[Asterisk-Users] chan_zap graceful failure
I was wondering if anyone is working on graceful failure for chan_zap? Let me explain the situation. We are using a T100P and TDM400P (4 FXS for fax). There was a major power outage and asterisk went down after the UPS (not a graceful shutdown -- my fault, no apcupsd running). As soon as power came back, the server started. However when it loaded wcfxs, port 3 on the card failed the tests (I assume from the module not being unloaded before power off). Because this one port failed the test, chan_zap failed to load and asterisk will not start. An unload and load of modules would not fix it. I had to unload and restart (a clean restart). While the unclean shutdown can be controlled in the future, I have had ports go bad and when they do asterisk will not start until the offending lines are removed from zapata.conf. This is not a very resilient solution (especially if you are not on site). I would much prefer for asterisk to keep running with what it has got. I will be looking into the code (and this might be fixed in cvs-head), but I would like to start a discussion on this first. Thanks, Jeb -- Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_zap graceful failure
Andrew Kohlsmith wrote:
It has nothing to do with not being unloaded; I've seen the wctdm driver fail
to detect modules correctly. Run it again and it works just fine. Some kind
of minor tweak is in order, I believe.
As an interim solution, your asterisk starup script should try to unload any
modules and reload them upon asterisk failure... preferably in a loop:
while(1) {
unload modules
sleep 1
load modules
start asterisk
sleep 5
}
I imagine at this point in time your startup script either does not loop, or
it doesn't try to unload/load the modules inside the loop.
While I like the idea (and will look into it -- might need a wait, etc),
as I said in original post, unloading and reloading did not fix the
problem. It took a clean shutdown (unload and restart) to fix the problem.
So regardless of why the card has failed, I would like to discuss making
chan_zap fail gracefully. For example if you have a
Dial(Zap/3/${NUMBER}, and Zap/3 does not exist, asterisk will spit a
warning (not fail to startup). However if you have that channel = 3 in
zapata.conf, chan_zap will fail and prevent asterisk from starting.
I would think that everyone would prefer asterisk to start and have
parts of the dialplan fail, rather than have asterisk not load at all.
As I said, I have not checked the behavior of cvs-head, I just wanted to
discuss making asterisk more resilient.
Thanks for the tip and I will look into it.
Jeb
--
Jeb Campbell
[EMAIL PROTECTED]
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Re: [Asterisk-Users] chan_zap graceful failure
Andrew Kohlsmith wrote: No; if the driver didn't load that's a major problem. Remember that if the channel doesn't exist all the subsequent channels move up... serious potential security issues. Good points. What if it kept the number (so nothing moved up), but marked the channel inuse (or even better a new non working status code). I'd rather have the system as it is, where it fails out with an error that is easy to understand so I can fix the problem. I agree that it should be a very loud error (and possibly repeated notifications on the console). But I also think that it should be able to limp along. What would you think of a commercial phone system that completely dies when one port dies? I appreciate your points (thats why I wanted to discuss this), maybe we could satisfy both with an option in zapata.conf (keepgoing=yes)? Thanks, Jeb -- Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] US$200 bounty for * paging feature
Henry Devito wrote:
I am already doing this with AGI, PERL, and PHP to set up the page
groups. I will release the code as open source if people are
interested. I'm not the best PERL scripter in the world but it works.
Attached is the agi I'm using. This is a modified script from a post on
voip-info. This works with our Cisco's that are setup like this: Line1
- XXX and Line2 -- XXX_i (for intercom).
The modifications from the stock script are paging to SIP/XXX_i (not
SIP/XXX), dynamic conferences based on original callerid, and playing
the beeps (Cisco just answers so this gives users a warning).
There is code to see if SIP/XXX is in use, and if so not to call
SIP/XXX_i, but the users wanted to see all pages so it is commented out.
Zones would be real easy with some arrays (as the conference is dynamic
based on the person calling) and the variables are there to check inuse,
etc.
extensions.conf:
[paging]
exten = *999,1,AGI(page.agi|${CALLERIDNUM})
exten = *999,2,Wait(1)
exten = *999,3,Playback(beep)
exten = *999,4,MeetMe(${CALLERIDNUM},dtqpA)
exten = *999,5,Hangup
[add-to-allcall]
exten = _X.,1,Playback(beep)
exten = _X.,2,MeetMe(${EXTEN},dmqpwx)
exten = _X.,3,Hangup
Really easy to modify. Have fun.
Again this could be cleaner, but I got that other script working and
haven't had the time or need to clean it up.
Jeb Campbell
[EMAIL PROTECTED]
#!/usr/bin/perl
#
#
# allcall.agi will add all your Polycom sip phones to a meet me
# conference for use in office wide paging
#
# It takes arguments in the form of SIP/ where is your
# sip extension. (can be any number of digits) The first argument
# is the originating caller and additional arguments are any other
# phone lines you wish to exclude
#
use strict;
use File::Copy;
# A Few Variables to Set and Initialize
#
#
my $outgoing = '/var/spool/asterisk/outgoing';
my $temp = '/var/tmp';
my $asterisk = '/usr/sbin/asterisk';
my $audio_out= 'console/dsp';
my @bypass = ();
my @meetme_calls = ();
my @rawsips = ();
my @sips = ();
my @intercoms= ();
my $callerid = Error;
# Parse out the Sip phones to exclude
#
# This truly shows my lack of understanding of perl
#
foreach (@ARGV) {
@bypass = split ( / /, $_ );
}
# This is our originating caller. I need his
# callerid so that others will know who the paging
# pest is:
#
$callerid = $bypass[0];
$callerid =~ s-SIP/--g;
# Setup an array with all the sip phones
#
# I think I could use the Asterisk::AGI here
# and also the incominglimit in sip.conf to accomplish
# this, but I'm not that good.
@rawsips = `$asterisk -rx sip show inuse`;
chomp(@rawsips);
shift (@rawsips);
shift (@rawsips);
@rawsips = sort (@rawsips);
#Jeb
# split to sips and intercoms
@sips = grep ( /^\d{3,4} / , @rawsips );
@intercoms = grep ( /^\d{3,4}_i / , @rawsips );
# Now check each sip phone to see if it's in use and also
# against our exclude list. If it passes both, it's
# added to our array of calls to make
foreach (@sips) {
my $sipphone = $1 if /(\d{3,4}) /;
my $sipinuse = substr( $_, 16, 1 );
unless ( grep ( /$sipphone/i, @bypass ) ) {
#if ( grep ( /${sipphone}_i/i , @intercoms ) and $sipinuse == 0 ) {
if ( grep ( /${sipphone}_i/i , @intercoms ) ) {
push ( @meetme_calls, make_call(SIP/${sipphone}_i) );
#push ( @meetme_calls, SIP/${sipphone}_i );
}
}
}
# The array is complete. The push line is uncommented
# if you want to add audio out to the intercom
#
#
# push ( @meetme_calls, make_call($audio_out) );
# Now move each call file to the outgoing directory
#
# Here's some more perl ugly
#
foreach my $call (@meetme_calls) {
move( $temp . '/' . $call, $outgoing . '/' . $call );
}
#print join(\n,@meetme_calls) . \n;
exit 0;
sub make_call {# makes the call file and returns the name
my $stripslash = $_[0];
$stripslash =~ s/\///g;
my $tempcall = $temp . '/' . $stripslash . $$;
my $callbase = $stripslash . $$;
open( call, $tempcall );
print call EOF;
Channel: $_[0]
MaxRetries: 1
Retry: 0
RetryTime: 60
Context: add-to-allcall
Extension: $callerid
Priority: 1
SetVar: ALERT_INFO=Ring Answer
CallerID: All-Call $callerid
EOF
close(call);
return $callbase;
}
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[Asterisk-Users] FRAME_CONTROL (5) dropping calls on PRI
After turning on full debug logs and getting users to report dropped calls, I have had 2 dropped calls in as many days. drop1: Apr 11 16:11:14 DEBUG[15029]: Got a FRAME_CONTROL (5) frame on channel Zap/3-1 drop2:Apr 13 09:13:37 DEBUG[5563]: Got a FRAME_CONTROL (5) frame on channel Zap/1-1 I can post full debug logs if anyone thinks that would help. Per yesterday's Line Noise thread, I have used zttest to check timing and used hdparm to set hda to udma2. zttest mostly reports 100% to 99.9875% with a few times dipping to 99.90%. I'm modifying zttest to log the time (like the log files) and to only report when under 99.98% (I will post a patch and bug when it is ready) so I can run it for a day without logs every second. Is anyone else seeing this and or have a solution (other than to correlate timing drops with this FRAME_CONTROL (5) )? I will post back when I have more info. Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip transfer and redirect in a Company setting
I have an asterisk box setup and dialplan that is something like this: (t1/pri) | [incoming] 1234,1,Dial(SIP/secretary,30,rt) 1234,2,Voicemail([EMAIL PROTECTED]) Now the t in the dial lets the sec transfer with # and if the person transferred to is unavail it goes to their voicemail -- that works great. However if the sec tells her phone to redirect to another phone (CFWDall on a 7960) asterisk will redirect that call to that phone. However it uses the sec's context to dial, which if redirecting internally included voicemail. So if the sec redirects to another phone and that phone does not answer, the redirected phone's voicemail plays and not the companies. I just wanted to see if anyone else had this problem (and a solution). Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avaya Partner ACS system, pre 7.0
Sean Kennedy wrote: I'll give you more details: We have two offices. Medium sized office ( 20 phones ), and a small office in a different city ( 5 phones ). We have plans to open a business office that will handle all incoming phone calls. This business office will be next door to the medium sized office. So the plan would be two fold: FXO ports for incoming, and FXS ports to the avaya system itself. Honestly, the way this setup is looking, it may be more problematic to band aid the situation as apposed to simply doing a full phone system upgrade. I'd prefer to do that, but I'm making sure there isn't a shortcut so we could reuse our old equipment and avoid the retraining of our staff. Thank you for the input. Sean, I have hooked up an Asterisk and Avaya system. I don't remember the Avaya specs, but I asked for help here, so it should be in the archives. I used asterisk as an IVR (data input application) off a T1 from the Avaya. Some recommendations: 1. Don't bother unless you can go T1/E1 into the Avaya. From the mix of incoming and outgoing you would need a channel bank anyway. And you have to figure what your time would be worth to debug and maintain on such a Frankenstein system. 2. Do you admin the Avaya yourself? If not, figure the costs of setting this up and making changes. 3. Do you have T1 cards and the latest Avaya software. We were going to extend our IVR to put voip phones off it, but would have had to pay to upgrade to the latest software to do any advanced routing (ie, DID) from asterisk to the avaya. In the end we rolled out Asterisk at the smaller location to debug it and will soon be replacing the Avaya at the main location. This was much cheaper than to upgrade the Avaya, pay to setup the Avaya to work with asterisk, debug the monster, then upgrade the main site to VOIP later anyway. Assuming you want to go to asterisk, I would roll out asterisk at the small location to debug, then upgrade the other locations and tie them in to the VOIP system. However if all you want is to make free calls between the locations, you get a 4 port TDM card (2 FXO, 2 FXS) at each location and hook into the avaya. You could even setup asterisk and the avaya to pass the digits back and forth for extensions (but unless you do it yourself, it will cost a bit to setup). Good luck, Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any Software Echo Cancellation in Asterisk?
Matthew Boehm wrote: For $2500 you can buy the TE411P which is a 4 port T1/PRI card that has onboard echo cancellation. Or you can send in your TE410P and $1000 and they will upgrade it for you. Do you know where we can read up on this? I couldn't find any info on digium's site. I'm very interested in this, and even a timeframe for availability would be great news. Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm400p and dell 2600 poweredge
Grant McInnes wrote: Hi all: I've been developing and testing on a tdm400p card and it's been going well. As you probably know, the tdm400p needs an ide power supply, but the dell poweredge 2600 that this card is destined for eventually has all the power supplied on the backplane with no ide cables. I used a molex splitter from another drive. On the topic of the dell 2600, I had to change servers as I could not get the t100p and tdm400p on their own irq's (It also had the raid card which is not supported in the new megaraid driver, but that is another story). Anyway in the bios, I could only set the irq to share with the raid controller or the gigabit ethernet. I was getting tons of irq misses which caused frame slips (and bad echo) on the t1. Anyway, please let me know how it goes and watch for irq misses. ** This was a while back, and it could have been a 2650, so please verify this for yourself ** Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm400p and dell 2600 poweredge
Dennis Webb wrote: I'm using tdm400's here and was curious about the irq misses. My zttool won't compile for some reason and I haven't researched it enough to be worried about it yet. Is zttool the way to diagnose the irq issue and does zttool work with the tdm400 boards. I don't have a shared irq issue but here's the output of /proc/interrupts. is there anything else to check? cat /proc/interrupts CPU0 CPU1 0: 172897707 0IO-APIC-edge timer 1: 11 0IO-APIC-edge i8042 2: 0 0 XT-PIC cascade 8: 2 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 11: 0 0 IO-APIC-level ohci_hcd 12: 50 0IO-APIC-edge i8042 14: 76 1IO-APIC-edge ide0 18: 172811936 0 IO-APIC-level wctdm 20: 172802850 0 IO-APIC-level wctdm 22: 172807318 0 IO-APIC-level wctdm 24: 172791427 0 IO-APIC-level wctdm 26: 361826 0 IO-APIC-level cciss0 28: 11479585 0 IO-APIC-level eth0 31: 293841 0 IO-APIC-level aic7xxx, aic7xxx NMI: 5767 0 LOC: 172894484 172894411 ERR: 0 MIS: 0 You appear to be fine as far as sharing irq's goes-- I must have had a 2650 as I used the megaraid driver (Perc). Sorry, to the list -- it was a while ago. The quickest way (I know) to check for irq misses is to simply cat /proc/zaptel/1 and do that for each number [1,2,etc] that is in there (you get a number for each card). If it doesn't say irq misses = number then you don't have a problem. Also I would not be worried (especially on non t1) to have a few misses since you loaded the modules (that's when it starts the count). When I was having problems, I was getting hundreds of irq misses a minute. For zttool to compile you must have newt installed (or libnewt libnewt-devel for rpm distros -- something like that anyway). Hope that helps. Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: AstLinux 0.2.2 released
Kristian Kielhofner wrote: Hello Everyone, I have released AstLinux 0.2.2. There are way too many improvements to list here, but here is a short summary: Linux 2.4.27, iptables, mini_httpd (with PHP SSL), phpconfig, AstShape traffic shaping, tftp server, OpenSSH, proftpd, Soekris Net4801 and Pentium-MMX and higher x86 support. There is actually WAY more software, but I couldn't possibly list it all. It is now available as a Windows install package (or 32mb Compact Flash image). AstLinux 0.2.2 occupies around 26mb of disk space once expanded to flash. The gzip'd CF images are about 15mb and the Windows installer isn't much more (for both images, the PDF user guide, and a copy of Putty). Wow -- very impressive. I have it downloaded but can't install it yet (at work now). I was working on the same thing and using a Gentoo build system for a 2.6 kernel and uclibc, but then went to glibc -- and now have a full system (perl, python, etc) in ~57M. Anyway I was just wandering if you had your build sources/scripts online so that people could customize Astlinux? I for one would like to be using the stable cvs. Thanks, Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo Problem
Brian M. Arlinghaus wrote: I've got mostly Cisco 7960s and a few Analog phones on TDM Ports. On the 7960s, the echo is quite bad. On the TDM ports, it is there, but not as bad. I have tried setting echo cancellation to various numbers, but have had no luck. This began after a HEAD version of * was installed. Since then, I installed what I think is the latest stable version (Asterisk CVS-v1-0-12/14/04-16:49:32) and the echo is still there. A support guy at Digium said it was a SIP problem. Just wanted to second this. I have about 20 7940's, 2 7960, and a 4 port FXS for fax machines going into a Bellsouth T1 (pri) and we get echo on some calls. I turned on echotraining (not for the faxes of course -- echocancelwhenbridged=no) and it will train out, but I thought that voip - pri could not have echo problems. Anyway, please keep me updated if you figure out a (real) solution to this. Jeb -- Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Dropped Calls - Audit, Restore, Idle state
We are using Asterisk on a T1 (pri) to Bellsouth. Calls are getting dropped calls every 60 minutes and we asked Bellsouth to debug the line. (Note, this was told to me and I have no idea what it means) They said that we are sending an Audit and then Restore (to Idle state) which causes the calls to be dropped. Has anyone else experiencing this? Any tips to verify or fix? (zaptel.conf) loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 (zapata.conf - relevant) overlapdial=yes switchtype=national signalling=pri_cpe context=incoming callerid=asreceived group=1 channel = 1-12 Thanks for any tips, Jeb -- Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI Dropped Calls - Audit, Restore, Idle state
Peter Svensson wrote: Asterisk tries to restart all idle line once an hour. This should only affect the channels that are idle but on our equipment (a Panasonic PBX) all channels are dropped. We use the following patch to basically disable the resets: Thanks for the tip and I imagine that will do the trick. If I get more time, I would like to investigate how asterisk is resetting idle channels (and getting active ones) and I agree that it would be nice to either be able to set a interval or a time in the configs. I would prefer a set time to let asterisk do its thing at 2 am. Come to think of it, I could use your patch to effectively disable the interval, and use a cron script to restart asterisk nightly. Thanks again, Jeb -- Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco SIP Firmware HERE!!!!
Matthew Boehm wrote: Thats not even the newest firmware, 7.2 is newest. Isn't this illegal? Matthew Definitely illegal, but 7.3 is the latest SIP firmware. Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] PoE switch question (Netgear FSM7326P works with Cisco)
Kevin P. Fleming wrote: Sean Kennedy wrote: http://www.cdw.com/shop/products/default.aspx?EDC=568864 Can anyone tell me if this switch will be able to supply a Cisco 7940 phone with power? I've heard of PoE issues with differing switches and the like, and I don't know how to check to see if this switch will be able to drive the phone in question. I would suspect it will not. Cisco phones do not implement standard PoE, they are proprietary. Actually the Netgear FSM7326P does work (we have 4 of them). And they work with regular PoE. Great price for layer 3, 24 ports of 10/100 PoE, and 2 gigabit. Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (Analog Intercom) PagePal by ATT -- was hooked to a Merlin
I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of an analog phone to page system, but of course I would like to reuse what is there. Thanks for any advice or pointers, Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco poe
By far the best poe (price/performance) I have seen for Cisco poe (or standard poe) is the Netgear FSM7326P. http://www.cdw.com/shop/products/default.aspx?EDC=568864 It is a managed layer3 poe switch (24 port) with 2 gigabit ports also. Works out of the box with Cisco and Snoms (it auto detects which polarity they want). No adapters needed for either. And it is about $1100. We are using 4 of them and love them. Jeb Campbell [EMAIL PROTECTED] Christopher L. Wade wrote: I know this is on the wiki, I just want to confirm so I don't blow up my cisco phones. I've got several cisco 7940's all running using cisco power cubes. However, my boss wants me to switch just a few over to poe, but doesn't want to fork out the dough for a nice cisco poe switch, or anybody else's poe switch for that matter. So my question is, what is the '99.999% sure/safe' poe injector solution that most people are using for the cisco phones? Right now I'm looking at buying the 3-Com 3CNJPSE (qty 2-3) to power the few specific locations where a power cube just wouldn't look right, like a conference room table for example. :) I know this solution, thanks to the fact that it is a 'hack', is far from the 99.999% I just stated, but it also seems to be the only low-end solution for poe. Am I right, or just plain blind? Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905/7912 SIP image location (on Cisco's site)
Just wanted to say Thanks to the Asterisk community -- all links are bookmarked now! Jeb Campbell [EMAIL PROTECTED] On Sep 21, 2004, at 4:54 PM, John Hill wrote: Her is the 7905-12 page http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Vallee Sent: Tuesday, September 21, 2004 4:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7905/7912 SIP image location (onCisco'ssite) Hi, I don't know if it the only way to do that, but it is how I did it. You need to have a valid cco account and your Smartnet contract have to be associated with your cco login. Next go to http://www.cisco.com/public/sw-center/ and when you are logged in, you can find the firmware of your choice under Voice Software section. If you have a smartnet account, don't hesitate to contact cisco technical support. Regards Christian Vallee -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jeb Campbell Envoyé : 21 septembre 2004 14:17 À : [EMAIL PROTECTED] Objet : [Asterisk-Users] Cisco 7905/7912 SIP image location (on Cisco'ssite) Hello all, I feel dumb asking this, but does anyone have a link to the SIP firmware for the 7912 on Cisco's site? I have a SmartNet contract, but I just can't find the link (you can search for 7960 sip firmware and find that fast). Thanks for the help, Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7905/7912 SIP image location (on Cisco's site)
Hello all, I feel dumb asking this, but does anyone have a link to the SIP firmware for the 7912 on Cisco's site? I have a SmartNet contract, but I just can't find the link (you can search for 7960 sip firmware and find that fast). Thanks for the help, Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax Detect problem (have consulted archives, wiki irc)
On May 2, 2004, at 1:02 PM, Ryan Courtnage wrote: I'm personally using it with spandsp and having no problems, but YMMV. If you want to enable it, goto line 60 of dsp.c and uncomment that #define so it looks like this: /* Define if you want the fax detector -- NOT RECOMMENDED IN -STABLE */ #define FAX_DETECT Then recompile and install. Jeb, thanks for taking the time to reply - i could have easily killed days looking for this! It's working fine now, but indeed, my mileage may vary ... which brings up the question; What is the reason FAX_DETECT is not recommended in Stable? No idea? The number of people using it seems pretty small (compared to all * use), so if they were having problems, if would make sense to change the default. But really I have no idea -- and it is working great for me! Jeb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P Integrated (DV) T1 - Public IP Range
On Apr 30, 2004, at 9:20 PM, mitchel wrote: The IP addresses assigned by my telco/isp to use for the serial (ppp) interface is 67.153.163.202 with a gateway of 67.153.163.201 and a netmask of 255.255.255.252. Thanks to Shido I have a nice firewall script which routes all the network traffic behind the asterisk server out the data channels and to the internet. If I go to www.network-tools.com it shows my ip address as 67.153.163.202. Going back in time to when I was using the Adtran router I was provided by my isp with the public ip address range of 67.153.163.210-67.153.163.214 with a gateway of 67.153.163.209 and a netmask of 255.255.255.248. With these settings I was supposedly able to use public real ip addresss on my own servers so that I could host miscellaneous things on it (mail, http, dns, etc). What I would like to do is map all or at least one of those ip addresses to an internal server. After doing a lot of reasearch and asking many questions I found out that to do what I need means that in theory no server anywhere will be assigned the ipaddress of 67.153.163.210 but only a simple routing rule will do the trick which is what I was trying to do with the following iptables rules. iptables -t nat -A PREROUTING -p tcp --dst 67.153.163.210 -j DNAT --to-destination 193.169.0.210 iptables -A FORWARD -d 193.169.0.210 -j ACCEPT Well the problem is that you don't have those addresses anymore. Here is your network (from ipcalc -- binary omitted): Address: 67.153.163.202 Netmask: 255.255.255.252 = 30 Wildcard: 0.0.0.3 = Network: 67.153.163.200/30 Broadcast: 67.153.163.203 HostMin: 67.153.163.201 HostMax: 67.153.163.202 Hosts/Net: 2 As you can see 200 and 203 cannot have hosts on them, so you have 201 and 202. But you are 202 and your isp is 201! In short, if you want more hosts get your isp to give you more addresses. Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI - GET DATA not working on current stable cvs (anyone else?)
Has anyone else had trouble with the AGI command GET DATA on the latest stable cvs? I can't get it to work with asterisk-perl, or by using print statements and reading stdin. I get 200 result= (timeout). (this is from the print statements, and asterisk-perl reports nothing). But asterisk is getting DTMF because my menu in extensions.conf works. I will go through the code Friday, but I just didn't know if anyone else was seeing this? Setup: 23 voice pri from Avaya PBX to Asterisk IVR. Thanks, Jeb Campbell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
On Apr 7, 2004, at 4:23 PM, Darren Nay wrote:
My question is. Is there a way to make asterisk aware of the
calling-from (callerID) number so that it will automatically detect
the number and then go directly to asking them to input their
password.
From show application VoicemailMain try:
exten = 1001,1,VoiceMailMain(${CALLERIDNUM})
We use:
exten = 1001,1,VoiceMailMain(s${CALLERIDNUM})
to skip password
Change extensions accordingly.
Jeb
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Re: [Asterisk-Users] Avaya - Asterisk IVR (working!)
To all those that have been following this thread (and helping me) -- it is working. The TN767 (refurb, but I just found out) was bad. Ah. Plugged into another card that went to a Shiva dialup box -- and boom, right in. So to all trying this, I know you can set this up in no time flat (as long as your phone person is fast at setting up the routes). Anyway, if you have questions, feel free to email me -- I want to pass on the knowledge that so many here have helped with. Jeb Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SoftFAX/spandsp cvs access
Steve, first thanks for the great work (especially the bugfixes). As development on SoftFAX/spandsp is especially fast and from the source it appears that you are using version control, it would be very nice for us users and testers to have read access to a repository. My cvs/subversion is internal and I understand not opening up systems, but maybe digium would host this if you cannot as this is a HUGE feature for asterisk. Thanks, Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P T1 problem (Avaya - asterisk IVR) (update)
This is an update on the status of an Avaya Definity w/ TN767E to Asterisk T100P (no crossover) If anyone has a working Avaya TN767 config print out I would really appreciate it. I have followed the config on the web, but cannot get it going and I really can't confirm the Avaya config (but I could show the phone guy that someone has it working). Anyway, the only stuff off list was trying to debug the connection. 1. With a crossover there is no sync (YELLOW and RED alarms) 2. With standard cable I get a pri error that they think they are the NET, but we are the NET. (This is asterisk 1.0 stable and the directions from voip-info) I'm almost to the point of getting them to send the TN767E back and getting a TN464, but I would like some confirmation either way (ie someone to swear they have done it -- and I will even pay someone to help if they have done it and can send a current config). So to all about to do this, I wish you better luck than I have had. I'm sure if you have more Avaya experience you will be fine; I just can't get in there and try things. I will get this working and post configs. Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P T1 problem (Avaya - asterisk IVR) (update)
On Mar 29, 2004, at 11:06 AM, Eric Wieling wrote: Jeb Campbell wrote: Anyway, the only stuff off list was trying to debug the connection. 1. With a crossover there is no sync (YELLOW and RED alarms) 2. With standard cable I get a pri error that they think they are the NET, but we are the NET. (This is asterisk 1.0 stable and the directions from voip-info) If they think they are NET then make Asterisk CPE, if they think they are CPE then make Asterisk NET. Tried it both ways (and changing the Avaya, then both ways again). No dice. Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftFAX/spandsp: installing and results on Gentoo
On Mar 23, 2004, at 8:27 AM, Stephen Davies wrote: My conclusion - spandsp seems to have a compatibility issue with tiff-3.6.1. (Unless my install approach was broken). And - you may want to make the two missing header files available for more than just 3.5.7 of libtiff. Running gentoo here also, and flawlessly. I think you are right about that involving libtiff. I'm running 3.5.7-r1, but I changed the ebuild. If you change the ebuild from make ... install to make ... install-private, you get the headers you need (it's in the function src_install). Anyway that's my 2 cents, Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Passing Argument to AGI
On Mar 23, 2004, at 5:32 PM, Unavailable ID wrote:
h,1,AGI(hangup.agi|string-argv-2348448)
I have been using this for perl:
exten = s,1,AGI(fax-incoming.agi|arg1arg2 with spacesarg3)
(thats a pipe and then the args with to separate them)
Then at the top of the perl script do:
@ARGV = split('',$ARGV[0]);
Now they are split!
Jeb Campbell
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Cell: 865-385-1437
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Re: [Asterisk-Users] T101P
On Mar 22, 2004, at 10:41 AM, Chris Tooley wrote: Can this be used as a RAS product to develop a PPP connection? Chris Should work -- more info at http://www.voip-info.org/tiki-index.php?page=Asterisk%20zapras http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20ZapRAS Note -- I have not tried these myself. Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Continue Macro after Hangup
Quick question -- How do I continue a macro after hangup (I need to run
a script)
I'm using RxFax(Spandsp) and it exits -1 (I even changed the code to
return 0, but no luck)
My macro ends with -- Hungup 'Zap/3-1'
Here is the Macro (the System(echo)'s are for debugging)
(Only the first System gets called)
[macro-fax-incoming]
exten = s,1,SetVar(FAXDIR=/var/spool/asterisk/fax/new)
exten = s,2,SetVar(FAXNAME=${TIMESTAMP}-${CALLERIDNUM})
exten = s,3,System(echo -e
${CALLERIDNUM}\n${CALLERIDNAME}\n${FAXNAME}
/var/spool/asterisk/fax/test)
exten = s,4,RxFax(${FAXDIR}/${FAXNAME}.tif)
exten = s,5,System(echo -e
${CALLERIDNUM}\n${CALLERIDNAME}\n${FAXNAME}
/var/spool/asterisk/fax/test2)
exten = s,6,System(/etc/asterisk/mail_fax.pl ${CALLERIDNUM}
${CALLERIDNAME} ${FAXNAME})
exten = s,105,System(echo -e
${CALLERIDNUM}\n${CALLERIDNAME}\n${FAXNAME}
/var/spool/asterisk/fax/test3)
exten = s,106,System(/etc/asterisk/mail_fax.pl ${CALLERIDNUM}
${CALLERIDNAME} ${FAXNAME})
exten = s,107,Hangup
And the result is only the first test getting executed.
Thanks,
Jeb Campbell
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Re: [Asterisk-Users] T100P T1 problem (Avaya - asterisk IVR)
James, Thanks so much for taking the taking the time to help me figure this out and learn something. It is a Definity, but I'm not sure about the card -- just a basic t1 card is all I know (on Monday I could get more info). Is there a command to find out which card is installed? or if that is enough to get started with the commands? Thanks again, Jeb On Mar 20, 2004, at 11:37 PM, James Coberly wrote: What Avaya card are you using? What model of system? Definity, Merlin, etc? With this I should be able to send you the base commands to review the card slot settings for the PXB James- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P T1 problem (Avaya - asterisk IVR)
Hello all, I'm having a problem with a T1 connection to a Avaya PBX (asterisk is an IVR). I could not get pri working and now I'm simply trying to get asterisk working with fxs_ks. Questions: 1. Is there anyway to troubleshoot or see what is being sent on the T1. zttool shows no errors, and the Avaya rings the t1 if wct1xxp is loaded, and gives busy if it is not loaded -- so I think something is being sent down the line, I just don't know what it is (asterisk -vc shows nothing) 2. Is there anyone with Avaya PBX - asterisk experience on the list? I'm remote, but I can login -- I just don't know what commands can troubleshoot the connection. Config: zaptel.conf (relevant section) span=1,1,0,esf,b8zs fxsks=1-24 zapata.conf [channels] context = demo switchtype = national signalling = fxs_ks group = 1 channel = 1-24 Thanks for any time and help Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P T1 problem (Avaya - asterisk IVR)
Yes, I followed those directions when I attempted to use pri, but I had no luck. (Note, I could not verify that those instructions were followed -- I have no experience with Avaya's). If you know the commands to verify those instructions on the Avaya side, I would appreciate the tips (Google was no help). If you are positive those directions are correct I will try again -- I ask because on irc #asterisk, some people thought those were wrong (like to use esf framing). Thanks for the response, Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 On Mar 20, 2004, at 8:30 PM, James Coberly wrote: Check out : http://www.voip-info.org/wiki-Asterisk+Avaya Depending upon the card you are using in the Avaya, you should set it up as a tie trunk on the Avaya side. James- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
