Re: [asterisk-users] Question on AEL2 string comparisons

2013-07-03 Thread Satish Barot
On Thu, Jul 4, 2013 at 12:24 AM, James B. Byrne wrote: > I have this code in a dial plan: > > exten => _417XX,n,GotoIf($["${CALLERID(num)}" > > "SIP/41799"]?notfromlocal) > exten => _417XX,n,GotoIf($["${CALLERID(num)}" < > "SIP/41700"]?notfromlocal) > > The value of "${CALLERID(num)}" appears to b

Re: [asterisk-users] Executing Script after MixMonitor is called

2013-07-03 Thread Satish Barot
On Thu, Jul 4, 2013 at 1:30 AM, Gopalakrishnan N < gopalakrishnan...@gmail.com> wrote: > I tried with hangup cause but my script is not executed... also I tried > the same script with mix monitor itself no sucess. > > The script what I have is, am converting wav file to flac format.. > On 11 Jun 2

Re: [asterisk-users] Queue questions - Asterisk 11

2013-07-03 Thread Satish Barot
On Wed, Jul 3, 2013 at 7:40 PM, Administrator TOOTAI wrote: > Le 03/07/2013 15:07, Satish Barot a écrit : > >> [...] >> >> Then you should add Local channel as a queue member and dial your SIP >> member from Local channel context. A little hint here. Suppose you have a >> support queue configured

Re: [asterisk-users] Asterisk stops registering

2013-07-03 Thread Eric Wieling
See cli.conf.sample -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ian Pilcher Sent: Wednesday, July 03, 2013 7:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk stops registerin

Re: [asterisk-users] Asterisk stops registering

2013-07-03 Thread Ian Pilcher
On 07/03/2013 05:51 PM, David Duffett wrote: > I would enable SIP debugging, but only for that provider. Will that increase the logging verbosity? > This can be done on the Asterisk command line by using either of the > following: > > *> sip set debug peer > > or > > *> sip set debug ip Can

Re: [asterisk-users] Asterisk stops registering

2013-07-03 Thread David Duffett
I would enable SIP debugging, but only for that provider. This can be done on the Asterisk command line by using either of the following: *> sip set debug peer or *> sip set debug ip On 3 July 2013 23:25, Ian Pilcher wrote: > On several occassions lately, my home Asterisk box has stopped

[asterisk-users] Asterisk stops registering

2013-07-03 Thread Ian Pilcher
On several occassions lately, my home Asterisk box has stopped registering with my VoIP provider. I haven't been able to reproduce the problem, and the log doesn't contain anything useful. How can I increase the log verbosity for SIP registration-related events? I've looked through logger.conf a

Re: [asterisk-users] Executing Script after MixMonitor is called

2013-07-03 Thread Gopalakrishnan N
I tried with hangup cause but my script is not executed... also I tried the same script with mix monitor itself no sucess. The script what I have is, am converting wav file to flac format.. On 11 Jun 2013 11:17, "Satish Barot" wrote: > And yes if you want to use System application in your dialpl

[asterisk-users] Question on AEL2 string comparisons

2013-07-03 Thread James B. Byrne
I have this code in a dial plan: exten => _417XX,n,GotoIf($["${CALLERID(num)}" > "SIP/41799"]?notfromlocal) exten => _417XX,n,GotoIf($["${CALLERID(num)}" < "SIP/41700"]?notfromlocal) The value of "${CALLERID(num)}" appears to be "SIP/41712-0181" -- Executing [41720@from-internal:5] GotoI

[asterisk-users] Calls drop after transfer

2013-07-03 Thread Carlos Chavez
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have an Asterisk 11.4 SIP only system. We are using a SIP trunk for outside calls. We are having a problem with calls dropping after a transfer. Outside call awswered by phone 101 101 transfers to 100 (attended transfer) call is dropped af

[asterisk-users] Custom dial plan for internal transfers of external calls

2013-07-03 Thread James B. Byrne
Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 We use Snom870 handsets with firmware v.8.7.3.19. I am trying to develop a custom dial plan to invoke a distinctive ring-tone when an external call is transferred internally. Based on an earlier solution I discovered I

Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"

2013-07-03 Thread Amit Patkar | ATPL
Faced some issues with size. So deleted some content. But this was full file and we are using default file. Hi Matt, I have pasted entire say.conf here. It has datetime extension. ; ; language configuration ; [general] mode=new; method for playing numbers and dates ; old - using ast

Re: [asterisk-users] Queue questions - Asterisk 11

2013-07-03 Thread Administrator TOOTAI
Le 03/07/2013 15:07, Satish Barot a écrit : [...] Then you should add Local channel as a queue member and dial your SIP member from Local channel context. A little hint here. Suppose you have a support queue configured in queues.conf ;queues.conf [support] ... ... member => Local/1000@member

Re: [asterisk-users] SIP. Call-limit dialstatus

2013-07-03 Thread I.Pavlov
Thanks for answer. For correct dialstatus I use now: Set(DIALSTATUS=${IF($[ "${SIPPEER(${EXTEN},curcalls)}" >= "${SIPPEER(${EXTEN},limit)}" ]?BUSY:${DIALSTATUS})}); I tried to use Busy app and got CDR(disposition)=BUSY, but in this way I can't redirect *calling* channel to voicemail, becaus

Re: [asterisk-users] Queue questions - Asterisk 11

2013-07-03 Thread Satish Barot
On Wed, Jul 3, 2013 at 2:37 PM, Administrator TOOTAI wrote: > Hi Satish > > Le 03/07/2013 09:15, Satish Barot a écrit : > > >> On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI >> > ad...@tootai.net>> wrote: >> >> Hi all, >> >> I have to questions about queues. Member is a phone like

Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"

2013-07-03 Thread Matthew Jordan
On Tue, Jul 2, 2013 at 11:03 AM, Amit Patkar | ATPL wrote: > Hi Matt, > > As required, please find DEBUG trace for datetime function. I have used > this function in Dialplan to capture DEBUG trace. I hope, this can help us > in resolving the issue. > > [Jul 2 15:54:44] DEBUG[2698] chan_sip.c: C

[asterisk-users] CEL events

2013-07-03 Thread Jairo
Dear list. This is probably a complex subject but is that right to consider: a) each distinct linkedid field value in a mysql CEL table as a unique call? b) the duration of a call as the period (eventtime fields) between BRIDGE_END and BRIDGE_START events of the same linkedid sequence? (not cons

Re: [asterisk-users] SIP. Call-limit dialstatus

2013-07-03 Thread Steven Howes
On 3 Jul 2013, at 12:28, I.Pavlov wrote: > [2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter: > Call to peer '0014' rejected due to usage limit of 1 > -- Couldn't call 0014 > == Everyone is busy/congested at this time (0:0/0/0) > -- Executing [0014@sub_pbxdialco:50

[asterisk-users] SIP. Call-limit dialstatus

2013-07-03 Thread I.Pavlov
Hi all. We have a problem with correct dialstatus and cdr(disposition) when using call-limit. When call-limit reached dialstatus is CHANUNAVAIL and CDR(disposition)='NO ANSWER' -- Executing [0014@sub_pbxdialco:49] Dial("SIP/1295-01f8", "SIP/0014,12,tTkK") in new stack == Using SIP RT

Re: [asterisk-users] Queue questions - Asterisk 11

2013-07-03 Thread Administrator TOOTAI
Hi Satish Le 03/07/2013 09:15, Satish Barot a écrit : On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI mailto:ad...@tootai.net>> wrote: Hi all, I have to questions about queues. Member is a phone like SIP/myphone and only one member in the queue. At first, DIALSTATUS d

Re: [asterisk-users] Queue questions - Asterisk 11

2013-07-03 Thread Satish Barot
On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI wrote: > Hi all, > > I have to questions about queues. Member is a phone like SIP/myphone and > only one member in the queue. > > At first, DIALSTATUS doesn't return any status. How to now if a call in > queue has been answered or if caller jus