[asterisk-users] Manager Originate and Callerid ?
I'm using Manager API Originate to initiate calls from SIP channels (via phpagi FWIW) and it all works well except ...the CallerID for the SIP channel specified in users.conf isn't set for the call :-( If I explicitly set the Callerid in the Manager Originate API call then it works but the API is actually being run from another server which doesn't 'know' the correct Callerid number and name for any given SIP phone so can't set them. I'm calling the Manager API with the following:- Action: Originate Channel: SIP/101 Context: from-sip Exten: 01234567890 Priority: 1 Timeout: 2 ActionID: foo This results in the Callerid(name) and Callerid(num) being blank for the call. The 'from-sip' context is exactly the same as my SIP phones are using and when manually dialing the Callerid info is correctly picked up from users.conf. Any ideas why this is and how I can get the Manager API Originate call to use the correct Callerid info? -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | -- _ -- 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] DAHDI has broken by pbx :-(
On 11/10/2011 09:38 AM, Russell Brown wrote: I copied /etc/asterisk/zapata.conf to /etc/chan_dahdi.conf Did you really copy it to /etc/chan_dahdi.conf or to /etc/asterisk/chan_dahdi.conf ? Yes it was to /etc/asterisk/chan_dahdi.conf (oh how I wish the problem was being caused by something that simple!). Also, What is the output of 'module reload chan_dahdi'? asterisk*CLI module reload chan_dahdi.so -- Reloading module 'chan_dahdi.so' (DAHDI Telephony Driver w/PRI) asterisk*CLI I know it's loading as Span 1 is working; making and taking calls with no problem. FWIW, asterisk*CLI dahdi show status Description Alarms IRQbpviol CRC Fra Codi Options LBO T2XXP (PCI) Card 0 Span 1OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T2XXP (PCI) Card 0 Span 2OK 0 0 46 CCS HDB3 CRC4 266-399 feet (DSX-1) asterisk*CLI I've, obviously, been doing more trial-and-error and noticed that when I run with libpri-1.4.11.5 (no other change, just install libpri and restart asterisk) the ISDN causecode returned from a call on Span 2 is actually 21 and not 18, as it is with libpri-1.4.12, and the channels on span 2 don't get reset each time. Weird. This is driving me loopy as the 'old' ISDN PBX on span 2 is used as a glorified 'channel bank' with fax machines, modems and other analogue stuff on it that really does need to work :-( -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | -- _ -- 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
[asterisk-users] Missed calls and groups
Is there a SIP header I can set (for Snom and Yealink phones if that's relevant) or any other mechanism to tell a phone to ignore a particular call from it's missed call list? I have bits of the dialplan that ring groups of phones eg: exten = 200,1,Dial(Sip/112SIP/113SIP/114) and I don't want such calls being recorded by the phone as a missed call. Calls to the specific phone I do want displayed so just disabling the Missed Calls feature on the phone doesn't cut the mustard. Ideas? (I'd also want this to work with Queues but let's see about the basics first) -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | -- _ -- 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] [OT] Yealink Phones
Quoth Andrew Thomas:- Have you seen the 'Action URL' bit yet? Makes everything almost key-system like ;) I saw it in the DSS key settings but havn't worked out anything useful to do with it yet? What are you using it for (and how?)? -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | -- _ -- 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
[asterisk-users] Paging a message. How?
I'm scratching my head trying to work out a way of sending a pre-recorded message as a 'Page' to a list of phones ( Oi! you muppets you've left the server room door open! or somesuch message :-) controlled by an external trigger. I can do a normal page (phones auto-answer on speaker) with SipAddHeader but that doesn't let me play a pre-recorded message. Any suggestions? -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | -- _ -- 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
[asterisk-users] ControlPlayback skip forward fails on mp3 file
Using Asterisk 1.4.31 and addons 1.4.11, ControlPlayback get confused
when skipping forwards on an mp3 file (it seems to work fine on wav's).
I'm calling it from an AGI like so:
$agi-exec('ControlPlayback',$filename . |4000|#|*|8|0|7);
The first four times I press the '#' key it does indeed skip forwards;
but the fifth and subsequent times pressing '#' makes it skip
*backwards*.
If I restart from the beginning (by pressing '7') it again skips
forwards four times then starts going backwards.
The MP3's are created by:
/usr/local/bin/lame --quiet -h -b16 --noshort --preset phone recording.wav
recording.mp3
Any Ideas or should I report it as a bug.
--
Regards,
Russell
| Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] BT ISDN-30 Call Failures
) len=9 [Mar 17 17:04:46] VERBOSE[13006] logger.c: Call Ref: len= 2 (reference 79/0x4F) (Terminator) [Mar 17 17:04:46] VERBOSE[13006] logger.c: Message type: ALERTING (1) [Mar 17 17:04:46] VERBOSE[13006] logger.c: [1e 02 81 88] [Mar 17 17:04:46] VERBOSE[13006] logger.c: Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) [Mar 17 17:04:46] VERBOSE[13006] logger.c: Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] [Mar 17 17:04:46] VERBOSE[13006] logger.c: -- Executing [846...@isdn_in:2] Set(Zap/20-1, CALLERID(name)=Banana) in new stack snip [Mar 17 17:04:48] VERBOSE[13006] logger.c: -- Executing [...@macro-stdexten:38] Dial(Zap/20-1, Sip/192|15|wWtTkK) in new stack [Mar 17 17:04:48] VERBOSE[13006] logger.c: -- Called 192 [Mar 17 17:04:48] VERBOSE[13006] logger.c: -- SIP/192-02980aa0 is ringing [Mar 17 17:04:49] VERBOSE[13006] logger.c: -- SIP/192-02980aa0 is ringing [Mar 17 17:04:50] VERBOSE[13006] logger.c: -- SIP/192-02980aa0 is ringing [Mar 17 17:04:52] VERBOSE[13006] logger.c: -- SIP/192-02980aa0 is ringing [Mar 17 17:04:55] VERBOSE[13006] logger.c: == Spawn extension (macro-stdexten, s, 38) exited non-zero on 'Zap/20-1' in macro 'stdexten' [Mar 17 17:04:55] VERBOSE[13006] logger.c: == Spawn extension (macro-TELESALESUSER, s, 3) exited non-zero on 'Zap/20-1' in macro 'TELESALESUSER' [Mar 17 17:04:55] VERBOSE[13006] logger.c: == Spawn extension (real_isdn_in, 846092, 1) exited non-zero on 'Zap/20-1' [Mar 17 17:04:55] VERBOSE[13006] logger.c: -- Executing [...@real_isdn_in:1] Hangup(Zap/20-1, ) in new stack [Mar 17 17:04:55] VERBOSE[13006] logger.c: == Spawn extension (real_isdn_in, h, 1) exited non-zero on 'Zap/20-1' [Mar 17 17:04:55] VERBOSE[13006] logger.c: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request *** Who requested this disconnect? Any ideas? [Mar 17 17:04:55] VERBOSE[13006] logger.c: q931.c:2967 q931_release: call 79 on channel 20 enters state 19 (Release Request) [Mar 17 17:04:55] VERBOSE[13006] logger.c: Protocol Discriminator: Q.931 (8) len=9 [Mar 17 17:04:55] VERBOSE[13006] logger.c: Call Ref: len= 2 (reference 79/0x4F) (Terminator) [Mar 17 17:04:55] VERBOSE[13006] logger.c: Message type: RELEASE (77) [Mar 17 17:04:55] VERBOSE[13006] logger.c: [08 02 81 e6] [Mar 17 17:04:55] VERBOSE[13006] logger.c: Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) [Mar 17 17:04:55] VERBOSE[13006] logger.c: Ext: 1 Cause: Recover on timer expiry (102), class = Protocol Error (e.g. unknown message) (6) ] *** Ummm... 'Protocol Error' doesn't sound nice! [Mar 17 17:04:55] VERBOSE[13006] logger.c: -- Hungup 'Zap/20-1' Any Ideas? Thanks in Advance. -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | -- _ -- 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
[asterisk-users] ISDN30 Timing Sources (Jon Morgan)
Quoth Jon Morgan jon.mor...@motors.co.uk We have a 2 port Digium TE220P card, one span is configured to connect to our ISDN30 provider (British Telecom), the other span connects to our internal PBX. Here's the zaptel.conf snip: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 FWIW, I (also BT ISDN30 on span 1 with a PBX on the second port of a TE205P) have the following zaptel.conf. span=1,1,1,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,1,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 I do all my call recording in asterisk so can't comment on that but the PBX users are not complaining about the quality. -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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] UK Vodafone messaging, ISDN, Wrong CallerID being used.
I've a strange problem with CallerID when calling Vodafone mobile's from my Asterisk Box. If I dial out on my ISDN-30, setting the CallerID to my DDI (XXX802), a Vodafone mobile correctly shows the incoming call with this number and if the phone's not answered it shows a missed call from XXX802. All very good. However, if the call is not answered and goes to the Vodafone messaging service and the Vodafone user then uses the 'call back' option, they end up ringing back on XXX800 (which happens to be the lowest number in our range and the billing number for our ISDN-30). How are Vodafone getting *TWO* CallerID numbers for the same call? Anyone know a way I can stop this happening? TIA. -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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] Howto change CDR record on calling channel from called thread?
I'm tearing (what's left of) my hair out on this one :-(
shortform
How can I set the CDR(userfield) in the calling thread from the dialplan
(actually a macro called from a feature) in the called thread?
long version
I use mixmonitor to record calls driven by entries in the asterisk
database for selected phones. As part of this dialplan, I set the
CDR(userfield) to the mixmonitor filename. I later (in a small web app)
use this userfield to detect recorded calls, get the filename and play
them back.
This all works very very well when the recording is started and the
CDR(userfield) is set by the originating channel.
However, I would like to enable recording from a feature code (*1) on an
ad-hoc basis for those phone that are not set to autorecord and set the
corresponding CDR(userfield) so my web app can spot recorded calls.
This again works well *if* the feature code is used on the calling
channel. However, if it's on the *called* channel then the
CDR(userfield) doesn't get updated because AFAIUI the CDR record is all
in the calling thread.
Can anyone tell me how I can set the userfield for the CDR record in the
*called* thread?
I've tried setting variables (__RECORDINGFILENAME) and setting
CDR(userfield)=${RECORDINGFILENAME} in the hangup code but this again
only works if the feature is initiated by the calling thread.
Putting aside my userfield for filenames requirement, I'd have thought
that the ability to set the CDR(accountcode) via a feature on the called
channel would have been done before... but I've googled to no avail on
that either.
Can anyone help?
--
Regards,
Russell
| Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] Setting CDR(userfield) from Macro called from feature doesn't work with cdr_mysql
cdr_mysql doesn't set the userfield when it's set inside a macro
called from a feature (1.4.25, addons 1.4.8).
I have a feature code:
autorecord = *1,self,Macro,apprecord
The apprecord macro looks like:
[macro-apprecord]
exten = s,1,Playback(beep)
exten =
s,n,Set(RECORDFILE=/var/spool/asterisk/autorecord/${STRFTIME(${EPOCH},,%Y/%m/%d/%H%M%S)}-${UNIQUEID}-^-${CALLERID(num)})
exten = s,n,Set(CDR(userfield)=${RECORDFILE})
exten = s,n,MixMonitor(${RECORDFILE}.wav)
exten = s,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = s,n,NoOp(CDR(userfield) = ${CDR(userfield)})
exten = s,n,MacroExit
The NoOp shows the userfield is set correctly but the userfield is blank
in my MySQL cdr database. I set CDR(userfield) elsewhere in the dialplan
and this works so it seems to be related to being set within a macro.
Any idea what I'm doing wrong?
--
Regards,
Russell
| Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] BT ISDN-30 Pri getting 'stuck' on outgoing calls.
, circuit-mode (16) User information layer 1: A-Law (35) [18 03 a1 83 87] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Preferred Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 7 ] [1e 02 84 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 21 83 31 32 33 39 36 35 34 35 32 36] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '1239654526' ] [70 07 81 38 34 36 30 37 37] Called Number (len= 9) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '846077' ] -- Making new call for cr 2 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) q931.c:3545 q931_receive: call 2 on channel 7 enters state 6 (Call Present) Sending Receiver Ready (49) [ 02 01 01 62 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 049 P/F: 0 0 bytes of data -- Restarting T203 timer q931.c:2810 q931_call_proceeding: call 2 on channel 7 enters state 9 (Incoming Call Proceeding) Delaying transmission of 77, window is 7/7 long Stopping T_203 timer Starting T_200 timer -- Restarting T200 timer Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 2/0x2) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 87] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 7 ] q931.c:2838 q931_alerting: call 2 on channel 7 enters state 7 (Call Received) Delaying transmission of 78, window is 7/7 long Starting T_200 timer -- Restarting T200 timer Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 2/0x2) (Terminator) Message type: ALERTING (1) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending q931.c:3009 q931_disconnect: call 33612 on channel 26 enters state 11 (Disconnect Request) Delaying transmission of 79, window is 7/7 long Starting T_200 timer -- Restarting T200 timer Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 844/0x34C) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] If you've got this far then Thanks! and well done! Any help much appreciated. -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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] Call Stealing
Quoth Geoff Lane ge...@gjctech.co.uk... AIUI, you need to set up the conference before leaving the extension on which you took the call. Yes you do. You'd need to explicitly send the call to a conference, listen and remember the conference number. FWIW, Call Stealing is a feature I miss from my Argent PBX :-( It was nice to wander off and be able to grab an existing call to my extension from any phone that I picked up. I've not been able to find a way of doing this in Asterisk. -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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] CDR Rewrite -- Questions to the users (Steve Murphy)
Quoth Steve Murphy... Date: Mon, 12 Jan 2009 08:51:03 -0700 QUESTIONS: Which of the two would you see being useful to you? Obvious comment really but given LEG based CDR, one can determine the 'Simple' data but you can't work it the other way. I'd therefore find LEG based CDR more useful as the granularity (time on Hold, in Queue, Waiting on pre-xfer ring etc etc) would be good. -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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] Asterisk Generating NetworkOOO (ISDN Cause Code 38)
I have a legacy ISDN PBX (Network Alchemy Argent Office) connected to Span 2 of a Digium Wildcard TE205P. Recently Calls from this PBX have been failing with ISDN Cause Code 38 (Network Out of Order!). The problem seems to be getting worse and is now effecting more calls than not (although this could just be because I'm aware of it). Once the ISDN PBX has decided the Network's Out of Order and torn down the call, the Asterisk box still has the channels bridged and up. Can anyone suggest a course of action I might take to getting this sorted? Do I need to change something in zapata.conf? (included below). Here's the log from the ISDN PBX showing the call failing: 2592754mS CMCallEvt:v=1129 State, new=Ringing old=Dialled,0,0,BState 2592800mS CMCallEvt:v=1129 State, new=Ringing old=Dialled,0,0,Astate 2595647mS ISDNL3Evt: v=0 stacknum=0 State, new=Active, old=Delivered id=1130 2595657mS CMLineRx: v=1 CMConnect Line: type=Q931Line 1 Call: lid=0 id=1130 in=0 BChan: slot=0 chan=16 2595657mS CMCallEvt:v=1129 State, new=Connected old=Ringing,0,0,BState 2595729mS CMCallEvt:v=1129 State, new=Connected old=Ringing,0,0,Astate 2596653mS ISDNL3Evt: v=0 p1=0,p2=1001,p3=5,p4=0,s1= 2596654mS ISDNL3Evt: v=0 stacknum=0 State, new=NullState, old=Active id=1130 2596659mS ISDNL3Evt: v=0 p1=0,p2=1000,p3=0,p4=0,s1= 2596665mS CMLineRx: v=1 CMReleaseComp Line: type=Q931Line 1 Call: lid=0 id=1130 in=0 Cause=38, NetworkOOO 259mS CMCallEvt:v=1129 State, new=Idle old=Connected,0,0,Astate 2596667mS CMCallEvt:v=1129 State, new=Idle old=Connected,0,0,BState 2596670mS CALL:2009/01/0612:04,00:00:01,002,1780471800,O,0173323,0173323,Modem0,,,0 but Asterisk thinks the call is up and bridged: asterisk*CLI show channels Channel Location State Application(Data) Zap/1-1 (None) Up Bridged Call(Zap/46-1) Zap/46-1 0173323x...@alchemy: Up Dial(Zap/g1/0173323|) 2 active channels 1 active calls asterisk*CLI Here's the relevant bit from zaptel.conf: # Second port span=2,0,1,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 # # Global data loadzone=uk defaultzone=uk ...and here's the bit from my zapata.conf for Span 2. ; ; Network Alchemy ; group = 2 switchtype=euroisdn context = alchemy usecallerid=yes signalling = pri_net resetinterval=1 callerid=asreceived pridialplan=unknown prilocaldialplan=unknown useincomingcalleridonzaptransfer=yes channel = 32-46 Can anyone help? -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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] Dynamic Feature Playback acting on *both* channels?
I'd like to be able to playback a file to *both* channels in a call as a result of a DTMF feature. Can anyone suggest how I might do this? I thought of using a DYNAMIC_FEATURE to call a macro that starts a dynamic meetme but the macro only gets to control the 'caller' or 'callee' :-( Failing that I'm trying to provide a simple means of playing back a recorded message during a phone call controled by someone's phone (the actual message will have been pre-selected by an external app). Any suggestions? -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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] Pre-routing manipulation of calls
Quoth Kevin DeGraaf ke...@kdegraaf.net
I want to do some manipulation (CallerID name override) to all incoming
calls before they are routed. I would prefer to avoid duplicating the
necessary code in each DID extension stanza, even if it's just a call to
a macro.
I do it like so:
[isdn_in]
exten = _X.,1,Set(CALLERID(name)=Banana)
;
exten = _X.,n,Goto(real_isdn_in,${EXTEN},1)
;
[real_isdn_in]
;
; DDI planning
;
exten = _871800,1,Goto(groups,200,1)
exten = _871802,1,Macro(stdexten|112|Sip/112)
exten = _871803,1,Macro(stdexten|113|Sip/113)
exten = _871804,1,Macro(stdexten|114|Sip/114)
;...
HtH
--
Regards,
Russell
| Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] Howto grab back call transfered from SIP phone
Once in a while, someone mis-dials when transfering a call on their Snom SIP phone (using the Transfer button). Instead of sending them to, say, 1940; they mistype and enter 194 or 190 or somesuch. This ends up on the PSTN (for which three digit calls are valid); not what anyone wanted. On our old PBX (Network Alchemy Argent Office) there was a dialcode that grabbed back the last call that went through your extension - very useful when you realised what you'd done. Is there any way of programming this in Asterisk? I've googled to no avail :-( -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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-users Digest, Vol 51, Issue 26
Quoth Jared Smith... On Wed, 2008-10-08 at 18:51 +0100, Russell Brown wrote: Can anyone point me at a code fragment (C would be nice) that I could use to subscribe to hints on a * box? I'd like to write a small (hopefuly efficient) widget to show custom device states and believe that a subscription to the hint would be the most efficient but I'm very open to suggestions. Maybe I'm missing something here, but wouldn't it be a whole lot easier to get this information via the Asterisk Manager Interface, rather than writing a C program that knows how to parse SIP messages? Yes I could do it through the Manager and a polling loop but wouldn't that be somewhat inefficient? By using the SIP Hints I was hoping that my code would only be told when a hint changed. Any pointers anyone? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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] Sample code fragement for subscribing to hints wanted.
Can anyone point me at a code fragment (C would be nice) that I could use to subscribe to hints on a * box? I'd like to write a small (hopefuly efficient) widget to show custom device states and believe that a subscription to the hint would be the most efficient but I'm very open to suggestions. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra Park Softkey
Quoth: OCG Technical Support [EMAIL PROTECTED] Although we've programmed the softkeys per the manuals, they seem to have no effect (just dead). For example, our 57i is setup like this: I had similar problems and ended up using the speeddial inband functionality. FWIW, my 57i's setup like so: softkey4 type: speeddial softkey4 label: *Park softkey4 value: #,700 softkey4 line: 1 softkey4 states: connected So the bottom left soft key does and says Park when connected. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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] Lamps on Snom phones
Quoth Phil Knighton [EMAIL PROTECTED] I've incorporated the kind responses from other list members, such as setting call limits but to no avail! I've checked the function key settings on the Snom, and adjusted it to match the suggestion from another list member - nothing. Not one lamp on any of the phones will work, and I'm completely baffled as to why. A wild stab in the dark what version of the Snom firmware are you running? The lamps work for me on a Snom 370 running 7.1.28 and worked on other 7.1.last few releases but can't remember how far back. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken
Quoth robert boardman [EMAIL PROTECTED]
Tzafrir Cohen wrote:
On Wed, Jan 02, 2008 at 07:41:40PM +, Russell Brown wrote:
Something in the, fairly, recent series of Asterisk updates has broken
DIGITAL call passthrough.
I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a
Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover
cable).
This PBX used to be able to make and receive DIGITAL type ISDN calls
through the Asterisk box... but something in the latest generation of
updates has broken it and although the calls seem to work the old PBX
just won't route traffic. Voice calls still work fine.
I've proven it's something in Asterisk by connecting the old PBX
directly to our ISDN PRI line and it still works fine.
What version is good? What version is bad?
Well I've had a fun few hours testing versions and eventually found out
what brings the problem to light.
I went back through versions of Asterisk et al until I got bored and
reinstated a complete backup from last August onto my Asterisk box
Voila! it worked (for inbound calls anyway).
Working my way forward in time... I eventually discounted all the
Asterisk, Zaptel and Libpri versions and boiled it down to me having
DYNAMIC_FEATURES=automon#autorecord#testfeature1
in the [globals] section of extensions.conf.
If I change this to DYNAMIC_FEATURES=automon then incoming DIGITAL calls
work. If DYNAMIC_FEATURES has anything more than this then it doesn't.
As a workaround, I've now got:
exten = _X.,n,Set(DYNAMIC_FEATURES=)
exten = _X.,n,Dial(Zap/g2/${EXTEN})
in the forward-to-my-old-PBX bit of the dialplan.
This works with 1.4.17, Zaptel 1.4.7.1 and libpri 1.4.3 (the current
stuff).
I have an outstanding problem with this,I have found that if you set
overlapdial to no on the internal leg ie connected to the pabx it works,
but you will have to set the pabx to dial en-block ie send all digits at
once
WRT Outgoing calls... this might help (unsetting DYNAMIC_FEATURES for
outbound stuff didn't do anything) but my old PBX is resisting dialing
en-block so calls fail :-(
Why-o-why setting DYNAMIC_FEATURES causes the PPP hookup from my old PBX
to fail I really can't imagine. Any developers care to comment? (I'm
happy to insert debug and send info)... or should I file a bug report?
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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Re: [asterisk-users] Lamps on Snom phones
Quoth Phil Knighton [EMAIL PROTECTED] I've just completed porting from Asterisk 1.2 to 1.4. Anyway, one lingering issue is that the function key lamps on our Snom phones have all stopped working! I'm using Snom's with 1.4.X The fkeys on the Snom's are set to Type Extension and the number is sip:[EMAIL PROTECTED] (no quotes). This also means that you can press John's button to ring his phone. Works for me (ISTR the hints for sip phones are automagically generated by Asterisk 1.4.X; check it with CLI show hints). I also use func_devstate module and Snom buttons to do things like light up when a group is in night service. I've a hint like so: exten = 200,hint,Custom:nightservice200 in my dialplan and point a Snom button at sip:[EMAIL PROTECTED]. Then in my dialplan that sets/clears nightservice I have the following: exten = _*20*200.,n,Set(DEVSTATE(Custom:nightservice200)=INUSE) and exten = _*21*200.,n,Set(DEVSTATE(Custom:nightservice200)=NOT_INUSE) Voila! The group gets put into night service and the corresponding light on my Snom370 comes on. HtH -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken
Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough. I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover cable). This PBX used to be able to make and receive DIGITAL type ISDN calls through the Asterisk box... but something in the latest generation of updates has broken it and although the calls seem to work the old PBX just won't route traffic. Voice calls still work fine. I've proven it's something in Asterisk by connecting the old PBX directly to our ISDN PRI line and it still works fine. It looks like the Asterisk box is trying to make a DIGITAL type call: CLI -- Accepting overlap call from 'xx' to '0xx' on channel 0/15, span 2 -- Starting simple switch on 'Zap/46-1' -- Executing [EMAIL PROTECTED]:1] Set(Zap/46-1, CALLERID(number)=xx) in new stack -- Executing [EMAIL PROTECTED]:2] Set(Zap/46-1, SipChan=Zap/46) in new stack -- Executing [EMAIL PROTECTED]:3] Set(Zap/46-1, SipNo=46) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(Zap/46-1, Sip is 46) in new stack -- Executing [EMAIL PROTECTED]:5] GotoIf(Zap/46-1, ?grecord:gnorecord) in new stack -- Goto (alchemy,0xx,18) -- Executing [EMAIL PROTECTED]:18] Dial(Zap/46-1, Zap/g1/0xx||TWK) in new stack -- Requested transfer capability: 0x08 - DIGITAL -- Called g1/0xx -- Zap/1-1 is proceeding passing it to Zap/46-1 -- Zap/1-1 is ringing -- Zap/1-1 answered Zap/46-1 -- Channel 0/15, span 2 got hangup request, cause 16 -- Hungup 'Zap/1-1' CLI The debug available on the old PBX (Network Alchemy Argent Office) is errr... limited :-( (read non-existant). Any suggestions on how I can even start debugging this? Again I'm sure it's Asterisk 'interfering' and I know it used to work in earlier 1.4.X versions. TIA -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] Simple Asterisk to Asterisk SIP Call Setup?
I have two Asterisk systems that can route to each other via a VPN with firewalls disabled for testing purposes. Each Server can see (tested via nmap) UDP port 5060 on the other. So... I thought that I could simply use a Dial command in Server A's config to place a SIP call to Server B... but it doesn't seem to work. Server A (192.168.1.33) has: exten = *136,1,Dial(SIP/[EMAIL PROTECTED],30) but whenever a user on Server A dials '*136' the call doesn't complete and the CLI shows: Executing [EMAIL PROTECTED]:1] Dial(SIP/112-0071f650, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- SIP/10.10.111.13-00793520 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) I can't see anything in Server B's logs from 192.168.1.33 What am I missing? Any pointers to help me get this working? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] Receptionists Phone suggestions? (Not Snom370)
Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Something with (potentially) lots of BLFs, large(ish) screen, headset and most importantly the ability to transfer calls? I've installed five Snom 370s that seemed ideal but my client is very very unhappy as the Snom 370 can't transfer a call correctly if there's another call coming in (details below if you/re interested). I've verified this problem with Snom who's response is that the receptionist should answer all of the incoming calls before trying to do a transfer - That's just Bonkers! So... any suggestions? Details of Snom 370 problem for the record: Snom370 gets a Call (Call A). Snom370 answers Call A. Call A wants to be transferred to Phone C. Snom370 has another call ringing (Call B). Snom370 presses HOLD button gets Dialtone. Call A is on Hold, Call B still ringing. Snom370 Dials Phone C (Call C). Snom370 talks to Call C. Snom370 presses TRANSFER. The display shows: CallA CallB The soft keys now show and . Pressing them does nothing. When the TRANSFER button is pressed again, CallA is connected to CallB (the original caller is now talking to the previously unanswered party) not what one wanted to happen! It's not difficult to see why my client is throwing their toys out of the pram and I'm going to have to replace the Snoms at my expense :-( -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] Ringing Groups, SIP Forward and looping problem
I've a big problem with SIP forwarding back into 'ringing groups'
creating what can only be described as call storms :-(
I have a 'ringing groups' of SIP phones with an effective dialplan (much
simplified) like so:
; Purchase ledger
[ptsn_inbound]
exten = _846061,1,Dial(Local/[EMAIL PROTECTED])
[groups]
exten = 6061,1,Macro(QUEUEING_GROUP_WITH_NS,${EXTEN},Purchase)
[macro-QUEUEING_GROUP_WITH_NS]
...
exten = s,n,Dial(Sip/110Sip/111Sip/112Sip/113Sip/114)
...
If Sip/110 sets their SIP phone (SNOM 300 FWIW) to call forward to 6061
then all seems fine and calls to 110 end up in the group.
If Sip/113 *also* sets their SIP phone to call forward to 6061 then
Asterisk seems to get into a state where the calls bounce around,
ringing the phones but seemingly not allowing the call to be answered.
A 'restart now' is the only way out while this call storm is in
progress.
I'm guessing that having two SIP phones redirecting back into the
ringing group is what's causing the problem but can't think of a way
around it.
Can anyone suggest a cure?
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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Re: [asterisk-users] Manager Originate without phone off hook?
Quoth Moises Silva... May be I am missing something, but, manager command DBPut should do the trick of putting the DB value. And, since you are already using the manager interface, you are using PHP or PERL to connect to the Database, why not wait for the DBPut command response and from the script execute wget?? Yes I'm using DBPut but the GUI (in tcl/tk FWIW) is running on a different network to the phones so the http request has to come from the Asterisk box and not the one running the GUI. I guess I'm going to have to write an API and call that with Originate but I just wondered if anyone had a better (read easier!) suggestion. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] Manager Originate without phone off hook?
To answer my own question I found a way to acheive what I wanted so
here's my solution for the record (might help someone else if they
search the archives).
In the Dialplan setup the following entries:
[snom_setdndon]
exten = _.,1,NoOp(Dummy Routine Called for ${EXTEN})
exten = _.,n,TrySystem(wget -qb -O /dev/null -o /dev/null
http://admin:[EMAIL PROTECTED]/dummy.htm?settings=savednd_mode=on)
exten = _.,n,Hangup
[snom_setdndoff]
exten = _.,1,NoOp(Dummy Routine Called for ${EXTEN})
exten = _.,n,TrySystem(wget -qb -O /dev/null -o /dev/null
http://admin:[EMAIL PROTECTED]/dummy.htm?settings=savednd_mode=off)
exten = _.,n,Hangup
and then from the manager interface one can do:
Action: Originate
Channel: Local/[EMAIL PROTECTED]
Application: NoOp
Data: Setting DND
A bit convoluted but it works for me.
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] Manager Originate without phone off hook?
I'm trying to keep the DND status of my Snom phones and the astdb in
line but I'm stuck on integrating my gui DND button which talks to *
using the manager interface (actually it uses Astmanproxy as the gui
host is on a different network to asterisk and can't see the Snom's
across the network).
All's working fine in my Dialplan; when someone dials the code for
DND-on or DND-off I can do:
exten = *08,n,System(wget -qb -O /dev/null -o /dev/null
http://admin:[EMAIL PROTECTED]/dummy.htm?settings=savednd_mode=off)
exten = *08,n,Set(DB(DND/SIP/${MYSNOM})=0)
which turns the DND indicator on the phone off or on in line with the
database record. That's Great.
However, I'm completely flummoxed on getting a GUI DND button to work
sensibly via the Manager interface.
I could use 'Originate' to make the phone dial '*08' but that forces the
user to pickup the phone when they click the GUI DND button. Not Good :-(
So... can anyone suggest how I can use the Manager interface to set an
astdb record and send a request to the Snom to turn its DND indicator on
or off at the same time?
Thanks in advance.
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] Snom 300 Hints and LIne Buttons
Can anyone help with BLF for Snom 300s ? (Asterisk 1.4.10.1) I've setup hints for a couple of Snom 300's but Asterisk doesn't send Extension Changed messages to subscribed phones unless the second 'line' button is used (I've tried Snom's version 6 and 7 and two difference 300s). On the Asterisk Console I don't see any message when picking up a Snom 300 and dialing the hold music (or making any otehr call). As soon as I put the first call on hold though (by pressing the L2 button), Asterisk pops up the message xtension Changed 116 new state Hold for Notify User Russell. If I drop the first 'line', there's no message from Asterisk. When I flip back to the second line Asterisk says Extension Changed 116 new state Idle for Notify User Russell - even though it's patently not! This obviously makes the BLF lamp on my Snom 370 pretty useless as it only lights up when the Snom 300's got two lines going :-( Can anyone point me in the right direction to getting this fixed? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] 1.4.10.[0,1] crashes when call parked
100% repeatable (for me). Sip phone A calls Sip phone B. Either Sip phone A or B does #700. The party that keyed #700 gets the parked announcement (eg 701) and the other party get MOH. There is still an audio channel between the two SIP phones at this point. When the party that typed #700 hangs up, Asterisk crashes. This has been working in previous 1.4's (but not 1.4.10) and I havn't changed my parking config. Here's what comes up on the console as it crashes. -- SIP/Testsnom-00709570 Playing 'digits/7' (language 'en') -- SIP/Testsnom-00709570 Playing 'digits/0' (language 'en') -- SIP/Testsnom-00709570 Playing 'digits/1' (language 'en') -- Added extension '701' priority 1 to parkedcalls -- Stopped music on hold on SIP/115-0072f7a0 == SIP/115-0072f7a0 got tired of being parked == Spawn extension (macro-stdsip, s, 6) exited non-zero on 'SIP/Testsnom-00709570' in macro 'stdsip' == Spawn extension (macro-stdsip, s, 6) exited non-zero on 'SIP/Testsnom-00709570' *** glibc detected *** asterisk: double free or corruption (out): 0x2ab23a7ed9f0 *** === Backtrace: = /lib/libc.so.6[0x2ab23a620733] /lib/libc.so.6(__libc_free+0x84)[0x2ab23a6208b4] asterisk(ast_channel_free+0xf6)[0x438fa6] asterisk(ast_hangup+0x35a)[0x43b84a] /usr/lib/asterisk/modules/res_features.so[0x2b8298c0] asterisk[0x4a719c] /lib/libpthread.so.0[0x2ab239da23ca] /lib/libc.so.6(__clone+0x6d)[0x2ab23a67f55d] === Memory map: Anyone got any ideas? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] Howto generate a Manager Event from the Dialplan?
I'd like to be able to generate a Manager Event from the dialplan but can't seem to find a way to do it. Alternatively, trigger and Event when a record in AstDB gets changed. Can anyone point me in the right direction? Thanks. By way of explanation, I've a app that connects to astmanproxy and I'd like it to know when a call group gets put into Nightservice. Putting the call group into Nightservice is done in the dialplan and sets a record in AstDB. It would be infriendly to poll AstDB; hence the requirement for the dialplan to trigger an Event. The call group could also be put into Nightservice by setting the appropriate record directly in AstDB; hence the Event triggered on a AstDB record change. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] How does one use sip_autoreg
I've RTFM and Googled but can't seem to get sip_autoreg to work (or perhaps I'm just completely missing the point of it). (what I'd like to do is avoid having to put explicit entries for every SIP phone into extensions.conf). Asterisk is creating entries in the (virtual) context sip_autoreg: asterisk*CLI dialplan show sip_autoreg [ Context 'sip_autoreg' created by 'SIP' ] '112' = 1. Noop(112) [SIP] '113' = 1. Noop(113) [SIP] '114' = 1. Noop(114) [SIP] -= 3 extensions (3 priorities) in 1 context. =- asterisk*CLI and in my dialplan I have: [from-sip] include = actual_sip_autoreg include = other_stuff_and_widgets ...in [actual_sip_autoreg] I have: [actual_sip_autoreg] include = sip_autoreg exten = _.,n,NoOp(Here I am Mr Mgoo) exten = _.,n,Playback(tt-weasels) But when a SIP phone dials 114 the Here I am Mr Mgoo NoOp doesn't get executed. Here's what the console says: asterisk*CLI -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/Russell-0071e7b0, 114) in new stack -- Executing [EMAIL PROTECTED]:2] Congestion(SIP/Russell-0071e7b0, ) in new stack == Spawn extension (from-sip, 114, 2) exited non-zero on 'SIP/Russell-0071e7b0' asterisk*CLI ...a dialplan show actual_sip_autoreg produces: asterisk*CLI dialplan show actual_sip_autoreg [ Context 'actual_sip_autoreg' created by 'pbx_config' ] '_.' = 7. NoOp(Here I am Mr Mgoo) [pbx_config] 8. Playback(tt-weasels) [pbx_config] Include ='sip_autoreg' [pbx_config] -= 1 extension (2 priorities) in 1 context. =- asterisk*CLI So... anyone got any idea why the NoOp and thence the Weasels don't get executed? Ta in advance. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] Asterisk 1.4.5 Inserting Random Digits in Dialed Number!
Eeeeck! Asterisk is inserting random digits in dialed numbers.
So far I've seen it insert a 2 after the STD (area) code and insert an
extra 6 or 7 in the STD code. It's pretty repeatable although the
inserted number changes.
My Config is: Asterisk 1.4.5, Zaptel 1.4.3, Digium TE205P (rev 02).
There's an ISDN PBX on the second span and a BRI euroisdn on the first.
Calls from the ISDN PBX (Network Alchemy Argent Office FWIW) get put
into the 'alchemy' context which contains:
[alchemy]
;
include = dial_pstn
;
; Dial Out to PSTN
;
[dial_pstn]
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Congestion
;
Pretty simple and didn't cause problems before I went to 1.4.4/5
Here's the console log from a failing call (non-significant numbers
obscured to protect the guilty!).
*CLI -- Accepting overlap call from '1780471800' to '0173' on channel
0/15, span 2
-- Starting simple switch on 'Zap/46-1'
[Jun 28 12:00:32] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '3' received
on Zap/46-1, duration 0 ms
[Jun 28 12:00:32] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '2' received
on Zap/46-1, duration 0 ms
[Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '3' received
on Zap/46-1, duration 0 ms
[Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '5' received
on Zap/46-1, duration 0 ms
[Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end 'x' received
on Zap/46-1, duration 0 ms
[Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end 'x' received
on Zap/46-1, duration 0 ms
[Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end 'x' received
on Zap/46-1, duration 0 ms
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/46-1,
Zap/g1/017332235xxx) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/017332235xxx
-- Zap/1-1 is proceeding passing it to Zap/46-1
As you can see, the ISDN PBX passed 01733 235xxx but Asterisk decided to
dial 01733 2235xxx (an extra '2' after the STD code).
Arrgh!!! This is driving both us and the people who're getting
called incorrectly potty :-(
Can anyone help? Thanks in advance.
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] SIP DTMF not acted on for features in 1.4.0b3
Asterisk seems to be ignoring DTMF for features in Asterisk 1.4.0b3 My SNOM sends the dtmf-relay and Asterisk gets it because I can see it in the sip debug. However, once seen, Asterisk doesn't actually do anything about it. I've checked features and that seems fine. Is this a bug or something that I've screwed up? For the record, here's the features setting: asterisk*CLI show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer *2 One Touch Monitor *1 Disconnect Call * * Park Call #72 Dynamic Feature Default Current --- --- --- testfeature no def #9 Call parking Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-720 asterisk*CLI and here's a SIP trace of me pressing '*' during a call (which according to my features should Disconnect the Call. asterisk*CLI --- SIP read from 192.168.1.12:5060 --- INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK-llnm8m4u2wef;rport From: Russell 112 sip:[EMAIL PROTECTED];tag=tmyszljbna To: sip:[EMAIL PROTECTED];tag=as0b7389e4 Call-ID: [EMAIL PROTECTED] CSeq: 14 INFO Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;line=gv8x1x75;flow-id=1 User-Agent: snom360/6.5.1 Content-Type: application/dtmf-relay Content-Length: 22 Signal=* Duration=160 - --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * asterisk*CLI --- Transmitting (no NAT) to 192.168.1.12:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK-llnm8m4u2wef;received=192.168.1.12;rport=5060 From: Russell 112 sip:[EMAIL PROTECTED];tag=tmyszljbna To: sip:[EMAIL PROTECTED];tag=as0b7389e4 Call-ID: [EMAIL PROTECTED] CSeq: 14 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/112-0070a2c0] asterisk*CLI Can anyone suggest what's wrong here? Thanks. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] RE: Extending Avaya IP Office ISDN30e with Asterisk
Has anyone hooked up * as an extension/trunk of an Avaya system that has around 2 ISDN30e's. I'm currently running an Asterisk box between my ISDN30 PRI and my Argent Office (pre Avaya takeover of Network Alchemy but still the same box as the Avaya IP Office). All it took was a two PRI digium card and a PRI crossover cable between the Asterisk box and the Argent. You'd need a four port card of course. All seems to work well. I did have some issues with the Argent rejecting data calls when the DDI wasn't set to receive them but relied on the default 'route all DATA type calls to this service' in the Argent. I cured this by setting an explicit incoming call route for these DATA calls on the Argent and it worked. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] Trouble with regexten
Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ] '98766' =1. Dial(Sip/Tim) [pbx_config] 2. Hangup() [pbx_config] Include ='sip_autoreg'[pbx_config] Include ='widgets'[pbx_config] -= 1 extension (2 priorities) in 1 context. =- asterisk*CLI and here's sip_autoreg (the regexten context): asterisk*CLI dialplan show sip_autoreg [ Context 'sip_autoreg' created by 'pbx_config' ] '114' = 2. Dial(Sip/Tim) [pbx_config] 3. Hangup() [pbx_config] [ Context 'sip_autoreg' created by 'SIP' ] '112' = 1. Noop(Russell) [SIP] '113' = 1. Noop(Richard) [SIP] '114' = 1. Noop(Tim) [SIP] -= 4 extensions (5 priorities) in 2 contexts. =- asterisk*CLI Dialing 98766 from Sip/Russell rings Sip/Tim as expected. Dialing 114 gives Not Found :-( I'm very confused any ideas why this doesn't work? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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
