hold; that is, when sip user A1, registered to server A puts sip caller B1, registered to server B, caller B1 gets server B's music on hold -- this removes the need of streaming audio from server A to server B while B1 is on hold, which in my scenario is a good thing. I post to the list trying to get peer feedback to my initial tests. The configurations I mention are always applied to both servers A and B. 1. If I set mohinterpret=passthrough + mohsuggest=default in the [general] section of iax.conf the local music on hold never works. Results: bad - A1 calls B1, B1 puts A1 on hold, A1 gets B's music bad - A1 calls B1, A1 puts B1 on hold, B1 gets A's music bad - B1 calls A1, A1 puts B1 on hold, B1 gets A's music bad - B1 calls A1, B1 puts A1 on hold, A1 gets B's music 2. If I set mohinterpret=passthrough + mohsuggest=default in the specific peer/user (friend, actually) section I get improved results but not perfect (or, at least, as I'd like them to be). Results: good - A1 calls B1, B1 puts A1 on hold, A1 gets A's music bad - A1 calls B1, A1 puts B1 on hold, B1 gets A's music good - B1 calls A1, A1 puts B1 on hold, B1 gets B's music bad - B1 calls A1, B1 puts A1 on hold, A1 gets B's music Fortunatelly, the good cases seem to be the most plausible ones. So, in my observation, the mohinterpret=passthrough behaviour is not symmetrical; that is, the hold signalling only seems to travel one way along the IAX trunk... From the side receiving the call to the side initiating it, and not the other way around. Can anyone verify this behaviour ? Am I doing something wrong or is this expected / by design behaviour ? Should I file a bug against 1. ? Against 2. ? Extra points question: Since the calls in this case are remote, from site A to site B, the codec in use is G.729 which, as you might well know, is really awfull at supporting music since it's been designed for voice only. How would one have the RTP stream renegotiated during call to G.711 when entering music on hold (local, of course, after fixing my issues above!) and back to G.729 when back to conversation ? (ok, this probably needs patching the source !... but maybe someone has an idea or has taken a different approach at this...) :-) Thanks a lot for any feedback, -- exvito ___ -- 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] Call recording problems from queue
I don't have access to an asterisk system right now (nor any other sort of information source) but I seem to recall that from 1.4 onwards the config option for recording queue calls is named differently... Is it mixmonitor ? Check you 1.4 queues.conf sample. PS: I'm not really sure about this one! -- exvito ___ -- 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] Transferring Unanswered Calls
I wouldn't know how to do it the way you mention it, via local channels... Our implementation performs ringing transfers via AMI redirect... The user action is performed on the desktop, not on the ringing phone. -- exvito ___ -- 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] Had it with Dell Garbage
On Tue, Mar 4, 2008 at 7:54 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Mar 04, 2008 at 03:05:43AM +, Ex Vito wrote: ...can you expand on that please ? I'm on my way to getting one of the newer Digium TE220B PCIe dual T1/E1 to put on such a system. How can I tell if the card has a TigerJet chipset ? Off-topic: Are those just the cards that show up as TigetJet ISDN Modem or Network device on lspci? (and no, the TE220B is not among them) Tzafrir, Tilghman, Darren, ...thanks for the feedback. :-) -- exvito ___ -- 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] Had it with Dell Garbage
On Tue, Feb 26, 2008 at 10:51 PM, Joshua Kinard [EMAIL PROTECTED] wrote: Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very likely, 380's as well). I just learned this the hard way. --J ...can you expand on that please ? I'm on my way to getting one of the newer Digium TE220B PCIe dual T1/E1 to put on such a system. How can I tell if the card has a TigerJet chipset ? -- exvito ___ -- 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] CallerID shows wrong values in manager interface
I've struggled with this recently. In short: - Observed behaviour is expected as of asterisk 1.2 and later, as previously described by Mojo - If you want to get the caller id for the channel calling (dialling) into that channel for that specific Newstate: Ringing event, you can use the 'o' flag to the Dial command; in this case you'll get old asterisk 1.0 behaviour -- do you really want to depend on such an old behaviour ? well I decided I didn't... - Otherwise, you'll need to track other events (IIRC, at least, Dial, AgentCalled, Newstate, etc) in the AMI so as to know who is calling who at a given instant - BEWARE: if memory serves me right (search the list archives in the Nov/Dec timeframe), the behaviour is not 100% homogeneous for different channel types SIP, ZAP, mISDN, IAX, etc. What this means for a simple Dial() from one channel to the other is that a) at times you get the Dial event first then the Newstate: Ringing event; and that b) with other/different orig/dest channel types you'll get the events in the reverse order... Nothing much but: i) you'll have to track them either way and ii) it reveals that the AMI events aren't 100% clean!!! :/ -- exvito On Feb 1, 2008 12:08 AM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: The snippet is asterisk telling you I'm just letting you know that the correct caller id for Channel: SIP/103-098500d8 is CallerID: 103 This is absolutely correct, it's just not a piece of information you expected to be receiving at that point. You probably also received a packet like that with the following: Channel: SIP/101- CallerID: 101 telling you, again, the caller id for only that channel. Moj ___ -- 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] Possible Conflicts with Junghanns 4 Port BRI and 8 Port Sangoma Analog in Same Box?
On Nov 26, 2007 4:06 PM, Steve Totaro [EMAIL PROTECTED] wrote: I wonder if anyone on the list has run a server with both types of cards installed? Results? Again, like Geert, not quite the same, but happily running quad BRI (beronet, HFC based) + TDM400. The BRI is running through mISDN 1.1.7 + chan_misdn, the TDM is, of course, running through zaptel 1.2.21 + chan_zap. So we're running two distinct stacks from asterisk to the HW. Not sure if you're having issues with said config or not via bristuffed zaptel... and wheather or not that will conflict with Sagoma's drivers -- with which I have no experience yet. However I can in our experience that the two distinct stacks operate very well. Feed us more info, if needed, -- exvito ___ --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] AMI Newstate Ringing events -- Inconsistent caller id ?
channel ? Oddly enough, the caller id is the destination caller id; or, in other words, the caller id info is the same as if it was the destination channel that was initiating the call. Example: If Zap/1 is Dial'ing() SIP/st2030 via extension 202, the Newstate event that AMI triggers contains 202 as the caller id instead of 200 -- (yes, the caller ids are correctly setup in the respective sip.conf / zapata.conf!) That's why you see a second extension (502) Dial'ing() the same SIP/st2030 channel; if Zap/1 Dial's() SIP/st2030 via the 502 extension, the Newstate Ringing event will contain 502 as the caller id !!! - The Problem - Wrap up - Of course I traced more information and noticed that in the failing circumstace -- Dial() to SIP channels -- there is something different happening the the overall sequence of AMI events: - If we Dial() a SIP channel, the Dial event is fired BEFORE the Newstate Ringing event. - If we Dial() a Zap channel, the Dial event is fired AFTER the Newstate Ringing event. - Of course, if we Queue(), there is no Dial event, there is a Join event fired BEFORE the Newstate Ringing events. Oddly enough (lucky enough?), everything seems fine regarding Newstate Ringing events triggered by the queue. So my main questions are: 1. Is the behaviour I'm observing supposed to be like this ? Or is this a bug ? 2. If the behaviour is correct, what explains the inconsistency ? (so that I can correctly account for it in programming the event watcher daemon) 3. How can I be certain that other channel types won't have a third different type of behaviour ? POSSIBLE APPROACH / WORKAROUND (which I wouldn't like to take until I understand what's going on here!) -- in the failing case, the Dial event precedes the Newstate Ringing event and contains correct info for both channels and caller id; it is not difficult to track and associate those two events so as to correct the information contained in the Newstate Ringing event. Well, this has been a long one! I hope I was clear enough in my explanation and I hope some one can shed some light into this. Thanks a lot in advance for any insights! :) Regards, PS: I can provide simplified traces of the events if needed. -- Ex Vito ___ --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] AMI Newstate Ringing events -- Inconsistent caller id ?
On Nov 23, 2007 6:58 PM, Moises Silva [EMAIL PROTECTED] wrote: I added the senddialevent, but not the condition you see below. That one was added by someone else. It seems that determine wheter or not the current extension will be set for outgoing calls. Setting OPT_ORIGINAL_CLID may fix your problem. That goes to explaining a few things - great. :-) I've checked UPGRADE.txt from asterisk-1.2.x source and it's clearly documented; I also checked the mailing list discussion you mentioned and other info. In short: it has been decided that the Newstate Ringing event (1) for the destination channel should have the clid associated to the exten for that dest channel by default... ok. The 'o' option for the Dial() application provides older 1.0 behaviour where such event will hold the caller id for the call originator. (1 - and probably others, but in this context, this is the relevant one) So it seems we're left with two paths: 1. Add the 'o' option to all Dials() and monitor exclusively Newstate Ringing events 2. Forget the 'o' option in Dials(), and monitor both Newstate Ringing and Dial events which contain all the needed info. I'm really not sure what is the best -- my initial testing for both 1. and 2. gives the same results, but following 1. seems to be less future-proof. Any opinions ? But then, there is at least one circunstace where there is no Dial event; that's when the Newstate Ringing events are triggered by the queues ! Fortunately, in these cases, the Newstate Ringing event contains the originator clid... Aargh !... I hate inconsistencies. :-/// Olle makes a joke about something that is very true. Dial() is monster. So you might have found a bug, since the 'o' option does not explain why the Dial event is generated before/after newstate events for some technologies and other inconsistencies you mentioned. Yep... Honestly, after having understood the reason why some of my Newstate Ringing events didn't contain the info I expected and how I can deal with that by resorting to track the Dial events, this is what worries me the most. -- Why on earth do I get Newstate Ringing events before the respective Dial events ?! Of course (luckily ?!), again, in these cases, the Newstate Ringing event contains the originator caller id information... since it is the snippet of code that Moises pasted to his message, in Dial() app, that does the trick and, apparently, it hasn't been run yet ! (go figure...) By the way. I just wrote an application that allow to execute AGI using the Manager interface, so that might be helpful for you. http://bugs.digium.com/view.php?id=11282 I'll have a look... Hope this post helps you a bit. It sure did ! Thanks a lot. -- exvito ___ --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: How to configure SIP domain on SPA942
On Nov 20, 2007 6:13 PM, Philip Prindeville [EMAIL PROTECTED] wrote: Yeah, I looked at LinksysSPATFTPProv.pdf... It doesn't say, however, how to get the phone's configuration out as a flat XML file. Only how to push the file back into the phone. wget http://ip-address-of-phone/admin/spacfg.xml ...gives me an XML file with my spa922 configuration. -- exvito ___ --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] Distributed FAX - How to best complement asterisk ?
Hello all, I'd like to thank everyone's input which I'll sumarize and comment on bellow. As in all complex solutions, there are no quick answers and no 100% correct solutions. There are trade-offs to be made among very different possiblities... Of course, the purpose of my original post was exactly get some feedback on what I initially designed and to widen my perspective on the particular subject by hearing different approaches to the problem. It's been great ! :-) For those interested, here is the summary: 1. from Mojo with Horan Company, LLC sheet-fed PDF scanners - desktop PDF - print to HylaFAX good: - nice idea, makes use of centralized HylaFAX server bad: - needs investment in replacing current equipment not sure if: - FAX users are PC savvy - there is a PC near every FAX 2. from Andreas van dem Helge suggests using a T.38 fax provider good: - would offload the gatewaying to a provider need to know: - whether T.38 is effectively solid under such scenario (see last comment, below) he also comments: - no success with callweaver T.38 gateway with some betas (answer to his question: the channel banks allow for the connection of analog FAX machines to the asterisk servers via PRI) - then says the topology I presented has too many PRIs: PSTN --PRI-- ast 1.2 --PRI-- AS5300 --SIP-- T.38 ATA he suggests something I don't quite understand (are these three parallel flows ? or does it represent one PRI going to a single AS5300 which would deliver the calls to T.38 ATAs or asterisk based on DDI ? what's the difference between the last two lines, can the AS5300 talk SIP/T.38 directly to an ATA without a SIP proxy ?): PSTN --PRI-- AS5300 --SIP-- ast 1.2 PSTN --PRI-- AS5300 --SIP-- ast 1.4 --SIP-- T.38 ATA PSTN --PRI-- AS5300 --SIP-- T.38 ATA 3. from Olivier shares information he got from Cantata where T.38 requires good levels of QoS my comment: I though T.38 was created to bypass those types of technical hurdles -- interesting ! (as I'll note below, Steve Underwood helps clarifying this notion) 4. from Phillip von Klitzing suggests that some bigger MFC printer/copy/fax combos can do FAX via SMTP good: - great, if it's over SMTP it'll work bad: - small offices won't justify such a big investment (I used to work for HP, I know how much those beasts can cost!) ;-) ...unless anyone's aware of a small FAX machine that can do SMTP ! (btw, there are some sheet-fed network scanners that can do SMTP -- see first comment) he also recalls an important issue: are you sure you want to rely 100% on IP only in your sattelite offices ? It might be wise to have 1 (analog?) line installed anway great point -- this has always been a possibility in the back of my mind... the only thing we'd loose in a setup where the remote office FAXes are directly attached to local analog lines is the ability to do integrated CDR processing for those FAX usages 5. from Benny Amorsen reminds that those big MFC boxes require the fax as email address for sending -- maybe too complex in day to day usage ? how tech savvy are the users ? another good point -- apart from their cost, in terms of usability, they might come short... or be too complex for someone with basic FAX machine abilities 6. from Steve Underwood reminds that T.37 (store and forward instead of realtime) is the answer to reliability... T.38 isn't all that robust, it just isn't as awful as FAX over VoIP he then concludes In a sane world all FAX would have been T.37 from a few months after the spec was released great info -- so, where is the T.37 compliant equipment ? (gateways, ATAs, FAX machines ?) Again, thanks a lot for the feedback (keep those posts coming!). Meanwhile I'll move on to further investigate some of the alternatives you proposed. Cheers, -- exvito ___ --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] Click to Talk Web Applications with Asterisk
On 10/9/07, Senad Jordanovic [EMAIL PROTECTED] wrote: zoachien wrote: Google for mexuar. Zoa Or look at one that works with MS Windows, Linux or Apple http://www.bicomsystems.com/products/C/P/319/382/ FYI, Mexuar's solution -- Corraleta SDK -- *works* with win, linux and mac, from direct experience. What's not so clear from the OP is what is meant by click-to-call: a) Automated dialing solutions via PSTN ? b) Call via a web embedded soft-phone ? (this would be Mexuar) -- exvito ___ --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] Distributed FAX - How to best complement asterisk ?
Hi list, I'm evaluating a private telephony scenario of about 20 locations - 300 phones, 50 FAX machines. Initial overview points to the installation of asterisk at three locations connected to the PSTN via ISDN PRI. All other locations, small by themselves, would get SIP phones managed by asterisk, since there is good IP connectivity between all sites. Now on to the subject... Handling FAXes: 1. On the locations where asterisk is installed, the solution is trivial; either by connecting FAXes to FXS ports on channelbanks or by managing faxes with iaxmodem + Hylafax. Probably a combination of both... 2. On the remaining locations we have a problem which I have been studying and trying to address... Faxing over IP. Side note: - I've read every recent mail on this mailing list regarding the subject - I've browsed the wiki to its fullest extent - I've googled a lot - I've read Steve Underwood's excelent summary on the subject (check it out at http://www.soft-switch.org/foip.html) Facts: a) FAX over VoIP will not work, so installing ATAs on the remote locations and bridging them with the PSTN FAXes is out of the plan. b) T.38 is the answer to FoIP c) asterisk 1.2 does not support T.38 d) asterisk 1.4 only does T.38 passthrough, not good enough e) CallWeaver seems to support T.38 gatewaying, although I'd rather move on with asterisk so as to leverage current experience and knowledge and to keep installed base with the same software. Possible solutions point to complementing asterisk installations with T.38 capable equipment. (of course, one other solution would be to subscribe to analog lines at each location! however, this would prevent us from performing FAX CDR accounting -- not a requirement, but a really nice-to-have). Having said all of this (and please correct me if I'm wrong) I'm looking for suggestions on how to best complement asterisk in such a scenario. The architecture I'm currently considering is: [PSTN] ---PRI--- [asterisk] ---PRI--- [PRI-to-T38 GW] ... ... --SIP/T.38--- [T.38 ATA] ---FXS--- FAX machine On incoming faxes, asterisk would simply Dial() the PRI leading to the PRI-to-T.38 GW which would be configured, according to the dialed number, to connect via SIP/T.38 to the respective T.38 ATA. Outbound would have the PRI-to-T.38 GW work the other way around, calling asterisk with the PSTN FAX destination number... Again, asterisk would only have to Dial() out to the PSTN. My questions: 1. What do you think of it, Is it feasible ? Does it make any sense ? How would you do it differently and why ? 2. I believe a Cisco AS53xx + Cisco ATAs would do the job. What about a Patton SmartNode 4960 + Patton ATAs ? (I have very little knowledge about Cisco equipment, but I'm almost 100% sure the Ciscos would do it... on the other hand, I've read most of the Patton docs and, again, I'm also almost 100% sure these would do it -- however, hands on experience and knowledge counts a lot!) 3. Roughly, how much would one expect to pay for one such PRI-to-T.38 gateway ? 5k, 10k, 20k ? Probably the Cisco version will be more expensive, no ? 4. Of course, I could use CallWeaver as a PRI-to-T.38 gateway... But then again, how solid would it be ? With which ATAs ? The CallWeaver website shows a very small amount of ATAs confirmed to be 100% working in T.38. 5. Would I need to have a SIP proxy between the PRI-to-T.38 gw and the T.38 ATAs or would they be able to talk to each other directly ? (I'd say this would depend on the specific equipment, but...) If that would be a requirement, which way would you go, asterisk 1.4 ? Would SER forward T.38 traffic ? Thanks for inputs and experiences in complementing asterisk with T.38 equipment. -- exvito ___ --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 cli - vi keybindings ?
On 9/24/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Sep 24, 2007 at 02:04:05AM +0100, Ex Vito wrote: Is there any way to setup the asterisk cli to use such keybindings ? ... Set in your environment: AST_EDITOR=vi before starting Asterisk. (See main/asterisk.c) Great ! Thanks a lot. :-) -- exvito ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] asterisk cli - vi keybindings ?
This might sound lika a small issu, but here it goes: I'm a long time unix user and my shell history usage and editing is configured to use vi keybindings; it's something that's already built into my fingers and using different bindings, like the arrow keys to fetch previous lines, really blows me !... :-( Is there any way to setup the asterisk cli to use such keybindings ? I took a quick glance at 1.4.11 source and found readline.[ch] files, but asterisk is not behaving to my inputrc configuration... Googled for a while to no effect. Am I alone in this ? Thanks in advance for any hint, -- exvito ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] not hearing dtmf tones
On 7/31/07, Jerry Geis [EMAIL PROTECTED] wrote: I am trying to re-create calling sendDTMF in an agi and not hearing the digit either. The above seems to re-create that without the AGI. ...you will have to configure your polycom / sip peer for inband DTMF if you want to hear the tones. -- exvito ___ --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] Suppress MusicOnHold in Queue
David L. West wrote: I want callers to go into the queue(s) and just hear ringing instead of MOH. Is this possible? ...use option 'r' for the Queue application. For more options, use 'show application queue' at the CLI. Cheers, -- exvito ___ --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] chan_isdn with HFC-compatible
Asterisk is loading the chan_misdn and lists mISDN when issueing show channeltypes - however it indicates Devicestate - No. when I look for misdn show stacks, it lists the single port of the ISDN-card, however indicates L2Link DOWN, L1LinkDOWN. so I guess theres something wrong, unfortunately I've no idea on what to check next...any pointers? maybe I'm missing something obvious, as this is my first installation of Asterisk + ISDN... Checklist: - The cable (after all L1 seems to be down) - If asterisk is to behave as a phone, check that the port is configured in TE mode - Verify PTP vs PTMP If needed: - Take a look at /var/log/asterisk/misdn.log - Increase verbosity in /etc/misdn-init.conf debug=3 should be a good starting point (reload mISDN modules after this change) -- exvito ___ --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
