Re: [asterisk-users] way to inquire status of T1 link
with digium cards i would use: #sudo asterisk -r once inside cli : dialer1*CLI zap show status Description Alarms IRQbpviol C4 T2XXP (PCI) Card 0 Span 1 RED0 0 T2XXP (PCI) Card 0 Span 2 RED0 0 dialer1*CLI so RED is an ALARM GREEN is OK here this is a digium card woth 2 pri interfaces i have unpluged them so you see the RED alarm ___ -- 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] Digium HPEC license counting
Matt Watson wrote: Found key 'HPEC-KEY1' for 8 channels. Found key 'HPEC-KEY2' for 4 channels. Found valid HPEC licenses for 13 channels. Since when does 8+4 = 13 ??? maybe I should ask thinkgeek.com to make another t-shirt like this one: http://www.thinkgeek.com/tshirts/itdepartment/60f5/ ? This is actually caused by a minor bug in zaphpec_enable itself, it will be fixed in a future release. Thanks for reporting it :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] way to inquire status of T1 link
You can also cat /proc/zaptel/1 and parse the output of that. This doesn't tell you if asterisk is happy though. For that you could use the asterisk SNMP (sub)agent Tim. - Original Message - From: Jerry Geis [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, April 13, 2008 1:38:00 AM (GMT) Europe/London Subject: [asterisk-users] way to inquire status of T1 link Is there a way to inquire of the T1 link status? I mean having cron (as example) execute a program that asks if the T1 status is OK.YEL or RED? then on RED I can send some alert? Thanks Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Similar option as promiscredir to use in transfer (REFER)
I made a similar question in a previous thread, but there was no answer, so I think I was not very clear making the question. What I need is some configuration that works like promiscredir=yes in sip.conf that enables me to do the same thing with transfer (REFER), letting me transfer a sip call to a non local sip address. Thanks in advance, Thiago Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/ ___ -- 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] X100M never goes on-hook state
You need to replace the FXO module Was there ever any sort of protection installed for the module? Not what your carrier provides, that protects their equipment, but any secondary protection? These modules are easily damaged. I know of one connected to a very old SXS switch that damages from a small inductive kick when it opens ( goes on hook ) John Novack Marlon Dutra wrote: Hi guys, I've been experiencing a very strange issue with my Digium Card TDM400 as of this week. It has two FXS and two FXO. The FXO modules (both of them) never goes on-hook after hanging up in Asterisk. It had worked perfectly well for over four years. I put an ammeter in series with the line and the card, and immediately after plugging the connector to the card, I got 26mA in the circuit and a dial tone from the carrier, where it should be zero amper (on-hook state). If a Dial() something, it works perfectly. I can Hangup() the call, freeing the channel in Asterisk, but the hardware keeps off-hook forever, locking the line. If I Dial() again, Asterisk opens the line, sends the DTMFs normally, but it doesn't work since the carrier thinks I'm still holding the first call. It behaves exactly the same way with another analog line. If I plug either of the lines and my other Digium card (TDM2400), it works ok. The same with my Brazilian DigiVoice FXO card. Ok, you all might say: your card is damaged, throw it away. Ok, I could do it, but now comes the funny part: If I put an DSL filter in series with the line and the card, IT WORKS PERFECTLY!!! The filter imposes 25 ohms over the circuit. Maybe that's causing the card to work. When I put the filter and the ammeter in series, I get zero amper when on-hook and 26 mA when off-hook, that's the expected behaviour. I'm not an expert in electricity, so I really don't know why the card is behaving that way. What does that resistance make for the card to start working ok? I know the DSL filter isn't only a resistor. Maybe it has another electrical component that's helping more than the resistor. Just a guess. Tomorrow I'll buy a 30-ohm resistor, take the DSL filter off, and test the card only with the resistor, to check it out. In order to isolate the problem even more, I plugged the FXO port in one FXS port. Immediately after plugging it, Asterisk announced at the console that someone went off-hook at the FXS port. So, it's not really a carrier issue. The FXS port is perfectly -48V on-hook, and about 20 mA in the circuit when off-hook, closer than the carrier to the standard values. Any clue is welcome. -- Dog is my co-pilot ___ -- 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] compilation of asterisk 1.4.19 with ilbc already on system
I already have ilbc installed on my system. The files
are:
/usr/include/ilbc/iLBC_decode.h
/usr/include/ilbc/iLBC_define.h
/usr/include/ilbc/iLBC_encode.h
/usr/lib/libilbc.a
/usr/lib/libilbc.la
/usr/lib/libilbc.so - libilbc.so.0.0.0
/usr/lib/libilbc.so.0 - libilbc.so.0.0.0
/usr/lib/libilbc.so.0.0.0
However, if I do a make in asterisk-1.4.19, it will
not detect that libilbc.a is already on the system. If
I manually remove codec_ilbc from MENUSELECT_CODECS
in menuselect.makeopts then codecs/codec_ilbc.c will
compile to codec_ilbc.o because it finds the header
files (/usr/include/ilbc/iLBC_{en,de}code.h) but it
will fail to compile to codec_ilbc.so because it will
try to make the ilbc subdir which I haven't downloaded
the ilbc source code to.
How can I tell the make system in 1.4.19 that ilbc is
already on the system and that it should link to
/usr/lib/libilbc.a?
Shouldn't the configure script do that?
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Re: [asterisk-users] way to inquire status of T1 link
When I execute the commands in my cli pri show status zap show status I get errors for both commands. I am running 1.4.19, with libpri 1.4.3, and zaptel 1.4.10. how do I get these commands? Jerry -- help shows: help ! Execute a shell command abort halt Cancel a running halt ael debug contexts Enable AEL contexts debug (does nothing) ael debug macros Enable AEL macros debug (does nothing) ael debug read Enable AEL read debug (does nothing) ael debug tokens Enable AEL tokens debug (does nothing) ael nodebug Disable AEL debug messages ael reload Reload AEL configuration agent logoff Sets an agent offline agent show Show status of agents agent show online Show all online agents agi debug Enable AGI debugging agi debug off Disable AGI debugging agi dumphtml Dumps a list of agi commands in html format agi show List AGI commands or specific help cdr status Display the CDR status console answer Answer an incoming console call console autoanswer Sets/displays autoanswer console dial Dial an extension on the console console hangup Hangup a call on the console console send text Send text to the remote device core clear profile Clear profiling info core set debug channel Enable/disable debugging on a channel core set debug Set level of debug chattiness core set debug off Turns off debug chattiness core set global Set global dialplan variable core set verbose Set level of verboseness core show applications Shows registered dialplan applications core show application Describe a specific dialplan application core show audio codecs Displays a list of audio codecs core show channels Display information on channels core show channel Display information on a specific channel core show channeltypes List available channel types core show channeltype Give more details on that channel type core show codecs Displays a list of codecs core show codec Shows a specific codec core show config mappings Display config mappings (file names to config engines) core show file formats Displays file formats core show file version List versions of files used to build Asterisk core show functions Shows registered dialplan functions core show function Describe a specific dialplan function core show globals Show global dialplan variables core show hints Show dialplan hints core show image codecs Displays a list of image codecs core show image formats Displays image formats core show license Show the license(s) for this copy of Asterisk core show profile Display profiling info core show switches Show alternative switches core show threads Show running threads core show translation Display translation matrix core show uptime Show uptime information core show version Display version info core show video codecs Displays a list of video codecs core show warranty Show the warranty (if any) for this copy of Asterisk database del Removes database key/value database deltree Removes database keytree/values database get Gets database value database put Adds/updates database value database show Shows database contents database showkey Shows database contents dialplan add extension Add new extension into context dialplan add ignorepat Add new ignore pattern dialplan add include Include context in other context dialplan reload Reload extensions and *only* extensions dialplan remove extension Remove a specified extension dialplan remove ignorepat Remove ignore pattern from context dialplan remove include Remove a specified include from context dialplan save Save dialplan dialplan show Show dialplan dnsmgr reload Reloads the DNS manager configuration dnsmgr status Display the DNS manager status dundi debug Enable DUNDi debugging dundi flush Flush DUNDi cache dundi lookup Lookup a number in DUNDi dundi no debug Disable DUNDi debugging dundi no store history Disable DUNDi historic records dundi precache Precache a number in DUNDi dundi query Query a DUNDi EID dundi show entityid Display Global Entity ID dundi show mappings Show DUNDi mappings dundi show peers Show defined DUNDi peers dundi show peer Show info on a specific DUNDi peer dundi show precache Show DUNDi precache dundi show requests Show DUNDi requests dundi show trans Show active DUNDi transactions dundi store history Enable DUNDi historic records feature show Lists configured features file convert
Re: [asterisk-users] way to inquire status of T1 link
On Sat, Apr 12, 2008 at 08:38:00PM -0400, Jerry Geis wrote: Is there a way to inquire of the T1 link status? I mean having cron (as example) execute a program that asks if the T1 status is OK.YEL or RED? then on RED I can send some alert? Those are lower-layer alarms. They will also appear on zttool and 'cat /proc/zaptel/*' -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] way to inquire status of T1 link
My guess is that you don't have any spans set up, or Asterisk doesn't have zaptel support... Is chan_zap.so loaded? -Jon - Original Message - From: Jerry Geis [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, April 13, 2008 1:27:56 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] way to inquire status of T1 link When I execute the commands in my cli pri show status zap show status I get errors for both commands. I am running 1.4.19, with libpri 1.4.3, and zaptel 1.4.10. how do I get these commands? Jerry -- help shows: help ! Execute a shell command abort halt Cancel a running halt ael debug contexts Enable AEL contexts debug (does nothing) ael debug macros Enable AEL macros debug (does nothing) ael debug read Enable AEL read debug (does nothing) ael debug tokens Enable AEL tokens debug (does nothing) ael nodebug Disable AEL debug messages ael reload Reload AEL configuration agent logoff Sets an agent offline agent show Show status of agents agent show online Show all online agents agi debug Enable AGI debugging agi debug off Disable AGI debugging agi dumphtml Dumps a list of agi commands in html format agi show List AGI commands or specific help cdr status Display the CDR status console answer Answer an incoming console call console autoanswer Sets/displays autoanswer console dial Dial an extension on the console console hangup Hangup a call on the console console send text Send text to the remote device core clear profile Clear profiling info core set debug channel Enable/disable debugging on a channel core set debug Set level of debug chattiness core set debug off Turns off debug chattiness core set global Set global dialplan variable core set verbose Set level of verboseness core show applications Shows registered dialplan applications core show application Describe a specific dialplan application core show audio codecs Displays a list of audio codecs core show channels Display information on channels core show channel Display information on a specific channel core show channeltypes List available channel types core show channeltype Give more details on that channel type core show codecs Displays a list of codecs core show codec Shows a specific codec core show config mappings Display config mappings (file names to config engines) core show file formats Displays file formats core show file version List versions of files used to build Asterisk core show functions Shows registered dialplan functions core show function Describe a specific dialplan function core show globals Show global dialplan variables core show hints Show dialplan hints core show image codecs Displays a list of image codecs core show image formats Displays image formats core show license Show the license(s) for this copy of Asterisk core show profile Display profiling info core show switches Show alternative switches core show threads Show running threads core show translation Display translation matrix core show uptime Show uptime information core show version Display version info core show video codecs Displays a list of video codecs core show warranty Show the warranty (if any) for this copy of Asterisk database del Removes database key/value database deltree Removes database keytree/values database get Gets database value database put Adds/updates database value database show Shows database contents database showkey Shows database contents dialplan add extension Add new extension into context dialplan add ignorepat Add new ignore pattern dialplan add include Include context in other context dialplan reload Reload extensions and * only * extensions dialplan remove extension Remove a specified extension dialplan remove ignorepat Remove ignore pattern from context dialplan remove include Remove a specified include from context dialplan save Save dialplan dialplan show Show dialplan dnsmgr reload Reloads the DNS manager configuration dnsmgr status Display the DNS manager status dundi debug Enable DUNDi debugging dundi flush Flush DUNDi cache dundi lookup Lookup a number in DUNDi dundi no debug Disable DUNDi debugging dundi no store history Disable DUNDi historic records dundi precache Precache a number in DUNDi dundi query Query a DUNDi EID dundi show entityid Display Global Entity ID dundi show mappings Show DUNDi mappings dundi show peers Show defined DUNDi peers dundi show peer Show info on a specific DUNDi peer dundi show precache Show DUNDi precache dundi show requests Show DUNDi requests dundi show trans Show active DUNDi transactions dundi store history Enable DUNDi historic records feature show Lists configured features file convert Convert audio file group show channels Display active channels with group(s) help Display help list, or specific help on a command http show status Display HTTP server status iax2 provision Provision an IAX device iax2 prune
Re: [asterisk-users] way to inquire status of T1 link
This is great stuff, thanks.
Steve Totaro
On Sat, Apr 12, 2008 at 9:27 PM, Jonathan C. Bailey
[EMAIL PROTECTED] wrote:
We use Nagios for network monitoring. We've got a check_pri script that
should be fairly universal. It will return critical for any alarm. Feel
free to use the script as you see fit. YMMV - may skin cats, etc (you know
the disclaimer drill)...
#! /usr/bin/python
# Checks PRI status - returns similar to the following:
# PRI span 1/0: Provisioned, Up, Active / PRI span 2/0: Provisioned, Up,
Active
import os, sys, socket
statusstring = ''
for file in os.popen('/usr/sbin/asterisk -rx pri show spans').readlines():
out = file[:-1]
if out.startswith('PRI'):
statusstring += ' / ' + out.strip()
if out.startswith('Unable to connect to remote asterisk'):
print Unable to connect to Asterisk instance
sys.exit(2)
print statusstring.strip()[2:]
if statusstring.strip()[2:].count(In Alarm) 0:
sys.exit(2)
# Nagios Return Codes
# OK = 0
# Warning = 1
# Critical = 2
# Unknown = 3
sys.exit(0)
-Jon
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 12, 2008 8:21:09 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-users] way to inquire status of T1 link
Jerry Geis wrote:
Is there a way to inquire of the T1 link status?
I mean having cron (as example) execute a program that asks if the T1
status is OK.YEL or RED?
then on RED I can send some alert?
What sort of adaptor?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Re: [asterisk-users] Is Asterisk really good??
On Sat, Apr 12, 2008 at 01:40:44PM -0700, Steve Edwards wrote: On Sat, 12 Apr 2008, Jay R. Ashworth wrote: On Fri, Apr 11, 2008 at 06:11:45PM -0700, Eugen Soare wrote: That was cool! thanks for the pdf. I'm in the midst of rearranging things (which are 2 to 3 times as large as they were then); I'll update that once I'm done. Double-plus cool. I'd be interested in sections like Rolling out a new server or How we maintain all the little configuration files without losing our sanity. I smell a magazine article. :-) The answer to the second question is likely going to become rsync or cfengine, but I haven't gotten that far yet... and we don't change them all that much anyway. VICIdial has *lots* of knobs. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] way to inquire status of T1 link
On Sat, Apr 12, 2008 at 08:38:00PM -0400, Jerry Geis wrote: Is there a way to inquire of the T1 link status? I mean having cron (as example) execute a program that asks if the T1 status is OK.YEL or RED? then on RED I can send some alert? Since Sangomas were original designed as unchannelized data cards, ifconfig exposes the alarm state; alarmed spans don't show RUNNING, while green ones do. I don't think you can distinguish between YEL and RED, but you probably don't need to. I gather you can also SNMP the Sangoma drivers. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Is Asterisk really good??
On Sun, 13 Apr 2008, Jay R. Ashworth wrote: On Sat, Apr 12, 2008 at 01:40:44PM -0700, Steve Edwards wrote: On Sat, 12 Apr 2008, Jay R. Ashworth wrote: On Fri, Apr 11, 2008 at 06:11:45PM -0700, Eugen Soare wrote: That was cool! thanks for the pdf. I'm in the midst of rearranging things (which are 2 to 3 times as large as they were then); I'll update that once I'm done. Double-plus cool. I'd be interested in sections like Rolling out a new server or How we maintain all the little configuration files without losing our sanity. I smell a magazine article. :-) That works, but I'm impatient. I'm up for peer review before publication. The answer to the second question is likely going to become rsync or cfengine, but I haven't gotten that far yet... and we don't change them all that much anyway. VICIdial has *lots* of knobs. I'm mainly interested in consistency in configuration. The method has to be sophisticated enough to handle this box has 2 Ethernet interfaces so I should configure OpenSER and Asterisk to listen to both IP addresses on ports 5060 and 5061 respectively. This would preclude rsync. I currently do it with shell scripts but I'm looking for something a bit more sophisticated. Puppet (http://reductivelabs.com/trac/puppet/wiki/AboutPuppet) was suggested during the Friday morning VOIP Users Conference. It's open source and written in Ruby. I just feel a bit silly installing yet another language just to support a support tool. The shell script approach has the advantage of light weight. I do a minimal Centos 5 install and wget a single script which does everything -- configures the network, installs packages (OpenSER, Asterisk, Zaptel, Libpri, MySQL), adds users, and configures everything from services to timezone. I may stick with it, but it's getting a bit combersome and am interested in what has worked for others. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Is Asterisk really good??
Steve Edwards [EMAIL PROTECTED] wrote: On Sun, 13 Apr 2008, Jay R. Ashworth wrote: On Sat, Apr 12, 2008 at 01:40:44PM -0700, Steve Edwards wrote: I'd be interested in sections like Rolling out a new server or How we maintain all the little configuration files without losing our sanity. I smell a magazine article. :-) That works, but I'm impatient. I'm up for peer review before publication. The answer to the second question is likely going to become rsync or cfengine, but I haven't gotten that far yet... and we don't change them all that much anyway. VICIdial has *lots* of knobs. I'm mainly interested in consistency in configuration. The method has to be sophisticated enough to handle this box has 2 Ethernet interfaces so I should configure OpenSER and Asterisk to listen to both IP addresses on ports 5060 and 5061 respectively. This would preclude rsync. Why do you think that that would preclude rsync? -- /\ Bernd Felsche - Innovative Reckoning, Perth, Western Australia \ / ASCII ribbon campaign | Great minds discuss ideas; X against HTML mail | Average minds discuss events; / \ and postings | Small minds discuss people. -- Eleanor Roosevelt ___ -- 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] Is Asterisk really good??
On Mon, 14 Apr 2008, Bernd Felsche wrote:
Steve Edwards [EMAIL PROTECTED] wrote:
I'm mainly interested in consistency in configuration. The method has
to be sophisticated enough to handle this box has 2 Ethernet interfaces
so I should configure OpenSER and Asterisk to listen to both IP addresses
on ports 5060 and 5061 respectively. This would preclude rsync.
Why do you think that that would preclude rsync?
Well, it may be based on my ignorance :)
Can rsync mung a stanza from iax.conf like:
[general]
disallow = all
allow = ulaw
mailboxdetail = no
notransfer = yes
port= 5036
register= ${HOSTNAME}:[EMAIL PROTECTED]
trunk = no
and insert the appropriate values?
Can rsync create /etc/sysconfig/openser like:
# Created by ./host-setup.sh on 2008-04-12 17:55:03
OPTIONS=
OPTIONS=$OPTIONS -l a.b.c.d:5060
OPTIONS=$OPTIONS -l a.b.c.e:5060
# (end of /etc/sysconfig/openser)
where a.b.c.d and a.b.c.e are the IP address of eth0 and eth1? (And the
3rd line would only be created if there are 2 interfaces.)
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] callers in queue passed to agents who accept only one call at a time
On Fri, 28 Mar 2008 06:33:42 -0700 (PDT), Vieri wrote: However, I can't use ringinuse=no in queues.conf because I'm running 1.2.27 (or is there a backport/patch?). iirc, there is a patch to backport ringinuse to 1.2.x. it's on mantis somewhere. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- 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 and the Mitel SX 200 integration
At 21:08 4/11/2008, Alexander Lopez wrote: Jorge is correct you will not get the information need via FXO/FXS unless you program the Mitel to do DTMF inband. It is possible but a cludge of a fix at best. We have successfully integrated several Mitel SX200 and SX2000 switches via the PRI (preferred) or T1 using EM_Wink (works but you have delays while waiting for the winks. (wink, wink :-) ). The Mitel is rock-solid Until the floppy disk dies. Then you have a huge doorstop and no phone system. The floppy drives aren't easily replaceable. We and the customer are anxiously awaiting the day we can sell that SX-200 and the ActiveVoice system to some poor unsuspecting souls. An Asterisk box can be made out of high-quality generic parts that are easily available. and depending on the size of the install a fork-lift replacement may not be desirable. I would start by replacing the VM (ActiveVoice) with and Asterisk box, you can give them unified messaging as well as a stable and current platform ( I have seen the Octel COV card catch on fire!!) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jorge Mendoza Sent: Friday, April 11, 2008 8:32 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Doug Subject: Re: [asterisk-users] Asterisk and the Mitel SX 200 integration I second Doug advice. Migrate to Asterisk asap. We have several Asterisk auto attendant integrated with Mitel, even billing using the Mitel's smdr. But voicemail is different. The COV card emulate a SS4 phone and receive information needed for a voice mail system. With FXO/FXS ports is not possible receive such information. Jorge Mendoza John covici wrote: Yep, you guessed it, an activvoice system. Anyway to make Asterisk act like that for a while? Thanks. on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote At 17:32 4/11/2008, John covici wrote: Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! Is it an ActiveVoice system? We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? We've got a client with the exact same setup. They have suffered long enough with this dinosaur. They are in the process of going with an all-Asterisk system. You would probably make more money trying an intermediate step using the SX-200 and Asterisk, but it would be obviously more costly for them as well as prolong their misery. It's your call, but I would recommend getting away from 30 year old technology as fast as you can run. The ActiveVoice system is a cantankerous 20 year old system in itself. You have just received 2 cents worth of advice for FREE! If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Similar option as promiscredir to use in transfer (REFER)
13 apr 2008 kl. 17.46 skrev [EMAIL PROTECTED]: I made a similar question in a previous thread, but there was no answer, so I think I was not very clear making the question. What I need is some configuration that works like promiscredir=yes in sip.conf that enables me to do the same thing with transfer (REFER), letting me transfer a sip call to a non local sip address. I'm still not really sure what you ask for, but I'll give it a try. The transfer() dialplan application supports generating a REFER from Asterisk to the client. If the call is not answered, it will send 302, if the call is in UP state (answered), Asterisk will send a REFER. Try it. Best regards, /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk SIP Masterclass, Orlando, Florida Next week * A few seats left - register today! ___ -- 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
