[Asterisk-Users] Call Rollover
Have a question about implementing Call Rollover with my
current extensions.conf configuration.
[macro-stdexten]
exten => s,1,Dial(${ARG2},20) ; Ring the
interface, 20 seconds maximum
exten => s,2,Voicemail2(u${ARG1}) ; If unavailable,
send to voicemail w/ unavail announce
exten => s,3,Goto(default,s,1) ; If they press
#, return to start
exten => s,102,Voicemail2(b${ARG1}) ; If busy, send
to voicemail w/ busy announce
exten => s,103,Goto(default,s,1) ; If they press
#, return to start
[mainmenu]
exten => s,1,Zapateller
exten => s,2,Wait,3
exten => s,3,Answer
exten => s,4,DigitTimeout,5
exten => s,5,ResponeTimeout,10
exten =>
s,6,Background(/var/lib/asterisk/sounds/vm/100/Work/MainMenu/SDSgreet)
exten => 1,1,Goto(salesmenu,s,1)
exten => 2,1,Goto(technicalmenu,s,1)
exten => 3,1,Directory,default
exten => 4,1,Goto,s|6
exten => 5,1,Goto(remoteline,s,1)
exten => i,1,Playback(invalid)
exten => t,1,Goto(s,7)
exten =>
s,7,Background(/var/lib/asterisk/sounds/vm/100/Work/MainMenu/goodbye)
exten => s,8,Hangup
include => SDSextensions
include => departmentextensions
[salesmenu]
exten => s,1,Ringing
exten => s,2,Wait,2
exten => s,3,Goto(SDSextensions,202,1)
exten => #,1,Goto(mainmenu,s,4)
[technicalmenu]
exten => s,1,Ringing
exten => s,2,Wait,2
exten => s,3,Goto(SDSextensions,204,1)
exten => s,4,Goto(SDSextensions,206,1)
exten => s,5,Goto(SDSextensions,207,1)
exten => #,1,Goto(mainmenu,s,4)
[SDSextensions]
include => macro-stdexten
exten => 201,1,Macro(stdexten,${KHVM},${KH})
exten => 202,1,Macro(stdexten,${BOVM},${BO})
exten => 203,1,Macro(stdexten,${GOVM},${GO})
exten => 204,1,Macro(stdexten,${DWVM},${DW})
exten => 205,1,Macro(stdexten,${RFVM},${RF})
exten => 206,1,Macro(stdexten,${BRVM},${BR})
exten => 207,1,Macro(stdexten,${CGVM},${CG})
exten => 208,1,Macro(stdexten,${AGVM},${AG})
exten => 209,1,Macro(stdexten,${MDVM},${MD})
exten => 300,1,Macro(stdexten,${TRVM},${TR})
Conventions used in my Macro stdexten lines:
InitialsVM = SIP address
initials = extension
Example for user John Doe:
JD=SIP/JohnDoe
JDVM=400
My issue is that I want any calls coming into my
[technicalmenu] context to rollover after a period of time to the next
extension, 206, and then to 207. If none of these
extensions pickup I want the call to route back to the
initital extensions voicemail. I am using the stdexten macro to implement the
busy and unavailable voicemail automation. I am using the Goto statements to goto
the [SDSextensions] context and use the stdexten macro. I want to know if
there is a way with my current implementation to enter the technical menu ->
follow the first extension to my [SDSextensions] context -> execute the
stdexten macro and then have that macro timeout without going to voicemail and then
execute the other extensions until the last extension is executed and then
return to the first extensions voicemail.
System Information:
Asterisk CVS version: 12/11/2003-12:22:41
System:
Dell Dimension 8300
Pentium 4 2.8GHz
512 Mb memory
120 Gb hard drive
3 X100P Cards
1 DSL Line
3 POTS
6 Snom200 Phones
Thanx in advance, this list has been very helpful, keep it
up!
Ryan R. Fligg
Secured Digital Storage, Inc.
104 SW 4th St.
Des Moines, IA 50309
Phone: (515)-244-6290
Cell: (720)-841-5802
Website: www.dstorage.com
E-Mail: [EMAIL PROTECTED]
[Asterisk-Users] X100P Cards have gone belly up?
Alright,
I have quite a problem on my hands and even the digium engineers are
stumped. First my system layout
Asterisk CVS-01/15/04-16:44:03 built by [EMAIL PROTECTED] on a i686 running Linux
X101P cards: 3
SNOM200 Phones: 5
2 outgoing lines and 1 incoming line (DSL)
Okay, so one day our systems stopped working all together. I know this is
very vague but I assure you that I did not change anything.
Our problem was that we were unable to dialout. Receiving calls works just
fine. Let me break this down.
Asterisk was up and running perfectly with no problems. The X100P cards
were detected upon bootup and seemed to be loading just fine.
In our dialplan I have setup several contexts in which our callers are
limited to Local, Long Distance and emergency/immediate numbers. A user
will
Dial a 9 before any call and the pattern matching will interpret what kind
of call it is, as is defined in my dialplan. When a user would try to
initiate ANY call the response from the console was that the X100P card was
dialing out, but the user would hear nothing on the Snom200 end even though
I could verify from the console that the X100P card was bridging the call to
the Snom200 phone.
I then hooked up an analog phone to the X100P card and found the the X100P
was only dialing about ½ the number that was entered in the Snom200 phone.
I was a little stumped here so I called the digium tech support and the
ssh'd into our machine and were stumped. They used zapbarge to listen on
the X100P card remotely and verified what I had heard. Their fix to the
solution was to ditch all my contexts and use one universal context
exten => _9,1,Dial(${OUT}/)
Now all my users dial 9 and then get tone to an outside line to dial.
This is fine and dandy for our office of 5 at the moment but the
scaleability of our system has been GREATLY compromised. All the future
accounting practices that I was going to implement with MySQL and other
features are not available now.
Here are the steps I took after getting the Digium solution:
1) Built a new Asterisk Box and tried the cards: Results: Same
2) Put a brand new card in the new asterisk box with current CVS:
Results: Same
3) Tested our lines and found that the current was about 75 mA, built
resistor packs and lowered it to 23 mA, industry standard
4) Used same hardware configuration as in 2 but with a new X100P card,
with lowered lines: Result: Same
5) Took configuration in 4 to a residential location with decent line
amperage and tested: Result: Same
So after determining that reordering new X100P cards from Digium would be
useless because I got the same results with our spare X100P, I am at a loss
as what to do next. I will include my any files any of you would need to
look at my configuration upon request because I don't want to make this
e-mail lengthy with my rather large extensions.conf file and others.
Thank you in advance for all your help.
Sincerely,
Ryan R. Fligg
Secured Digital Storage, Inc.
104 SW 4th St.
Des Moines, IA 50309
Phone: (515)-244-6290
Cell: (720)-841-5802
Website: www.dstorage.com
E-Mail: [EMAIL PROTECTED]
<>
[Asterisk-Users] Voicemail Password Digit Timeout
I was wondering if there was any way to change the digit timeout or some setting of that sort on the voicemail password entry. Currently when our users enter their passwords they have to enter them very rapidly, otherwise asterisk will log the number twice. So if someone entered a voicemail password of 1234 slowly and deliberately on our system the asterisk receives it as the following number, 11223344 and thus returns the passcode invalid message. System: Asterisk CVS-02/10/04-13:27:57 built by [EMAIL PROTECTED] on a i686 running Linux 3 X100P cards 5 Snom200 phones Sincerely, Ryan R. Fligg Secured Digital Storage, Inc. 104 SW 4th St. Des Moines, IA 50309 Phone: (515)-244-6290 Cell: (720)-841-5802 Website: www.dstorage.com E-Mail: [EMAIL PROTECTED] <>
[Asterisk-Users] Status Lights on Snom200 Phone Displaying the Status of PSTN Lines
Alright, this may seem like something relatively easy to do
but I must be missing something or had a neuron misfire. I am trying to
get
The Status lights on my Snom200 hardphones to display the
status of each one of my PSTN lines in my Asterisk server.
Current Config:
3 X100P cards
Asterisk CVS-02/25/04-18:06:52
5 Snom200 phones
I am currently using the following macro to dial out with my
extensions
[macro-stdexten]
exten =>
s,1,Dial(${ARG2},20,Ttr)
; Ring the interface, 20 seconds maximum
exten =>
s,2,Voicemail2(u${ARG1}) ; If unavailable,
send to voicemail w/ unavail announce
exten =>
s,3,Goto(default,s,1) ;
If they press #, return to start
exten =>
s,102,Voicemail2(b${ARG1}) ; If busy, send to voicemail
w/ busy announce
exten =>
s,103,Goto(default,s,1) ; If they press
#, return to start
My extension can then use this as follows:
exten => 202,1,Macro(stdexten,${USERVM},${USER})
I do not have an extension setup directly to dialout and I
am thinking that this might be part of my problem. I would also like to
know the status of certain extensions
To see who is on the phone and who is not by looking at the
soft key light on the Snom200. If anyone has any ideas, they would be
greatly appreciated.
Sincerely,
Ryan R. Fligg
Secured Digital Storage, Inc.
104 SW 4th St.
Des Moines, IA 50309
Phone: (515)-244-6290
Cell: (720)-841-5802
Website: www.dstorage.com
E-Mail: [EMAIL PROTECTED]
RE: [Asterisk-Users] x100p volume
Chris, Be careful with setting the volume in this manner. This tip might save you a bunch of headaches. I adjusted the TX on my X100P cards to low negatives values in accordance with the feedback I received from ztmonitor. When I achieved the levels that were satisfactory to me an interesting problem arose. My X100P would not longer dial out. This makes complete sense now, because the TX levels were not high enough to trigger an "off-hook" state. But at the time I was trying to minimize echo and changing a couple of other things. This problem completely baffled the Digium Tech guys and only after going to back to old CVS, different hardware, checking impedance on our lines and a little sitting back and digesting did it the solution present itself. So just a word of warning, if you are messing with these settings and you can no longer dialout, change them back before you check anything else. Sincerely, Ryan R. Fligg -Original Message- From: Chris Clifton [mailto:[EMAIL PROTECTED] Sent: Monday, March 08, 2004 9:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] x100p volume Calls to the pstn on our x100p experience a very low volume. Any parameters to change this ? 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 ___ 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] Good corporate level speaker phone for use with Asterisk
Hey, I am looking for corporate level solution for a conference phone to use with an Asterisk system. Any ideas? Current System: Asterisk CVS-02/25/04-18:06:52 Red Hat 9.0 3 X100P cards 3 PSTN lines Ryan R. Fligg Secured Digital Storage, Inc. 104 SW 4th St. Des Moines, IA 50309 Phone: (515)-244-6290 Cell: (720)-841-5802 Website: www.dstorage.com E-Mail: [EMAIL PROTECTED] <>
RE: [Asterisk-Users] two things
Try setting the dtmfmode=rcf2833 and also make sure you have suidperl installed. These things really helped in getting my X-Lite to work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Thursday, October 30, 2003 10:46 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] two things Hi, I'm having two problems. First – I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before>? Second thing: I get a WARNING:[1209214400]: File dsp.c, line 1198 (ast_dsp_process): unable to detect process 2 frames All the time. What gives? Please excuse a newbie…
[Asterisk-Users] Scaleable Solution for small office
Hi, Have posted to this list a couple of times and have always received great responses and help. I have a basic * system setup Using 3 X100P cards with 6 Snom200 IP phones. It was a bit of a struggle getting everything up and running but have been pretty happy with the flexibility and ease of *. My major problem is one that has been discussed on this list many times before. The echo from the X100P cards is completely irritating to my users. I have tweaked the system as much as I can, IE: switching tip and ring, using aggressive suppressor, tweaking the TX/RX settings. There is still echo in the first 0-8 seconds of the call while the echo cancellation is catching up. I understand the problem is that you do not hear this echo in analog systems even though it is present and it is only heard in the digital system because it is not as fast as analog. So here is my problem. Our CEO wants me to get rid of the system unless I can provide a solution that will expand and work just as good as an analog replacement PBX. He came to me with this decision after trying to use the Snom200 speaker phone for a conference call, not a good idea as we all know the speaker on this phone is not a conference phone replacement. He also was frustrated when he tried to patch a call that came to another extension, which transferred fine, but was disconnected when the receiving party made a switch on their PBX to another extension and it was dropped. Currently we have 4 POTS lines and I have 1 setup for DSL and the other 3 hooked into the X100P cards, 2 outgoing and 2 incoming, using the DSL as incoming rollover. We are planning on expanding our business pretty rapidly and he wants a system that will be easy to setup and scale. I know asterisk and VOIP phones are great for this but the little glitches in the phones, hardware wise, are not supporting my Asterisk decision. I would like to upgrade to a channel bank, and was wondering if anyone has had any echo issues with other digium hardware. I know that the X100P issue comes from not having ECAN DSP in the card. I was also wondering if anyone had any luck adding a conference phone such as the Polycom Soundstation Conference Phone. Any suggestions would be appreciated Ryan R. Fligg Secured Digital Storage, Inc. 104 SW 4th St. Des Moines, IA 50309 Phone: (515)-244-6290 Cell: (720)-841-5802 Website: www.dstorage.com E-Mail: [EMAIL PROTECTED]
RE: [Asterisk-Users] Live Music on Hold
Dan, I played around with this and have also been following the MOH posts. Playing MP3's seems to crash my system but I did get music on hold to stream. Here is my musiconhold.conf file: ; ; Music on hold class definitions ; [classes] ;default => quietmp3:/var/lib/asterisk/mohmp3/mp3 ;loud => mp3:/var/lib/asterisk/mohmp3/mp3 ;random => quietmp3:/var/lib/asterisk/mohmp3/mp3,-z ;stream => quietmp3:/var/lib/asterisk/empty,http://64.236.34.141:80/stream/1006,http:// 64.236.34.161:80/stream/1040,http://64.202.98.33:6230,http://69.10.147.34:80 40,http://64.202.98.75:6180,-z I just used a couple of my favorite internet radio stations. This is a start. I have documentation of exactly what I did to get it to work. Let me find it and I will post a followup reply. Good Luck. Ryan R. Fligg Secured Digital Storage, Inc. 1171 7th St. Suite 100 Des Moines, IA 50314-2525 Phone: (515)-244-6290 ext.205 Fax: (515)-244-6285 Cell: (515)-988-3773 E-Mail: [EMAIL PROTECTED] Website: http://www.dstorage.com _ From: Dan B. Long [mailto:[EMAIL PROTECTED] Sent: Thursday, April 08, 2004 3:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Live Music on Hold I have a small * system in my home (1 U100S, 1 X100P, 1 BT101, and 1 SPA2000) to handle my requirements. I would like to add Music on Hold and have been watching the forum to see if something would come across on this topic. The difference I am interested in is getting the music from a radio or someother external source. All references to MOH up to now have been using MP3 files and going through them. Is there some way to get the music from an external source instead of from files? Dan Long <>
