[asterisk-users] Call Parking/Pickup on a single button
Is it possible with asterisk to use a single button to park and retrieve a call? e.g. Button is labelled Park 701 - If it is not in use, park the current call to 701 - If it is in use (the associated LED will be lit), pickup the call at 701 (putting the current call [if any] on hold). A Polycom IP600 phone would have two or three of these keys (701, 702, 703). Any suggestions appreciated! Alvin ___ --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] Polycom SoundStation VTX 1000 with Asterisk?
Anyone successfully using the Polycom SoundStation VTX 1000 with Asterisk? I can't see any mention of it on the wiki page: http://www.voip-info.org/wiki-Polycom+Phones Thanks, Alvin ___ --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] Polycom IP601 call parking
One more Polycom IP601 question please (sorry for the long intro here
to document) ...
In order to closely approximate the behavior of the previous telephone
system that many of the users are familiar with, I have set up call
parking like this:
- features.conf [general] section contains:
parkext = ** ; What extension to dial to park
parkpos = 10-11 ; What extensions to park calls on.
context = parkedcalls ; Which context parked calls are in
parkcall = ** ; Park call (one step parking) - put
Kk in Dial options
- extensions.conf dialing context contains:
include = parkedcalls
exten = 10,hint,park:[EMAIL PROTECTED]
exten = 11,hint,park:[EMAIL PROTECTED]
[..]
exten = _20[1-7],1,Dial(SIP/${EXTEN},30,Kk)
Two buttons on the phones show Park 10 and Park 11, and flash
nicely when a call is parked there.
A typical phone specific file (mac)-directory.xml
(e.g.: 0004f123456-directory.xml) looks like:
?xml version=1.0 standalone=yes?
directory
item_list
itemfnPark 10/fnct10/ctsd1/sd
dc/ad0/adar0/arbb0/bbbw1/bw/item
itemfnPark 11/fnct11/ctsd2/sd
dc/ad0/adar0/arbb0/bbbw1/bw/item
[..]
/item_list
/directory
It works pretty well. A call is received and the recipient presses **
to autopark the call. The call is parked to the first available
parking slot (10) and the recipient hears ten and hangs up. The
Park 10 LED flashes on the phone to indicate the parked call. The
person picking up the call presses the Park 10 button and gets the
call.
But what I'd *really* like to do is to have the recipient push either
the Park 10 or Park 11 buttons to park the call to that slot
(instead of **), and then the person picking up the call presses the
same button on their phone (which is flashing) to pick up the call.
Is this possible with Asterisk and these Polycom phones? (Asterisk
1.4.13, Polycom sip.cfg 2.1.2)
Thanks,
Alvin
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[asterisk-users] Polycom IP601 (mac)-directory.xml changes don't update phone
Hi Polycom experts, I'm having a problem getting changes to the Polycom IP 601's (mac)-directory.xml file to update the button list on the phone. If the phone is newly provisioned (i.e. if I Format File System on the phone) then the new list will show up on the buttons, but of course this is pretty drastic way to do it. - Environment: Asterisk test setup with 7 phones, bootrom 3.1.3, sip 2.1.2. - I'm provisioning the phones with ftp and the phones happily download configuration files and upload logs. - The -directory.xml is pretty empty: ?xml version=1.0 standalone=yes? directory item_list /item_list /directory - A typical phone specific file (mac)-directory.xml (e.g.: 0004f123456-directory.xml) looks like: ?xml version=1.0 standalone=yes? directory !-- Ext: 202 -- item_list itemfnPark 10/fnct10/ctsd1/sddc/ad0/adar0/arbb0/bbbw1/bw/item itemfnPark 11/fnct11/ctsd2/sddc/ad0/adar0/arbb0/bbbw1/bw/item itemfnExt 201/fnct201/ctsd3/sddc/ad0/adar0/arbb0/bbbw1/bw/item itemfnExt 203/fnct203/ctsd4/sddc/ad0/adar0/arbb0/bbbw1/bw/item itemfnExt 204/fnct204/ctsd5/sddc/ad0/adar0/arbb0/bbbw1/bw/item /item_list /directory If I change a (mac)-directory.xml file on the ftp server, I'd like to push it out to Polycom phone and ideally update the buttons on the fly. What's the right way to do this? Thanks, Alvin ___ --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] PRI Setup problem
Thanks for all of the good suggestions. I've been able to get things working. I had been trying to use zaptel svn in order to get past error messages with compiling ztdummy.ko for the 2.6.22 kernel (http://bugs.digium.com/view.php?id=10426 which has been apparently been solved in svn). Too bleeding edge, I guess. I reverted back to a tried-and-true kernel (2.6.17), and then compiled the stable 1.4 zaptel/libpri/asterisk/wanpipe versions, and things are happy now. Regards to all, Alvin Stephen Bosch wrote: Alvin Austin wrote: I've recompiled with the latest svn sources for zaptel, libpri, and Asterisk. Wanpipe is 3.3.0.p4. Switched the T1 cable. Same result. Hmn -- when you recompiled, did you 1. clean out all the source directories? 2. remove the binaries? 3. recompile in the right order? I'm not sure using SVN is a good idea here. It should work with stable ;) Has the PRI been tested with test equipment? We should make sure there is a D channel before assuming misconfiguration. I don't think we can do even a loopback test if there is no D channel... -Stephen- ___ --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
[asterisk-users] PRI Setup problem
Hi everyone, I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI (Megalink) circuit and having some trouble getting it to handshake. Thanks for any help or suggestions to diagnose this problem. The problem is that Asterisk has trouble bringing up the link. I see the following lines every couple of minutes: == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active == Primary D-Channel on span 1 down [Oct 1 17:52:49] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! h87*CLI pri show spans PRI span 1/0: Provisioned, Down, Active == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down [Oct 1 17:55:20] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! h87*CLI pri show spans PRI span 1/0: Provisioned, Down, Active Of course I cannot dial out: -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/368-081f51d8, Dial Time of Day via PRI) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/368-081f51d8, ZAP/3|2446411|30|Tt) in new stack [Oct 1 18:01:27] WARNING[13623]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED] :3] Congestion(SIP/368-081f51d8, ) in new stack == Spawn extension (default, 2446411, 3) exited non-zero on 'SIP/368-081f51d8' If I turn on pri debugging, I see lots of: h87*CLI pri debug span 1 Enabled debugging on span 1 Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Periodically I see: Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 88] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 8 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- Timeout occured, restarting PRI q921.c:356 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending Set Asynchronous Balanced Mode Extended q921.c:150 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH == Primary D-Channel on span 1 down [Oct 1 18:04:09] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Any help is greatly appreciated! Thanks, Alvin ___ --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] PRI Setup problem
I've recompiled with the latest svn sources for zaptel, libpri, and Asterisk. Wanpipe is 3.3.0.p4. Switched the T1 cable. Same result. (It's a Sasktel Megalink T1/PRI circuit) CLI shows: ~~ == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down [Oct 1 20:15:19] WARNING[7120]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up CLI pri show spans alternates between: PRI span 1/0: Provisioned, Down, Active (most of the time) and: PRI span 1/0: Provisioned, Up, Active(during its retry sequence - below) with debugging enabled: CLI pri debug span 1 ~~ Sending Set Asynchronous Balanced Mode Extended [..] Sending Set Asynchronous Balanced Mode Extended -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up [above paragraph repeated 7 more times] Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- Timeout occured, restarting PRI q921.c:356 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending Set Asynchronous Balanced Mode Extended q921.c:150 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH == Primary D-Channel on span 1 down [Oct 1 20:30:49] WARNING[7003]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Sending Set Asynchronous Balanced Mode Extended [..] Sending Set Asynchronous Balanced Mode Extended # wanrouter status ~~ Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1| N/A | A101/1D/A102/2D/4/4D/8| 16 | 4 | 1| EXT | 0 | Wanrouter Status: Device name | Protocol | Station | Status| wanpipe1| AFT HDLC | N/A | Connected | File /etc/zaptel.conf is: ~~ # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:4 bus:41 span: 1] span=1,0,0,esf,b8zs bchan=1-23 dchan=24 File /etc/asterisk/zapata.conf is: ~~ ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A101 port 1 [slot:4 bus:41 span: 1] ; I also tried: switchtype=national switchtype=dms100 context=pstn-pri group=1 signalling=pri_cpe channel = 1-23 ~~ Still struggling; thanks for any help and ideas. Alvin ___ --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] What is the state of Asterisk Secure Remote Communications?
Hello all, The wiki has a fairly detailed description of the the issues involved with encryption of Asterisk calls: http://www.voip-info.org/wiki/view/Asterisk+encryption I'm interested in hearing what is working for people today. I think the ideal solution would be a hard phone that could be plugged in almost anywhere (dsl/cable modem, hotel, etc) and connect securely to a remote Asterisk server (both for signalling and the RTP media stream). This might be a standalone phone, or maybe one plugged into a small (broadband router sized) box. An example commercial phone system with this capability is the Mitel 3300 or SX-200 with 5xxx IP phones having teleworker capability. What solutions just work out there? (Just work means that the end user only has to know enough to plug stuff in to get a dial-tone and incoming calls). All ideas (commercial or otherwise) welcome. Thanks, Alvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Boost Polycom IP601 headset volume
Hi everyone, I have a user that needs a little extra volume on his Polycom IP 601 phone set for all calls (beyond what the volume control currently offers). Is there a provisioning setting for this anywhere? (I'd like to avoid a separate amplifier between the phone and handset if possible.) Alternatively, is there a way to have Asterisk 1.4.x boost the volume to a particular SIP device for all calls? Thanks for any ideas! Alvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 601 - To not make noise when there is VM
Google: polycom mwi beep -- http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio The solution given works for me... Alvin Dovid B wrote: Hi Guys, I have some Polycom 601's here. It is super annoying that the phone every so often beeps to let me know that I have a VM. Is there any way to turn that off ? (I just want the red led to blink that there is a VM). Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Add current caller to junk-callers-database
Hello, I'm wondering how one might set up a feature to add (in real-time) the current CallerID information to a junk-callers database. After answering a call from an outside line and determining that the call was from a telemarketer or the like, the user could dial an easy specific code (like ** or 77), which would cause the call to be transferred to a specific extension within the Asterisk dialplan, where the CallerID info would be added to the database, a recorded message played to the caller, and then the call terminated. When that CallerID phoned again, the call would be diverted to voicemail or whatever automatically (instead of ringing). Another specific extension could be used to add/delete entries from the database if necessary. Any suggestion on how to enable a feature code to do the initial transfer? Thanks, Alvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial question
Hall, Eric M. wrote:
Not sure why this works
exten = _3665[0-9],1,goto(test|${EXTEN}|1)
but this does not.
exten = _366[50-59],1,goto(test|${EXTEN}|1)
I would like to route 36650 – 36700 to a Context ‘test’ however I’m only
able to get 10 to work at a time. Any ideas?
The square brackets allow for matching single digits. See:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
How about trying:
exten = _366[5-9][0-9],1,goto(test|${EXTEN}|1)
exten = 36700,1,goto(test|${EXTEN}|1)
The first matches extensions 36650 thru 36699, the second picks up the
remaining extension 36700.
Alvin
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[asterisk-users] Asterisk outbound calling does not wait for answer before playback
Hello Asteriskers, :-)
We're trying to set up an outbound notification calling for system
alerts with Asterisk 1.4.0. We generate a call file in
/var/spool/asterisk/outgoing and the outbound call is originated through
Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that
Asterisk does not wait for the other side to answer before it starts
playing the message. So the person called answers the phone after the
second or third ring and only hears the tail end of the message and the
goodbye.
Ideally, we want to deliver the message immediately after the person
answers, or if an answering machine picks up, right after the beep.
Any suggestions?
(1) The call file generator script (works ok):
#!/bin/sh
TMPFILE=`mktemp /tmp/tmp.XXX` || exit 1
echo TMPFILE = $TMPFILE
cat EOT $TMPFILE
Channel: Zap/g1/phone_number_here
Callerid: SYSTEM
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: dialout
Extension: s
Priority: 1
EOT
mv -v $TMPFILE /var/spool/asterisk/outgoing
(2) The dialout context in extensions.conf (problem - starts playback
before call is answered)
[dialout]
exten = s,1,NoOp(Dialout)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=8)
exten = s,n,Set(MACHINE=0)
exten = s,n,Answer
exten = s,n,BackgroundDetect(silence/5,1000,50)
exten = s,n,NoOp(Ans Machine detected)
exten = s,n,Set(MACHINE=1)
exten = s,n,BackgroundDetect(silence/30,1000,50,30050)
exten = s,n,NoOp(Ans Machine Message Too Long)
exten = s,n,Hangup
exten = talk,1,GotoIf($[${MACHINE}=1]?machine:human)
exten = talk,2(machine),Goto(dialout-machine,s,1)
exten = talk,3(human),Goto(dialout-human,s,1)
[dialout-machine]
exten = s,1,NoOp(Dialout to Ans Machine)
exten = s,n,Playback(/tmp/asterisk-recording)
exten = s,n,Wait(1)
; we'd like to do something to wait for the beep here...
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
[dialout-human]
exten = s,1,NoOp(Dialout to Human)
exten = s,n,Playback(/tmp/asterisk-recording)
exten = s,n,Wait(1)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
(3) *CLI
-- Attempting call on Zap/1/1234567 for [EMAIL PROTECTED]:1 (Retry 1)
Channel Zap/1-1 was answered.
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, Dialout) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(Zap/1-1, TIMEOUT(digit)=5) in new
stack
-- Digit timeout set to 5
-- Executing [EMAIL PROTECTED]:3] Set(Zap/1-1, TIMEOUT(response)=8) in
new stack
-- Response timeout set to 8
-- Executing [EMAIL PROTECTED]:4] Set(Zap/1-1, MACHINE=0) in new stack
-- Executing [EMAIL PROTECTED]:5] Answer(Zap/1-1, ) in new stack
(Problem: Asterisk does not wait until the call is answered on the far end!)
-- Executing [EMAIL PROTECTED]:6] BackgroundDetect(Zap/1-1,
silence/5|1000|50) in new stack
-- Playing 'silence/5' (language 'en')
-- Executing [EMAIL PROTECTED]:1] GotoIf(Zap/1-1, 0?machine:human)
in new stack
-- Goto (dialout,talk,3)
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, dialout-human|s|1)
in new stack
-- Goto (dialout-human,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, Dialout to Human)
in new stack
-- Executing [EMAIL PROTECTED]:2]
Playback(Zap/1-1,/tmp/asterisk-recording) in new stack
-- Playing '/tmp/asterisk-recording' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:4] Playback(Zap/1-1, vm-goodbye)
in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing [EMAIL PROTECTED]:5] Hangup(Zap/1-1, ) in new stack
== Spawn extension (dialout-human, s, 5) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
[Feb 8 13:29:37] NOTICE[32512]: pbx_spool.c:351 attempt_thread: Call
completed to Zap/1/1234567
Thanks for any ideas on this!
Alvin
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Re: [asterisk-users] DB_DELETE Function in 1.4
Jeremiah Millay wrote:
Does anyone know what application I should place this function in? For
example with the DB function I currently do something like this to add
an entry to the asterisk database:
exten = s,n,Set(DB(AGENT/${MACRO_EXTEN:1})=${CALLERID(num)})
To delete the entries I do something like this:
exten = s,n,DBDel(AGENT/${MACRO_EXTEN:1})
DBDel is marked as deprecated in favor of the DB_DELETE function but it
returns a warning when using it with a dialplan application like Set:
exten = s,n,Set(DB_DELETE(AGENT/${MACRO_EXTEN:1}))
Will return:
-- Executing [EMAIL PROTECTED]:202] Set(SIP/2146-b6f09f30,
DB_DELETE(AGENT/2109)) in new stack
[Jan 23 16:51:24] WARNING[4010]: pbx.c:5827 pbx_builtin_setvar: Ignoring
entry 'DB_DELETE(AGENT/2109)' with no = (and not last 'options' entry)
and it doesn't delete the database entry.
Would DB_DELETE work in an application like NoOp? Just wondering if
anyone has any experience using this new function in 1.4.0.
Thanks,
Jeremiah
Online (CLI) reference:
*CLI core show function DB_DELETE
-= Info about function 'DB_DELETE' =-
[Syntax]
DB_DELETE(family/key)
[Synopsis]
Return a value from the database and delete it
[Description]
This function will retrieve a value from the Asterisk database
and then remove that key from the database. DB_RESULT
will be set to the key's value if it exists.
So here's what you do to delete a database entry in 1.4.0:
exten = s,n,Set(oldval=${DB_DELETE(AGENT/${MACRO_EXTEN:1})})
; saves the old value of that key (in your case the callerid)
; into ${oldval} and deletes it from the DB. You can look at
; the value for the key you just deleted.
exten = s,n,NoOp(oldval : ${oldval})
Have fun!
Alvin
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Re: [asterisk-users] Polycom Power Specs
FWIW, our Polycom IP601 phones use a transformer with output: 24VDC 500mA (center contact is positive). A Polycom reseller (or Polycom sales) could probably give you information on these other two models. Alvin Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What's up with DATETIME and TIMESTAMP in Asterisk 1.4beta3 ?
Hello,
In Asterisk 1.4 beta 3, the UPGRADE.txt file says:
Variables:
* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP},
${ACCOUNTCODE},
and ${LANGUAGE} have all been deprecated in favor of their related
dialplan
functions. You are encouraged to move towards the associated dialplan
function, as these variables will be removed in a future release.
However, neither the function or application for either of TIMESTAMP or
DATETIME seems to work in 1.4beta3...
exten = *333,1,NoOp(DATETIME() : ${DATETIME()})
exten = *333,n,NoOp(DATETIME : ${DATETIME})
exten = *333,n,NoOp(TIMESTAMP() : ${TIMESTAMP()})
exten = *333,n,NoOp(TIMESTAMP : ${TIMESTAMP})
Asterisk 1.2.9.1:
-
Dec 15 12:56:26 ERROR[26373]: pbx.c:1383 ast_func_read: Function
DATETIME not registered
-- Executing NoOp(channel, DATETIME() : 0) in new stack
-- Executing NoOp(channel, DATETIME : 20061215-12:56:26) in
new stack
Dec 15 12:56:26 ERROR[26373]: pbx.c:1383 ast_func_read: Function
TIMESTAMP not registered
-- Executing NoOp(channel, TIMESTAMP() : 0) in new stack
-- Executing NoOp(channel, TIMESTAMP : 20061215-125626) in new
stack
Asterisk 1.4.0-beta3:
-
[Dec 15 13:59:52] ERROR[28236]: pbx.c:1497 ast_func_read: Function
DATETIME not registered
-- Executing [*333@context:1] NoOp(channel, DATETIME() : )
in new stack
-- Executing [*333@context:2] NoOp(channel, DATETIME : ) in
new stack
[Dec 15 13:59:52] ERROR[28236]: pbx.c:1497 ast_func_read: Function
TIMESTAMP not registered
-- Executing [*333@context:3] NoOp(channel, TIMESTAMP() : )
in new stack
-- Executing [*333@context:4] NoOp(channel, TIMESTAMP : ) in
new stack
Any ideas?
Thanks,
Alvin
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[asterisk-users] Polycom dealers in Toronto/London ON
Hello, Any recommendations on Polycom Soundpoint IP601 dealers in the Toronto / London ON areas? Thanks, Alvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting Asterisk to work with GoogleTalk
Hello all, We're trying to get the Asterisk to GoogleTalk functionality working, using the latest asterisk svn code (we've also tried with 1.4beta2). SVN Asterisk's make update displays: Updated to revision 59. Updated to revision 44477. We've tried to follow the recipe (without success) in: http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk When Asterisk starts up, the WindowsXP GoogleTalk user (xyz456) sees the asterisk server (ast123) appearance. When it tries to call the asterisk server, it hears ringing, but Asterisk does not answer (there is no indication in the CLI that it has received a call, except for the messages below). Asterisk (run as: asterisk -cfvv) shows the following messages several times: JABBER: googletalk INCOMING: iq to=[EMAIL PROTECTED]/asterisk709EC6B7 from=[EMAIL PROTECTED]/gmail.F1D1B5C9 id=c type=result query xmlns=http://jabber.org/protocol/disco#info; identity category=client type=pc/ feature var=http://jabber.org/protocol/disco#info/ /query/iq -- JABBER: I Dont have an IQ!!! JABBER: googletalk INCOMING: presence from=[EMAIL PROTECTED]/gmail.F1D1B5C9 to=[EMAIL PROTECTED]showaway/showpriority0/priority caps:c node=http://mail.google.com/xmpp/client/caps; ver=1.1 xmlns:caps=http://jabber.org/protocol/caps/ status/x xmlns=vcard-temp:x:updatephoto//x/presence -- JABBER: I am available ^_* 13 -- JABBER: type is away -- JABBER: I Do know how to handle presence!! Would anyone shed some light on what we're missing here, please? Here are the relevant configuration file pieces... (1) sip.conf [general] context=from-gtalk bindport=5060 bindaddr=0.0.0.0 srvlookup=yes dtmfmode=rfc2833 relaxdtmf=no disallow=all allow=ulaw allow=alaw allow=gsm maxexpirey=30 defaultexpirey=180 canreinvite=yes nat=0 UserAgent=Asterisk echocancel=yes echocancelwhenbridge=yes (2) gtalk.conf (this file is not present. Should it be??) (3) jabber.conf --- [general] ;debug=yes ;autoprune=yes ;autoregister=yes [googletalk] type=client serverhost=talk.google.com [EMAIL PROTECTED] secret=gtpass port=5222 ;port=5223 usetls=yes usesasl=yes [EMAIL PROTECTED] statusmessage=Voice Calls Only timeout=100 (4) jingle.conf --- [general] context=from-gtalk ;context=default allowguest=yes [guest] disallow=all allow=ulaw context=from-gtalk ;context=guest [google] [EMAIL PROTECTED] disallow=all allow=ulaw context=from-gtalk connection=asterisk (5) extensions.conf (partial): -- ;incoming from GoogleTalk [from-gtalk] exten = s,1,NoOP(Incoming call from GoogleTalk to [EMAIL PROTECTED]) exten = s,n,Answer() exten = s,n,Playback(thanks-for-calling) exten = s,n,Dial(SIP/101,60,t) exten = s,n,Hangup ;outgoing to GoogleTalk [to-gtalk] exten = 190,1,NoOp(Calling GoogleTalk user [EMAIL PROTECTED]) exten = 190,n,Dial(Jingle/googletalk/[EMAIL PROTECTED]) (note that [EMAIL PROTECTED] and [EMAIL PROTECTED] are fictitious names for debugging only) - If you have this working, please share your sanitized configuration files. - Can you explain the messages JABBER: I Dont have an IQ!!! and JABBER: I Do know how to handle presence!! and what's required to correct the problems. Thanks much, Alvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Getting Asterisk to work with GoogleTalk
Bromont - wrote: It should work fine with 1.4Beta2 I use gtalk.conf instead of jingle.conf and this is what I would change in configurations (shown with the arrows): jabber.conf: [general] ;debug=yes ;autoprune=yes ;autoregister=yes [googletalk] type=client serverhost=talk.google.com [EMAIL PROTECTED]/Talk -- secret=gtpass port=5222 usetls=yes usesasl=yes [EMAIL PROTECTED] statusmessage=Voice Calls Only timeout=100 gtalk.conf: [general] context=from-gtalk allowguest=yes [guest] disallow=all allow=ulaw context=from-gtalk [google] [EMAIL PROTECTED]-- disallow=all allow=ulaw context=from-gtalk connection=googletalk -- extensions.conf: ;outgoing to GoogleTalk [to-gtalk] exten = 190,1,NoOp(Calling GoogleTalk user [EMAIL PROTECTED]) exten = 190,n,Dial(gtalk/googletalk/[EMAIL PROTECTED])-- Thanks for the note. After switching to using gtalk.conf instead of jingle.conf, and making the adjustments above, I find that the svn version of Asterisk dies with a core dump and backtrace when the XP Google Talk client tries to call it: JABBER: googletalk INCOMING: iq to=[EMAIL PROTECTED]/Talk6DD03373 type=set id=65 from=[EMAIL PROTECTED]/Talk.v96D358DB9Dsession type=initiate id=2561394071 initiator=[EMAIL PROTECTED]/Talk.v96D358DB9D xmlns=http://www.google.com/session;description xml:lang=en xmlns=http://www.google.com/session/phone;payload-type id=103 name=ISAC clockrate=16000/payload-type id=97 name=IPCMWB clockrate=16000 bitrate=8/payload-type id=99 name=speex clockrate=16000 bitrate=22000/payload-type id=4 name=G723 clockrate=8000 bitrate=6300/payload-type id=98 name=speex clockrate=8000 bitrate=11000/payload-type id=100 name=EG711U clockrate=8000 bitrate=64000/payload-type id=101 name=EG711A clockrate=8000 bitrate=64000/payload-type id=0 name=PCMU clockrate=8000 bitrate=64000/payload-type id=8 name=PCMA clockrate=8000 bitrate=64000/payload-type id=13 name=CN clockrate=8000/payload-type id=102 name=iLBC clockrate= JABBER: googletalk INCOMING: 8000 bitrate=13300/payload-type id=106 name=telephone-event clockrate=8000//descriptiontransport xmlns=http://www.google.com/transport/p2p//session/iq *** glibc detected *** asterisk: munmap_chunk(): invalid pointer: 0xb795cc32 *** === Backtrace: = /lib/i686/libc.so.6(__libc_free+0x18a)[0xb7d4903a] /usr/lib/asterisk/modules/chan_gtalk.so[0xb78f7242] /usr/lib/asterisk/modules/chan_gtalk.so[0xb78f7816] /usr/lib/libiksemel.so.3(iks_filter_packet+0x129)[0xb7a587e9] /usr/lib/asterisk/modules/res_jabber.so[0xb7a794e6] /usr/lib/libiksemel.so.3[0xb7a569c4] /usr/lib/libiksemel.so.3(iks_parse+0x5a0)[0xb7a54a40] /usr/lib/libiksemel.so.3(iks_recv+0x98)[0xb7a56368] /usr/lib/asterisk/modules/res_jabber.so[0xb7a75110] asterisk[0x80e924b] /lib/i686/libpthread.so.0[0xb7f41540] /lib/i686/libc.so.6(__clone+0x5e)[0xb7dae55e] === Memory map: 08048000-08137000 r-xp 16:02 585805 /usr/sbin/asterisk 08137000-08143000 rwxp 000ef000 16:02 585805 /usr/sbin/asterisk 08143000-0822b000 rwxp 08143000 00:00 0 [heap] b6d8e000-b6d9 r-xp 16:02 1324089 /usr/lib/asterisk/modules/func_vmcount.so b6d9-b6d91000 rwxp 1000 16:02 1324089 /usr/lib/asterisk/modules/func_vmcount.so b6d91000-b6d92000 r-xp 16:02 1324088 /usr/lib/asterisk/modules/func_uri.so b6d92000-b6d93000 rwxp 1000 16:02 1324088 /usr/lib/asterisk/modules/func_uri.so b6d93000-b6d95000 r-xp 16:02 1324087 /usr/lib/asterisk/modules/func_timeout.so b6d95000-b6d96000 rwxp 1000 16:02 1324087 /usr/lib/asterisk/modules/func_timeout.so b6d96000-b6d9a000 r-xp 16:02 1324086 /usr/lib/asterisk/modules/func_strings.so b6d9a000-b6d9b000 rwxp 3000 16:02 1324086 /usr/lib/asterisk/modules/func_strings.so b6d9b000-b6d9c000 r-xp 16:02 1324085 /usr/lib/asterisk/modules/func_sha1.so b6d9c000-b6d9d000 rwxp 16:02 1324085 /usr/lib/asterisk/modules/func_sha1.so b6d9d000-b6d9f000 r-xp 16:02 1324084 /usr/lib/asterisk/modules/func_realtime.so b6d9f000-b6da rwxp 1000 16:02 1324084 /usr/lib/asterisk/modules/func_realtime.so b6da-b6da2000 r-xp 16:02 1324083 /usr/lib/asterisk/modules/func_rand.so b6da2000-b6da3000 rwxp 1000 16:02 1324083 /usr/lib/asterisk/modules/func_rand.so b6da3000-b6da4000 r-xp 16:02 1324082 /usr/lib/asterisk/modules/func_md5.so b6da4000-b6da5000 rwxp 16:02 1324082 /usr/lib/asterisk/modules/func_md5.so b6da5000-b6da7000 r-xp 16:02 1324081 /usr/lib/asterisk/modules/func_math.so b6da7000-b6da8000 rwxp 1000 16:02 1324081 /usr/lib/asterisk/modules/func_math.so b6da8000-b6daa000 r-xp 16:02 1324080 /usr/lib/asterisk/modules/func_logic.so b6daa000-b6dab000 rwxp 1000 16:02 1324080 /usr/lib/asterisk/modules/func_logic.so b6dab000-b6dad000 r-xp 16:02 1324079 /usr/lib/asterisk/modules/func_groupcount.so b6dad000-b6dae000 rwxp
[asterisk-users] Problem with Background DTMF detection with A200D
Hi all,
I'm having trouble with Background DTMF detection, and would appreciate
any suggestions.
A call comes in to a Sangoma A200D PSTN line. A standard menu welcome
is used. Most of the time, callers have to wait until the message
completes in order to have their selection recognized. People end up
having to press the option number several times. Occasionally, you can
press the desired option digit during the message and it will be
selected right away while the Background message is still playing (this
is what I want all the time). Any suggestions?
Environment: Asterisk 1.2.10, zaptel-1.2.7, wanpipe-beta7-2.3.4.tgz
Machine has lots of horsepower: Pentium D 3.2 GHz, 2 GB RAM,
[general]
priorityjumping=no
autofallthrough=no
(...)
[from-pstn]
; Inbound calls from PSTN line
exten = s,1,NoOp(TIMESTAMP: ${TIMESTAMP})
exten = s,2,NoOp(CONTEXT: ${CONTEXT})
exten = s,3,NoOp(CALLERIDNUM: ${CALLERIDNUM})
exten = s,4,NoOp(CALLERIDNAME: ${CALLERIDNAME})
exten = s,n,Goto(mainmenu,s,1)
[mainmenu]
exten = s,1,NoOp(Main Menu)
exten = s,n,Wait,1
exten = s,n,Answer
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Playback(silence-1sec)
exten = s,n,Playback(silence-1sec)
exten = s,n,Background(mainmenu)
; Thank you for calling xxx.
;Please press 1 for AA;
;2 for BB;
;3 for CC;
;or 4 for DD.
;Press 0, or stay on the line for reception.
exten = 1,1,NoOp(Menu 1 - Dialing SIP/101 AA)
exten = 1,n,Dial(SIP/101,20,t)
exten = 1,n,Playback(silence-1sec)
exten = 1,n,Voicemail(u101)
exten = 1,n,Hangup
exten = 2,1,NoOp(Menu 2 - Dialing SIP/102 BB)
exten = 2,n,Dial(SIP/102,20,t)
exten = 2,n,Playback(silence-1sec)
exten = 2,n,Voicemail(u102)
exten = 2,n,Hangup
exten = 3,1,NoOp(Menu 1 - Dialing SIP/103 CC)
exten = 3,n,Dial(SIP/103,20,t)
exten = 3,n,Playback(silence-1sec)
exten = 3,n,Voicemail(u103)
exten = 3,n,Hangup
exten = 4,1,NoOp(Menu 1 - Dialing SIP/104 DD)
exten = 4,n,Dial(SIP/104,20,t)
exten = 4,n,Playback(silence-1sec)
exten = 4,n,Voicemail(u104)
exten = 4,n,Hangup
exten = 0,1,NoOp(Menu 0 - Dialing SIP/100)
exten = 0,n,Dial(SIP/100,20,t)
exten = 0,n,Playback(silence-1sec)
exten = 0,n,Voicemail(u100)
exten = 0,n,Hangup
exten = #,1,NoOp(Menu # - Access VOICEMAIL)
exten = #,n,Playback(silence-1sec)
exten = #,n,VoiceMailMain()
exten = #,n,Hangup
;
exten = t,1,NoOp(Menu t - Goto mainmenu,0,1)
exten = t,n,Goto(mainmenu,0,1)
;
exten = i,1,NoOp(Menu i - Playback pbx-invalid)
exten = i,n,Playback(pbx-invalid)
exten = i,n,Goto(mainmenu,s,1)
;end of [mainmenu]
;
In the zapata.conf file, the relevant parts are:
[trunkgroups]
[channels]
language=en
context=default
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
cidsignalling=bell
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
musiconhold=default
echocancel=yes
echocancelwhenbridged=yes
rxgain=3.0
txgain=0.0
immediate=no
faxdetect=no
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel = 1
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel = 2
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel = 3
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel = 4
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel = 5
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel = 6
;---
Thanks for any ideas,
Alvin
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[Asterisk-Users] QUESTION: RINGING CONTINUES DURING CALL
After searching around, I've been unable to to find any relevant info on this. Perhaps the group can help? I am seeing something strange with a new Sipura SPA-3000 (and I've noticed this also with an IAX softphone): When I dial 777, this dialplan (in extensions.conf) is run: exten = 777,1,Dial(Zap/1/2345678) exten = 777,n,Hangup The number is answered by the called party, but the ringing sound continues and is heard over top of the conversation. If I add an Answer line to the dialplan, this problem disappears: exten = 777,1,Answer exten = 777,n,Dial(Zap/1/2345678) exten = 777,n,Hangup This does not occur with a Grandstream BT-101 or an XTen SIP softphone. Why do some devices need the Answer line in the dialplan before the Dial line? (I'd rather not have to do a custom section for certain devices). Thanks for any info... Alvin ___ --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] Forget Asterisk@Home 0.7 :-) :-) 0.8 is out
0.8 appears to have been released. Start with that. It is very quick to update your current 0.6 or 0.7 iso. Just do this with rsync to do a differential copy: $ mv asteriskathome-0.7.iso asteriskathome-0.8.iso $ rsync -av --progress --partial \ prdownloads.sourceforge.net::sourceforge/a/as/asteriskathome/asteriskathome-0.8.iso . $ rsync -av \ prdownloads.sourceforge.net::sourceforge/a/as/asteriskathome/asteriskathome-md5sum.txt . $ grep iso asteriskathome-md5sum.txt $ md5sum asteriskathome-0.8.iso (The output of both should be the same...) Have fun! Alvin ___ 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] CallerID for incoming SIP calls to Asterisk connected phone
Hello all,
I'm having a problem with getting incoming callerid to a lan-connected
phone.
The Asterisk server is connected to the Internet, and a Grandstream
BT101 phone on a lan interface:
INTERNET (eth0) Asterisk (eth1) Grandstream (192.168.1.51)
The phone registers with the Asterisk server as ext 20.
I can initiate and receive calls from the Grandstream phone fine.
The Asterisk server has a sipphone.com registered account. When a SIP
call comes in from outside, the call completes fine, but the phone
always shows the telephone number of my Asterisk server, not the calling
party's SIP number. What's wrong?
What I really want is that for inbound calls, I see the callerid of the
SIP phone initiating the call.
Here are the (hopefully) relevant parts in the config files...
In sip.conf:
---
register = 1747xxx:[EMAIL PROTECTED]/1747xxx
[sipphone]
context=from-sip-external
type=friend
secret=sip_password
username=1747xxx
;host=proxy01.sipphone.com
host=198.65.166.131
callerid=My Name 1747xxx:
qualify=no
reinvite=no
canreinvite=no
insecure=very
[20]
context=from-sip-internal
type=friend
callerid=20
username=20
mailbox=20
secret=
host=dynamic
defaultip=192.168.1.51
canreinvite=no
dtmf=info
dtmfmode=rfc2833
;
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
In extensions.conf:
--
[globals]
TRUNK=Zap/1 ; FXO interface
SIPPHONEUSERID=1747xxx
[from-sip-external]
exten = ${SIPPHONEUSERID},1,SetCIDName(SIP - ${CALLERIDNAME})
exten = ${SIPPHONEUSERID},2,Dial(SIP/20,15)
exten = ${SIPPHONEUSERID},3,Goto(mainmenu,s,1)
exten = ${SIPPHONEUSERID},4,Hangup
...
Any suggestions/help would be greatly appreciated.
Thanks,
Alvin
PS: Please cc me directly on replies:
a a n (at) crlogic (dot) com
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