Re: [asterisk-users] PoE module
Here's a cheap solution for PoE piggybacked over your existing network. http://www.amazon.com/gp/product/B0002R6X9S On Jul 14, 2013, at 3:12 PM, bilal ghayyad wrote: > Hello; > > We have a cisco switches but they are not PoE and we need only to have PoE > device so the cables come for it first to provide the power and then goes to > the switch (to be like batch panel), is there something like this that can be > used for the IP Phones? > > Regards > Bilal > > -- > This message has been scanned for viruses and > dangerous content and is believed to be clean. > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
On Oct 30, 2010, at 2:28 PM, Zeeshan Zakaria wrote: > My main asterisk server is under unusual heavy attack, and so far > Fail2Ban has blocked about 30 IPs, from various different countries. > At this time it is blocking about 1 IP address every few minutes. > > Just wondering if anybody else is also experiencing unusually > increased hack attempts today? > > Zeeshan A Zakaria > It's been an extremely busy day for the exploiters. I moved my phone system from one circuit that I have (10Mb) to another that is behind a firewall (100Mb) and the fail2ban alerts are all gone. I'm not really concerned that someone will determine the passwords, as I use the phones serial numbers to determine that. But still, very irritating to see so many attempts at exploiting my phone system. fail2ban is nice, but I recommend you put your system behind a firewall and only allow necessary connections. pfsense is doing the trick for me. - Niles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Firmware
On Jul 21, 2010, at 7:05 PM, Apu Islam wrote: > Can any good men on this group share me the firmware of a Cisco 7960 Phone? > Currently this one has Call Manager Firmware installed, I am trying to > convert it into SIP. > Much appreciated. > > > Apu Try google keywords: "index of" P0S3-06-3-00.bin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Ubuntu
On Jun 4, 2010, at 8:40 AM, Danny Dias wrote: > Hello Asterisk users, > > I'm having a little problem with an Asterisk installation on Ubuntu, i had > installed many asterisks on CentOS but never in Ubuntu, the problem is that > Asterisk and DAHDI does not start at system start...i have to make > "/etc/init.d/asterisk start" and "/etc/init.d/dahdi start" manually every > time i reboot the machine (my laptop for testing) > > So, what should i do in order to solve this situation? Did you run update-rc.d on your asterisk/dahdi init.d scripts? Do man update-rc.d Niles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] migrate from zaptel to dahdi
On Jul 6, 2009, at 10:00 AM, Jerry Geis wrote: > Over the weekend I tried to migrate a system from > asterisk 1.4.25, libpri 1.4.1 ZAPTEL 1.4.12.1 > > to > asterisk 1.4.25, libpri 1.4.7 (not 1.4.1) and DAHDI 2.2.0 > > I removed all old zaptel by: >mv /etc/zaptel.conf /tmp >mv /etc/asterisk/zapata.conf /tmp > >rm /etc/init.d/zaptel >rm /etc/sysconfig/zaptel >rm /etc/modprobe.d/zaptel 2> /dev/null > /dev/null >rm /etc/udev/rules.d/zaptel.rules >rm /etc/rc.d/rc*/*zaptel >rm /sbin/zt* >rm -rf /usr/share/zaptel >rm -rf /usr/include/zaptel > > Then I just did a CLEAN install of dahdi, libpri and asterisk again. > > After upgrading incoming calls seemed to work just fine. > Outgoing calls gave me an error 99 > > > I have a TE205P installed. > > I did change the extensions.conf to use DAHDI and not Zap. > > I had to quickly change back as it is a production system. > > Any thoughts on what might have happened here? > I didnt know if have two libpri versions confused things or what? > > ANy thoughts for the next time I try are appreciated. > Jerry, Check the dahdichanname setting in asterisk.conf. I had the same issue myself - Niles ___ -- 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 on Mac OS X
On Apr 23, 2009, at 3:14 PM, Rick Dwyer wrote: > Hello list. > I posted this over on the Biz section but some of the members thought > I might find more people running Asterisk on the Mac over here. > > Here's my question: > > > I have looked at PHLink and PhoneValet and neither seem to be able to > do what I need, so I am looking at Asterisk. > > What I want to do is allow callers to call a our phone line and > unsubscribe their phone number from our call center list. So, > basically, when they call in, they would be greeted with a message > something like: "please enter your 10 digit phone number followed by > the pound sign". They would then have the number read back to them to > confirm it or reenter it. Once confirmed, it would write the phone > number to a text file for importing into MySQL or FileMaker. > > Is what I am trying to accomplish within the realm of what Asterisk > can do on the Mac platform... or any platform... and if so, how > difficult of an install is it? I have read varying accounts from it > being a breeze to being frustrating. The main distinction between running Asterisk on Linux as opposed to OSX, is that you'll have access to hardware device drivers. If you're going to be using a SIP/IAX Trunk, then you'll be just fine on OSX. What your attempting to do falls closer to the category of "breeze". You can install the asterisk-addon package to handle your SQL queries from within the dialplan, or you can use AGI to have a perl or php script do that work. Niles ___ -- 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] Fax solution for Asterisk
On May 21, 2008, at 9:55 AM, Sanjay Rajdev wrote: We are using Asterisk 1.4.13 on FC6 and have a T1 card with 20 DID's. We are wanting to use one of the DID's for Fax, is this possible or do we have to add some addition Hardware and what is the best way to do this. I know that similar thing would have been asked multiple time already, but I was not able to find anything that could answer my questions. Regards, Sanjay Rajdev I have 3 running installations of Asterisk using IAXmodem and Hylafax. Very very reliable, no additional hardware required. http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem http://www.voip-info.org/wiki/view/Hylafax Niles ___ -- 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] voice mail indicator on phone
Jerry, I'd imagine that you can achieve this through SIP Event Notify, via AGI using sipsak (www.sipsak.org) I'm doing a similar thing with Cisco phones, and it works great. Here's an example of what I pass to the phones. NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 From: ;tag=2427962554 To: Call-ID: [EMAIL PROTECTED] CSeq: 101 NOTIFY Contact: User-Agent: sipsak voicebox Event: simple-message-summary Content-Type: application/simple-message-summary Content-Length: 22 Niles On May 7, 2008, at 8:57 AM, Jerry Geis wrote: > Is there a method from the dialplan that I > can turn on a voicemail indicator on a polycom phone. Like a blinking > light or something. > > Then I would also need to turn it off. > > Is there a way to do that? > > Jerry > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting callerid across servers
On Mar 11, 2008, at 3:25 PM, Jerry Geis wrote: > I have a situation when a T1/PRI line comes into box 1 > then uses SIP over to box 2 and all my phones are on box 2. > if the person is not at their desk on ring no answer I am calling > their > cell phone > which places the call back over SIP to box 1 and out the T1 . > > How can I setup this configuration so the original caller ID will > show up > on the cell phone. > > Thanks, > > Jerry > > Jerry, What CID are you expecting to show on the cell phone? Based on what information you have provided, the original call is coming outside of your system, and you will not be able to duplicate their CID when you pass the call to your users cell phone. You can always screen the call though, allowing the recipient to know who is calling them. Niles ___ -- 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] Read function
On Mar 9, 2008, at 1:34 AM, Daniel Suleyman wrote:
> Dear all, interesting behaivior of the Read function.
>
> I have SIP phone(XLITE) attached to my Asterisk.
>
> SIP.conf
> [7007]
> type=friend
> qualify=900
> host=192.168.85.27
> dtmfmode=rfc2833
> disallow=all
> allow=gsm
> allow=alaw
> allow=ulaw
>
> extensions.conf
>
> 1,1,Answer;
> 1,2,Read(CNT,,2)
> 1,3,SayNaumber(${CNT})
>
> Function read do not write anything to CNT or write "".
>
> in SayNumber it is always equel to ""; even if I previously defins
> CNT = 123;
>
> And read function not exit if I pres #.(I think it is exit only on
> timeout)
>
> Strange can anybody point on mistake?
>
You have a spelling error at extension 1, priority 3.
SayNaumber, as opposed to SayNumber
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Re: [Asterisk-Users] upgrade to 1.2 beta 2 issue
On Nov 9, 2005, at 2:32 PM, Kevin P. Fleming wrote: Paul Dugas wrote: I run CVS a the house and have been getting these for quite some time now. I have an old-model IAXy that has been misbehaving in this manner for months. I've become desensitized ;) It may be a bug in chan_iax2 or the IAXy... I forgot to check yesterday, will do so now. I'm also using about 16 of the IAXy devices on this box, so I'm getting quite a few of these messages. Niles ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] voicemail issue with beta1 & beta2
I'm having a voicemail problem that I can't seem to eradicate, and I'm hoping to get some direction from this list. The phone system in question is using 1.2 Beta 2, using Realtime with about 5k users in the database. There have been a few messages that appeared to be blank, but when they forwarded to another extension with a prepended message, the original message would appear. Apparently some voicemail messages have an empty msg,txt file, which the voicemail system recognizes as a voicemail, but the corresponding msg.wav file won't play. When the user forwards what they believe is a blank message to my mailbox, the msg.txt file is rewritten, and I receive the prepended message followed by the original message. I didn't see this kind of issue in the archives, or in google. Please advise Thanks! Niles ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] upgrade to 1.2 beta 2 issue
Ever since I upgraded to beta2, the console is littered with these kind of messages: NOTICE[206]: chan_iax2.c:5654 update_registry: Restricting registration for peer 'kkai13' to 60 seconds (requested 0) Any way to suppress this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Realtime & Directory sorting issue
Hello, I'm using Asterisk 1.2.0-beta1 with Realtime, and I'm not receiving the users in alphabetical order. It appears that they are in numerical order based on the assigned mailbox, and reversing the sort by firstname, lastname option flag simply reverses the numerical order. I've not found any other inquiries about this issue, and I've tested it on two systems. (other, CVS-HEAD) Any help would be greatly appriciated. Niles ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Meetme issue
I've noticed that when I use the MeetMe app, it shows the Zap/pseudo
context
in a fax context. Anyone know what could cause this??
from my extensions.conf
exten => conf,1,Answer
exten => conf,2,SetMusicOnHold(rachmaninov)
exten => conf,3,Macro(isadmin)
exten => conf,4,GotoIf(${CONFADMIN}?5:7)
exten => conf,5,MeetMe(${CONF},aMXq); admin
exten => conf,6,Hangup
exten => conf,7,MeetMe(${CONF},AMXq); non-admin
exten => conf,8,Hangup
from console, show channels
Zap/pseudo-270871077 [EMAIL PROTECTED]:1 Rsrvd (None)
IAX2/[EMAIL PROTECTED] [EMAIL PROTECTED]:5 Up MeetMe(1701|aMXq)
Niles
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Re: [Asterisk-Users] MeetMe Application - Empty Conference room problem
I have MeetMe setup to pick an empty conference room, and it works great.conf,10,MeetMe(,aMXq)However, I also set the flag (X) to exit the conference room with any key, but I need a wayto get back to the room with the user having to key in the extension numbers.I'm familiar with this MeetMe variable:${MEETMESECS}I've dug this up in app_meetme.c pbx_builtin_setvar_helper(chan, "MEETMESECS", meetmesecs);Can I define another variable to match the conference room, and use to return the user?like, pbx_builtin_setvar_helper(chan,"MEETMEROOM", ???)I got no reply on this question, but I ended up finding a solution (In case anyone else needs to do the same)by modifying app_meetme.cSimple as this:meetme.patch881d880< pbx_builtin_setvar_helper(chan, "MEETMECHANNEL", conf->confno);___
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[Asterisk-Users] MeetMe Application - Empty Conference room problem
Hello,I have MeetMe setup to pick an empty conference room, and it works great.conf,10,MeetMe(,aMXq)However, I also set the flag (X) to exit the conference room with any key, but I need a wayto get back to the room with the user having to key in the extension numbers.I'm familiar with this MeetMe variable:${MEETMESECS}I've dug this up in app_meetme.c pbx_builtin_setvar_helper(chan, "MEETMESECS", meetmesecs);Can I define another variable to match the conference room, and use to return the user?like, pbx_builtin_setvar_helper(chan,"MEETMEROOM", ???)Is there a better solution?Thanks!Niles___
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[Asterisk-Users] Realtime Voicemail & mailcmd=
Hello, I need to alter the sendmail flags on my asterisk box so that I may define the reply address, and I read in the app_voicemail.c file that it can done in voicemail.conf /* Default mail command to mail voicemail. Change it with the mailcmd= command in voicemail.conf */ #define SENDMAIL "/usr/sbin/sendmail -t" I'm using Realtime, so my voicemail.conf file is empty. Changing the source would be simply enough, but I'd rather do it the right way. Any advice is appreciated! Niles ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MeetMe Marked user?
On Aug 24, 2005, at 11:21 AM, Doug Lytle wrote:[EMAIL PROTECTED] wrote: Hello,But does not go into how to mark a user. voip-info archives, and google didn't lead me to any clue, anddigging to app_meetme.c wasn't fruitful.Anyone have an example on how they marked a user in their dialplan? Create an extension that the user to be marked knows about, maybe even have it authenticate, mark the user and drop them into the conference.DougIf the Marked user isn't the first to enter the channel, then how does the MeetMe app know to put all otherusers on hold until Marked user arrives? This is still unclear to me.ThanksNiles Ingalls___ 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] MeetMe Marked user?
On Aug 24, 2005, at 7:40 PM, Doug Lytle wrote:[EMAIL PROTECTED] wrote: On Aug 24, 2005, at 11:21 AM, Doug Lytle wrote: Create an extension that the user to be marked knows about, maybe even have it authenticate, mark the user and drop them into the conference.Doug If the Marked user isn't the first to enter the channel, then how does the MeetMe app know to put all otherusers on hold until Marked user arrives? This is still unclear to me. Example:meetme.confconf => 1000extensions.conf; ** Normal users enter the conference here **exten => 4823,1,SetMusicOnHold(random)exten => 4823,2,Meetme(|Msciw)exten => 4823,3,Hangup(); ** Extension to mark conference users*exten => 4824,1,Authenticate(12345)exten => 4824,2,Meetme(|Asci)exten => 4824,3,Hangup()Users using extension 4823 and entering conference 1000 will listen to hold music until the marked users enters.Users using extension 4824 and entering a password of 12345 will be able to select conference 1000 as the marked user.DougThanks Doug,That clears it up perfectly.Niles___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MeetMe Marked user?
Hello,The MeetMe application offers this flag:'w' — wait until the marked user enters the conferenceAll other connected users will hear MusicOnHold until the marked user enters.But does not go into how to mark a user. voip-info archives, and google didn't lead me to any clue, anddigging to app_meetme.c wasn't fruitful.Anyone have an example on how they marked a user in their dialplan?ThanksNiles Ingalls___ 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] Auto Dial - message
Hello,
I've been reading over the auto-dial feature as described at:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%
20out%20deliver%20message
and have been successful at initiating a call, and placing it into
the dial-plan. What I haven't got to
work, is to be able to place a previously recorded message into
someone's voicemail. Using the sample.call
file, I put this together:
mycall.call
Application: Voicemail
Data: s19
Context: autodial
Extension: voice
Priority: 1
Set: audio=park1124863245
extensions.conf
[autodial]
exten => voice,1,Playback(${audio})
I get this error:
pbx_spool.c:202 apply_outgoing: At least one of app or extension must
be specified, along with tech and dest in file /var/spool/asterisk/
outgoing/3174144286.call
I'm trying to get audio file park1124863245.gsm to play into
extension 19's voicemail box.
Can anyone lend me a clue?
Thanks
Niles Ingalls
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[Asterisk-Users] Fax detection
Hello,
I'm attempting to configure my office Asterisk server to do fax
detection for each one of our DID's configured for different users.
Each person in our office has their own phone number, and I want each
to do both voice & fax.
Fax detection works great when configured like this:
exten => fax,1,Macro(faxreceive)
exten => fax,2,SetVar([EMAIL PROTECTED])
exten => h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
"${CALLERIDNUM} ${CALLERIDNAME}")
This will detect any incoming fax for this particular context and
basically work as expected.
I attempted to do some callerid matching to ensure the correct person
gets their fax.
When configured like this, Asterisk doesn't match the incoming DID to
the fax user in question.
exten => fax/3172152560,1,Macro(faxreceive)
exten => fax/3172152560,2,SetVar([EMAIL PROTECTED])
exten => h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
"${CALLERIDNUM} ${CALLERIDNAME}")
This didn't work!
I then tried this:
exten => fax,1,Macro(faxreceive)
exten => fax/3172152560,2,SetVar([EMAIL PROTECTED])
exten => fax/3172152561,2,SetVar([EMAIL PROTECTED])
exten => h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
"${CALLERIDNUM} ${CALLERIDNAME}")
this also didn't work, although it did everything but set the e-mail
variable.
Any ideas?
Niles
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[Asterisk-Users] upgrade to CVS 3/13/05, voicemail problems
Hello,
I upgraded my office from Asterisk 1.0.0 to Asterisk
CVS-HEAD-03/13/05-13:14:04 this weekend, and are now
experiencing some problems accessing voicemail. The web based interface
works fine, in addition to dialing 8500,
which is mapped to:
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
Each extension is setup like this:
exten => 74/74,1,VoiceMailMain2(${EXTEN})
exten => 74/74,2,Hangup
exten => 74,1,Dial(SIP/74,20,t)
exten => 74,2,Voicemail,u74
exten => 74,3,Hangup
exten => 74,102,Voicemail,b74
exten => 74,103,Hangup
this has worked fine for quite some time, but I'm now receiving an error
from all extensions when we attempt
to retrieve voicemail from the phones using the "Message" button on the
phone.
Here is the error I'm now receiving, and the user goes to a fast busy:
WARNING[9013]: pbx.c:1554 pbx_extension_helper: No application
'VoiceMailMain2' for extension (local, 225, 1)
Any help is greatly appreciated!
Thanks
Niles
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[Asterisk-Users] Re: upgrade to CVS 3/13/05, voicemail problems
WARNING[9013]: pbx.c:1554 pbx_extension_helper: No application 'VoiceMailMain2' for extension (local, 225, 1) I see now that VoiceMailMain2 has been depreciated "/VoiceMail is now replaced by VoiceMail2 in the CVS, so voicemail2 will be obsolete soon. The old voicemail is not included in the current CVS. /OJ dec 2003"/ sorry to bother Niles ___ 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] 7960 Dies when network cable connected
Christopher Jacob wrote: Hey All, I have a user whose Cisco 7960 died the other day. When you disconnect the Ethernet cable the phone boots (as far as it can w/o network connectivity) but as soon as you plug in the CAT5 it goes dead. (no lights, no sound, no display, nothing) Christopher, Just as a total guess, check to be sure the PoE portion of his cat5 cable are not either grounded, or touching each other. Also, be sure he's not connected to a PoE capeable switch since cisco is reverse polarity of the PoE standard. Niles ___ 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] Autostart Asterisk on Slackware?
Goran Dj. wrote: Maybe trivial question, but I cannot find an answer: How to autostart Asterisk (daemon) on Slackware 10? I know that I should put something in /etc/rc.d, but what? In my /etc/rc.d/rc.local # Put any local setup commands in here: /sbin/ztcfg /etc/rc.d/rc.hdlc /usr/sbin/asterisk ___ 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 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] BroadVoice Troubles
My question is simply, has anyone received a deposit from these people once you return the equipment in good order? I've been unable to contact them now for almost 2 whole months. Thanks, Bill Church [EMAIL PROTECTED] Bill, Although this is quite OT, I'll reply. I signed up for their service under the BYOD (bring your own device) plan, which clearly states that there is NO disconnect fee when you cancel. After a great deal of grief, I canceled per their rules (e-mail only) and waited almost two months before they finally acknowledged the cancellation. Of course, they were happy to charge my credit card during that period of time, and then they charged me a disconnect fee. It took them two weeks to reply to my complaint, telling me they would get back with me. That was a week ago, and still no reply or refund. Niles ___ 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: Linux routing with T100P problems
On Dec 29, 2004, at 12:18 AM, Matthew Boehm wrote: Hey gang, I was successful in recompiling my 2.4.20 kernel to support HDLC. I was successful in hooking up our T1 line into the zap card. I was successful in being able to ping equipment on the other end of the T1. I was unsuccessful in pinging the outside world from the other end of the T1. I've attached a cheezy image of the network. Here is the routing table: [EMAIL PROTECTED] root]# route Kernel IP routing table Destination Gateway Genmask Flags Metric RefUse Iface 10.0.5.2 * 255.255.255.255 UH0 0 0 hdlc0 10.0.0.0 * 255.255.255.0 U 0 0 0eth1 10.0.3.0 * 255.255.255.0 U 0 0 0eth1 65.78.109.0 * 255.255.255.0 U 0 0 0 eth0 127.0.0.0 * 255.0.0.0 U 0 0 0lo default 65.78.109.2 0.0.0.0 UG0 0 0eth0 There are 2 NICs (10.0.3.10, 65.78.109.10) and 1 T100P (10.0.5.1) on this box. Like I said above, from this machine I can ping everything in every attached network and the outside world. For some reason, I cannot ping the outside world if I am comming from the 10.0.0.* network on the diagram. From that network, I can ping 10.0.5.1 (this box) but nothing else. appears that your box isn't configured for NAT, so you want to brush up on iptables. Most distributions make this pretty easy, and of course each distro has a different approach on where to find the preconfigured scripts. (google) Niles ___ 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] hdlc + te410p + kernel 2.6.9 - anyone done this?
On Dec 17, 2004, at 5:35 PM, Kristian Kielhofner wrote: Hello, Can anyone out there confirm as in "Yes I am doing this right now" that this can be done? I know that the stuff from 2.6 was backported to 2.4.26 - and it works there (so says the wiki) but before I buy a bunch of hardware (or don't buy hardware, depending on how you look at it) I would love to know. Thanks! -- Kristian Kielhofner Why so dead set on Kernel 2.6, if you know hdlc is working in 2.4 ? Niles ___ 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 -- data?
On Nov 22, 2004, at 6:19 PM, Ken D'Ambrosio wrote: Hi, all. I'm thinking of provisioning my (non-PRI) T1 to be part data, part voice (currently, it's only voice). With the T100P, how do I get it to "do" data? Just load up the HDLC module, and it gets an IP address? Or...? I tried RTFM'ing, but there just doesn't seem to be much on it (leastwise, that's readily findable). Thanks much, Ken D'Ambrosio Go to www.voip-info.org and type in hdlc in the search field. There is plenty of information available for this configuration. Niles ___ 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] What do I need to ask my T1 supplier?
On Nov 3, 2004, at 5:32 PM, Scott Nelson wrote: My employer is switching to a new T1 supplier (it was AT&T, we are now going with XO), and sometime in the future we want to replace our PBX with an Asterisk system. What do I need to know to make sure the T1 line is "provisioned" (is that the right term?) correctly for a Digium T100P/TE410P/TE405P? They will split the T1 line into 10 channels of voice and 14 channels of data. From what I understand, they will terminate the T1 into a channel bank, and then from that give is 10 POTS phone jacks and one data port (to go to an Adtran router for our Internet access). Any comments and/or suggestions? Scott ___ Scott, you can skip the channel bank & router, and use asterisk with a T100P to serve your data & voice. You can find all the info you need on the Wiki http://www.voip-info.org/tiki-index.php? page=Asterisk%20Data%20Configuration I use this setup for 11 voice channels and 256K of data from Nuvox. Niles ___ 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] zt hook failed: Device or resource busy
On Nov 3, 2004, at 10:30 AM, Cian O'Sullivan wrote: Hello, I ordered the Devel lite kit, and installed it. I am just trying to get the FXO port to work, and am having trouble. To load the card I do the following. modprobe wcfxs modprobe wcfxo ztcfg -vv asterisk -vc My /var/log/asterisk/messages show add modprobe zaptel before the other modules. ___ 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] Vmail.cgi Bahhh!!
On Oct 20, 2004, at 8:49 AM, Your Own ISP .com wrote: OK, been at this for a few hours now. I am on Fedora 2 trying like heck to get the web based Vmail thing working. I have it to the point where I can login to it successfully but no messages ever show up there even though I know they exist. I am getting the voicemail messages OK in my email. I CHMOD'd the vmail dir to 777 just for testing purposes, I installed the required Perl Module. Not sure what's left to try, any ideas would be most welcome. Thanks, Todd Routhier Lightwave Technologies, LLC. Todd, asterisk records the files to the filesystem with root permissions, which a properly configured apache installation doesn't have access too. I worked around this by setting up a cronjob to chmod 777 all the voicemail files once a minute, which probably isn't the most elegant solution to this problem. my crontab entry: # cheap way to fix our permissions for voicemail * * * * * /etc/vm_chmod.bat > /dev/null /etc/vm_chmod.bat: #!/bin/sh chmod -R 777 /var/spool/asterisk/vm If someone else doesn't give a better solution, you can try this. Niles ___ 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] Wonderful Success with PAP2-NA
On Oct 19, 2004, at 4:17 PM, Brandon Patterson wrote: Brian approved providers get them from Linksys (volume), Tech Data, DH Dist and Mirco D. WHERE DID YOU GET THE PAP2-NA?!??!!? .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] -->IM's<--- MSN: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian you can get it from amazon: http://www.amazon.com/exec/obidos/tg/detail/-/B0002V8KX6/ ref=ase_interactiveda81-20/104-3724093-2519924?v=glance&s=electronics Niles ___ 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] Asterisk and SMP
On Oct 15, 2004, at 11:20 AM, Brian Wilkins wrote: Hi, I have a machine that does SMP (Symmetric Multi-Processing) and I was wondering if it would be a problem if I used a kernel that used SMP with Asterisk? Would it crash? Thanks - -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ 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 never had a problem with my asterisk dual p3 box, or my quad ppro asterisk box. I'd say you're safe. Niles ___ 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] Am I stupid or is my card DOA.?
** Niles Wrote | I had this exact same problem with my new TDM400P I just received last | week. | It's very inconvenient not to be able to restart my phone server | remotely, due to | the TDM400P not restarting. | I was able to successfully reboot my system earlier without | cold-booting the server | by unloading the zaptel modules before restarting the system. | You will will to unload the modules in the opposite order that you | loaded them, and may | want to place them low in your /etc/rc.d/rc.K file (in slackware) for | automation. | (example for rc.K) | | # unload zaptel modules on shutdown. | # asterisk should not be running when this occurs | | /sbin/modprobe -r wcfxs | /sbin/modprobe -r xct1xxp | /sbin/modprobe -r zaptel | | I haven't tested this extensively, but it has worked the one time I | tested it. | | Niles | Niles I tried your suggestion, but still get the hang of the TDM card (4 fxo modules) Regards Greg Greg, did you receive any errors when attempting to unload the modules, and was asterisk still running at that time? Be sure asterisk is not running, and that you remove the modules in reverse. I only tried it once, so it could have been a red herring. Niles ___ 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] RxFax multiple pages
On Oct 13, 2004, at 8:07 AM, Vladyslav wrote:
Hi.
Thank you all for your replies.
Now I do converting into pdf file and it's ok with multiple pages.
tiff2ps -a ${FAXFILE} | ps2pdf - ${FAXFILE//tiff/pdf}
On Wed, 2004-10-13 at 15:39, Steve Underwood wrote:
Vladyslav wrote:
You can also cut to the chase, and
tiff2pdf -p letter ${FAXFILE}
Niles
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Re: [Asterisk-Users] TDM01B Goes missing after reboot
On Oct 12, 2004, at 7:38 PM, Ian D. Wlloughby wrote: Hi All, I have just installed a TDM01B to fix my UK callerid and echo problems. In this respect everything is wonderful, however when I reboot wcfxs fails to load due to "No Device found". If I power off and on everything is fine. I noticed that wctdm does not appear in /proc/interrupts after the reboot but does after power off/on. This seems similar to other peoples problems, do I have a duff card (Revision H) or is this a bug in wcfxs ? Regards Ian Ian, I responded to a similar posting today. With any luck, this workaround will also work for you. http://lists.digium.com/pipermail/asterisk-users/2004-October/ 067004.html Niles ___ 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] Slackware 10.0/Asterisk 1.0 compile error
On Oct 12, 2004, at 7:10 PM, David McNett wrote: On 13-Oct-2004, Dee Lowndes wrote: If you compiled 0.9.1 on the same system make sure you remove all old source dir's, /var/lib/asterisk and that X is installed. I did this and it all installed perfectly well on my slack 10 system. I also had this same problem with slackware 10. Slackware 10 ignorantly installs the gtk2 libs even when you've opted not to install X11. This alone wouldn't be a problem, but the asterisk makefiles use the presence of gtk2 to determine whether or not to build the X11 components. I just took the lazy way out (hey -- it's slackware, right?) and installed the X libs on the box. That's all it took. heck, that sounds like the hard way instead of the lazy way. just do: removepkg gtk+ , then install * Niles -- David McNett <[EMAIL PROTECTED]> http://slacker.com/~nugget/ ___ 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
Re: [Asterisk-Users] Am I stupid or is my card DOA.?
On Oct 12, 2004, at 12:55 PM, [EMAIL PROTECTED] wrote: I have seen this behaviour as well with the t100p and tdm04b. I have to power down, reboots don't work. For cards as pricey as these you would think they would flush / refresh on a reboot. ps. still trying to get the t100p to work in a data/voice environment with little or no luck or documentation (I've got all the google and voip docs) still no luck. Regards Greg On 11-Oct-2004, Alex Barnes wrote: I had/have exactly the same problem with my X100P / TDM400P dev setup. I'm also having exactly the same problem with a TDM400P I received yesterday. I'm starting to suspect that seeing it work after swapping PCI slots is a placebo effect. Without moving the card it randomly seems to "work" about half the time. I have yet to successfuly get Asterisk to utilize the card, even on boots where the modules successfully load, but that might be misconfiguration on my part. I plan to call Digium for help either today or tomorrow. -- David McNett <[EMAIL PROTECTED]> http://slacker.com/~nugget/ ___ 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 I had this exact same problem with my new TDM400P I just received last week. It's very inconvenient not to be able to restart my phone server remotely, due to the TDM400P not restarting. I was able to successfully reboot my system earlier without cold-booting the server by unloading the zaptel modules before restarting the system. You will will to unload the modules in the opposite order that you loaded them, and may want to place them low in your /etc/rc.d/rc.K file (in slackware) for automation. (example for rc.K) # unload zaptel modules on shutdown. # asterisk should not be running when this occurs /sbin/modprobe -r wcfxs /sbin/modprobe -r xct1xxp /sbin/modprobe -r zaptel I haven't tested this extensively, but it has worked the one time I tested it. Niles ___ 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] Re: outgoing calls, based on caller extension
On Sep 24, 2004, at 1:07 AM, Tom Ivar Helbekkmo wrote:
[EMAIL PROTECTED] writes:
exten => 109/_9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
[...]
as you can see, I'm wanting 109 to dial out broadvoice_1, [...]
You've got the source and target reversed. :-)
You want the Caller ID of your local extension 109 to be "109", and
then you should say:
exten => _9NXXNXX/109,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
...and so forth. Note that the source specification can also be a
pattern, in which case it should, like the target, start with "_".
-tih
--
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
Fantastic Tom,
This did the trick.
I'm curious if there is a way I can do some kind of balancing if an
outgoing
connection is already being used?
I was thinking about using the System command with a python script to
keep
an inventory of what outgoing connections are being used, and use the
output
from the script to determine which SIP account to make the call from.
If there isn't an easier way, then I'll work into that direction.
Thanks
Niles
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[Asterisk-Users] outgoing calls, based on caller extension
Hello All,
I have multiple SIP accounts from two different providers, and I'm
wanting
to balance our outgoing calls based on extensions. I thought the
following
would work, but it's sending all calls through the last SIP account
listed
(extensions.conf)
exten => 109/_9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => 110/_9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => 101/_9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => 102/_9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
as you can see, I'm wanting 109 to dial out broadvoice_1, 110 via
broadvoice_2,
and so on.
I've tried a variety of things with no success. Any info would be great.
Niles
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Re: [Asterisk-Users] 1.0 Mirrors
Here's a HTML link I'll leave active for a few weeks: http://www.atheos.net/asterisk/asterisk-1.0.0.tar.gz Niles On Sep 23, 2004, at 9:21 AM, Greg Boehnlein wrote: Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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
Re: [Asterisk-Users] cisco 7960 CTLSEP
What version firmware does your phone currently have? on my 7940's & 7960's, I've had to stair step each firmware version, starting at 3 in order to get to 7.2 Niles On Sep 17, 2004, at 10:30 AM, Jan Baggen wrote: 2 new Cisco 7960 phones are requesting a CTLSEP file, seems like I triggered the universal application loader. I want to load the sip image 7.2 According to this Cisco information: http://www.cisco.com/en/US/customer/products/sw/voicesw/ps4967/ products_upgr ade_guides09186a008022a968.html#wp1047292 If the CTLSEP MAC file is not present or is empty, the phone proceeds in nonsecure mode with the hunt algorithm. But this does happen. I just loops in requesting the *.tlv file for hours. 17:57:35.515349 x.x.x.1.50926 > x.tftp: 31 RRQ "CTLSEP00115C330355.tlv" 17:57:36.231681 x.x.x.2.50947 > x.tftp: 31 RRQ "CTLSEP001121F12961.tlv" 17:57:36.513591 x.x.x.1.50926 > x.tftp: 31 RRQ "CTLSEP00115C330355.tlv" 17:57:40.232408 x.x.x.2.50947 > x.tftp: 31 RRQ "CTLSEP001121F12961.tlv" 17:57:40.512814 x.x.x.1.50926 > x.tftp: 31 RRQ "CTLSEP00115C330355.tlv" 17:57:44.238255 x.x.x.2.50948 > x.tftp: 31 RRQ "CTLSEP001121F12961.tlv" 17:57:44.513544 x.x.x.1.50926 > x.tftp: 31 RRQ "CTLSEP00115C330355.tlv" 17:57:45.235617 x.x.x.2.50948 > x.tftp: 31 RRQ "CTLSEP001121F12961.tlv" 17:57:48.519508 x.x.x.1.50927 > x.tftp: 31 RRQ "CTLSEP00115C330355.tlv" 17:57:49.235716 x.x.x.2.50948 > x.tftp: 31 RRQ "CTLSEP001121F12961.tlv" 17:57:49.516877 x.x.x.1.50927 > x.tftp: 31 RRQ "CTLSEP00115C330355.tlv" 17:57:53.238186 x.x.x.2.50948 > x.tftp: 31 RRQ "CTLSEP001121F12961.tlv" 17:57:53.517099 x.x.x.1.50927 > x.tftp: 31 RRQ "CTLSEP00115C330355.tlv" How to load the SIP image on these phones? The TFTP directory: -rw-r--r-- 1 root wheel13 Sep 17 17:12 OS79XX.TXT -rw-r--r-- 1 root wheel124716 Aug 13 19:23 P003-07-2-00.bin -rw-r--r-- 1 root wheel125120 Aug 13 19:24 P003-07-2-00.sbn -rw-r--r-- 1 root wheel124716 Aug 13 19:23 P0S3-07-2-00.bin -rw-r--r-- 1 root wheel 461 Aug 13 19:44 P0S3-07-2-00.loads -rw-r--r-- 1 root wheel587122 Aug 13 19:40 P0S3-07-2-00.sb2 -rw-r--r-- 1 root wheel571489 Aug 16 22:28 P0S3-07-2-00.zip thank you! --- Jan Baggen - [EMAIL PROTECTED] IP2 Internet BV / http://www.ip2.nl ___ 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
Re: [Asterisk-Users] Asterisk and software Raid
I've got a nearly identical system running on Slackware 9.1 with a Megaraid controller. Works better than expected! Niles > I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that I > can experiment with. I've been wanting to use Redhat with software Raid 1 > on an Asterisk server. > > Has anyone had any experience with software raid and Asterisk? Also, if > the software raid doesn't play, any recommendations for a hardware based > IDE Raid controller and suggestions on best practices for setting up the > disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be > appreciated. > > Thanks all ___ 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] Polycom IP 500 - MWI Not Working
Just make sure you have defined the mailbox number in your sip.cfg file. example: [222] context=local type=friend host=dynamic username=222 password=p222 dtmfmode=inband defaultip=10.1.2.222 mailbox=222 That's all it took for my asterisk + IP500 setup. Niles On Aug 10, 2004, at 3:33 PM, Wiley E. Siler wrote: Hello All, I have Polycom IP 500 phones which I would like to have message waiting indicators on. So far, I have my system running well but the problem I am seeing is that MWI doesn't seem to tell my phone that it should display a MWI state. The light does not show when you have message nor is there any indicator on the text lines of a message waiting. The wiki doesn't cover this enough to help me find why I do not get the notification on the phone when a message is waiting. Is tehre anyone out there with Polycom phones who has Message Waiting Indicators working with the IP 500? If so, can you tell me how you got it working, what variable to set in * or the Polycom cfg files? Thanks! W
Re: [Asterisk-Users] Asterisk on Sparc64
> > I'd have thought this was on Sparc64 (i.e. UltraSparc III) if it's a > Sun Ultra60, nought to do with Opteron. > > > Steve > The Opteron based SUN hardware are not called sparcs, so you're correct. Niles ___ 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] Voicemail Hangup detection issue
Hello, I finished an Asterisk installation this weekend, and I'm experiencing a problem when a user hangs up on a line after leaving a voicemail message. I found two similar issues when reading through the archives, and have not been able to resolve my issue from their answers. http://lists.digium.com/pipermail/asterisk-users/2004-April/042453.html http://www.marko.net/asterisk/archives/0212/.html At first, the voicemail messages would contain 7 minutes of busy signal at the end of the message, and now it contains about 40 seconds after adding busydetect=yes to zapata.conf and installing the latest CVS. Any ideas would be greatly appreciated. Niles Ingalls I'm using a Wildcard T100P, and have 11 incoming lines. zapte.conf span=1,0,0,esf,b8zs loadzone=us defaultzone=us fxsls=1-11 zapata.conf ; Zapata telephony interface ; Configuration file [channels] musiconhold=default language=en context=default switchtype=dms100 signalling=fxs_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes busydetect=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no channel => 1 channel =>3-11 group=2 callgroup=2 pickupgroup=2 context=fxo1 channel => 2 ___ 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
