[asterisk-users] OPTIONS Packet is retransmitting continuously
HI In My asterisk OPTIONS packet is retransmitting continuously ,any one know the reason for this.I am using asterisk 1.6.1.1. Thanks Nikhil -- _ -- 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] environment variable + res_mysql.conf
Hi All. I have export some db parameter in /etc/bashrc as follows ... export DB_NAME=xyz export DB_IP=1x.1x.1x.1x export DB_PWD=dkjfaoi Now, I want use these all environment variable into /etc/asterisk/res_mysql.conf file. Is there any way to do this..?? -- Regards, Chandrakant Solanki -- _ -- 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] How to check a number online or offline
Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -- _ -- 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] Mail list Woes?
On Jan 9, 2011, at 8:27 AM, William Stillwell wrote: > Anybody notice log delays in this list, and very small amount of traffic? I have noticed multiple hour delays between sending messages and seeing them back. Tom -- _ -- 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] Call parking question
I've been playing with call parking in Asterisk 1.8.1. I'm able to park a call and then pick it back up. However, on the second attempt, the #72 DTMF is ignored. Asterisk just passes that DTMF on to the caller and the call parking never happens. Shouldn't I be able to park a call more than once? -- Chris -- _ -- 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] system lockup when going into conference
Tzafrir Cohen wrote: > On Thu, Jan 06, 2011 at 07:01:00PM -0500, cov...@ccs.covici.com wrote: > > Hi. I have an asterisk system under Debian Leni using asterisk 1.8 with > > no Digium hardware -- and when I go into a meetme conference the system > > either locks up or is 100% cpu utilized or something -- I can't type > > anything and I have to physically reboot the system. The dahdi module is > > loaded and the last log entry is the playing of you are the only person > > in this conference,. > > > > How would I even start to debug this one? > > > > Any ideas would be appreciated. > > What version of DAHDI? PProblem solved by Shaun fixing a regression in dahdi-trunk. Thanks to him its now working. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] Mail list Woes?
On Sun, Jan 9, 2011 at 10:27 AM, William Stillwell wrote: > Anybody notice log delays in this list, and very small amount of traffic? There are posts by Kevin F. and others that there is an issue and they are working on it. ~~~ Andrew "lathama" Latham lath...@gmail.com ~~~ -- _ -- 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] Mail list Woes?
Hehe i thought it was me and never commented Sent from my android device. -Original Message- From: William Stillwell To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Sun, 09 Jan 2011 13:47 Subject: [asterisk-users] Mail list Woes? Anybody notice log delays in this list, and very small amount of traffic? -- _ -- 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] Definations of READ/WRITE parameters of manager.conf contexts?
You are the sort of person who takes the time to write something totally in vain. I never understand the sort of mentality people like you have. No need to respond with an obnoxious comment. ***I was hoping for pointers in asterisk's new wiki or somewhere else that I may not be aware of. On Sat, Jan 8, 2011 at 11:27 AM, Steve Edwards wrote: > On Fri, 7 Jan 2011, Bruce B wrote: > > I want to know each and every parameter's detail that can be included in >> the >> >> read= >> write= >> >> in manager.conf >> >> Where can I find this? >> > > 0) Try and spell check the subject a bit better. It will make it easier for > the 'next guy' to search for. > > 1) Google. > > 2) The Asterisk source code. Even if you aren't a C programmer grepping > through the source code can be very productive. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?
Thanks Paul. That is exactly what I was looking for. On Sat, Jan 8, 2011 at 2:07 PM, Paul Belanger wrote: > On 11-01-07 01:33 PM, Bruce B wrote: > > Where can I find this? > > > manager.conf.sample? > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Benefit of PRI vs SIP trunk calls
Jim Dickenson wrote:
I am running version 1.4.x. Where do I get PRICAUSE?
NoOP(Hangup Cause: ${HANGUPCAUSE})
Doug
--
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deserve neither Liberty nor Safety."
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Re: [asterisk-users] system lockup when going into conference
On Thu, Jan 06, 2011 at 07:01:00PM -0500, cov...@ccs.covici.com wrote: > Hi. I have an asterisk system under Debian Leni using asterisk 1.8 with > no Digium hardware -- and when I go into a meetme conference the system > either locks up or is 100% cpu utilized or something -- I can't type > anything and I have to physically reboot the system. The dahdi module is > loaded and the last log entry is the playing of you are the only person > in this conference,. > > How would I even start to debug this one? > > Any ideas would be appreciated. What version of DAHDI? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Are the Siren7 and Siren14 the G.722 HD voice codecs?
Actually, all of the conference phones are known by the "SoundStation" name and the desk phones are "SoundPoint." Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves > Original Message > Subject: Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD > voice codecs? > From: Steve Underwood > Date: Sat, January 08, 2011 11:16 am > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > On 01/08/2011 03:44 AM, Kevin P. Fleming wrote: > > On 01/06/2011 11:34 AM, mgra...@mstvp.com wrote: > >> We should also be very clear that the Siren codecs are supported on the > >> Polycom SoundStation conference phones and the VVX-1500 Business Media > >> Phones. These codecs are not supported in the SoundPoint desk phones. > >> The SoundPoint series support the more basic G.722 codec in the > >> IP335/450/550/560/650/670 models. > > > > The SoundPoint IP6000 and IP7000 conference phones (and maybe the > > IP5000, I haven't checked) also support G.722.1 and G.722.1C. > > > The IP6000 is actually model Polycom recommended for testing when we > implemented G.722.1. > > One of the annoying things about the Polycoms is trying to work out what > they can do. You have to search quite hard to find which codecs each > model supports. > > Steve > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mail list Woes?
Anybody notice log delays in this list, and very small amount of traffic? -- _ -- 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] Benefit of PRI vs SIP trunk calls
I am running version 1.4.x. Where do I get PRICAUSE? I tried making a call that
was not answered and I did not see any more information. The dumpchan of
DADHI/23-1 did not happen as that is in a macro that only gets called for an
answered call.
I only see this:
Executing [91112223...@empl:8] Dial("SIP/mine-0521",
"Dahdi/G1/111222|60|gM(out-dial)") in new stack
DEBUG[4907]: dsp.c:1682 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/111222
DEBUG[3188]: chan_dahdi.c:10135 pri_dchannel: Queuing frame from
PRI_EVENT_PROCEEDING on channel 0/23 span 1
-- DAHDI/23-1 is proceeding passing it to SIP/mine-0521
-- DAHDI/23-1 is ringing
DEBUG[3188]: chan_dahdi.c:1790 dahdi_enable_ec: Echo cancellation already on
-- DAHDI/23-1 answered SIP/mine-0521
-- Executing [...@macro-out-dial:1] DumpChan("DAHDI/23-1", "") in new stack
Dumping Info For Channel: DAHDI/23-1:
Info:
Name= DAHDI/23-1
Type= DAHDI
UniqueID= sys.domain.com-1294514614.2630
CallerID= 9111222
CallerIDName= (N/A)
DNIDDigits= (N/A)
RDNIS= (N/A)
State= Up (6)
Rings= 0
NativeFormat= 0x4 (ulaw)
WriteFormat=0x4 (ulaw)
ReadFormat= 0x4 (ulaw)
1stFileDescriptor= 35
Framesin= 189
Framesout= 176
TimetoHangup= 0
ElapsedTime=0h0m4s
Context=macro-out-dial
Extension= s
Priority= 1
CallGroup=
PickupGroup=
Application=DumpChan
Data= (Empty)
Blocking_in=(Not Blocking)
Variables:
MACRO_DEPTH=1
MACRO_PRIORITY=1
MACRO_CONTEXT=from-outside
MACRO_EXTEN=
DIALEDPEERNUMBER=G1/111222
TRANSFERCAPABILITY=SPEECH
DEBUG[4907]: app_macro.c:379 _macro_exec: Executed application: DumpChan
DEBUG[4907]: app_dial.c:1927 dial_exec_full: Macro exited with status 0
DEBUG[4907]: chan_dahdi.c:3464 dahdi_setoption: Set option AUDIO MODE, value:
ON(1) on DAHDI/23-1
DEBUG[4907]: chan_dahdi.c:3092 dahdi_hangup: Not yet hungup... Calling hangup
once with icause, and clearing call
DEBUG[4907]: chan_dahdi.c:3460 dahdi_setoption: Set option AUDIO MODE, value:
OFF(0) on DAHDI/23-1
-- Hungup 'DAHDI/23-1'
== Spawn extension (empl, 9111222, 8) exited non-zero on
'SIP/mine-0521'
-- Executing [...@empl:1] Verbose("SIP/mine-0521", "2|Hangup
SIP/mine-0521 with cause 16") in new stack
== Hangup SIP/mine-0521 with cause 16
-- Executing [...@empl:2] DumpChan("SIP/mine-0521", "") in new stack
Dumping Info For Channel: SIP/mine-0521:
Info:
Name= SIP/mine-0521
Type= SIP
UniqueID= sys.domain.com-1294514614.2629
CallerID= 444555
CallerIDName= Jim Dickenson
DNIDDigits= 9111222
RDNIS= (N/A)
State= Up (6)
Rings= 0
NativeFormat= 0x2 (gsm)
WriteFormat=0x2 (gsm)
ReadFormat= 0x2 (gsm)
1stFileDescriptor= 65
Framesin= 248
Framesout= 253
TimetoHangup= 0
ElapsedTime=0h0m0s
Context=empl
Extension= h
Priority= 2
CallGroup=
PickupGroup=
Application=DumpChan
Data= (Empty)
Blocking_in=(Not Blocking)
Variables:
DIALSTATUS=ANSWER
DIALEDTIME=5
ANSWEREDTIME=1
RTPAUDIOQOS=ssrc=671389293;themssrc=651772178;lp=0;rxjitter=0.001217;rxcount=248;txjitter=0.00;txcount=252;rlp=0;rtt=0.00
BRIDGEPEER=DAHDI/23-1
DIALEDPEERNUMBER=G1/111222
DIALEDPEERNAME=DAHDI/23-1
MACRO_DEPTH=0
RCStatus=0
MyChan=SIP
sipcallid=0b69233cd5469...@192.168.0.16
SIPUSERAGENT=Grandstream GXP2000 1.2.2.6
SIPDOMAIN=sys.domain.com
SIPURI=sip:m...@00.00.000.000:5064
-- Executing [...@empl:3] ExecIf("SIP/mine-0521",
"0|Set|DB(conf//haveadmin)=no") in new stack
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 7, 2011, at 12:44 PM, C F wrote:
> PRICAUSE will give you lots of info on why a call was hungup on. Not
> sure if SIP will give you the same.
>
> On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson wrote:
>> Does Asterisk, currently using version 1.4, get any more information about
>> the result of an outbound call made over a PRI line compared to a call via a
>> SIP trunk?
>>
>> As an example, in a PRI call there is this message that shows up on the
>> console:
>>
>> [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network.
>>
>> for a call to a fax machine. Does asterisk set anything that a dialplan can
>> access that can know the c
Re: [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?
On 11-01-07 01:33 PM, Bruce B wrote: > Where can I find this? > manager.conf.sample? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _ -- 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] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/08/2011 03:44 AM, Kevin P. Fleming wrote: On 01/06/2011 11:34 AM, mgra...@mstvp.com wrote: We should also be very clear that the Siren codecs are supported on the Polycom SoundStation conference phones and the VVX-1500 Business Media Phones. These codecs are not supported in the SoundPoint desk phones. The SoundPoint series support the more basic G.722 codec in the IP335/450/550/560/650/670 models. The SoundPoint IP6000 and IP7000 conference phones (and maybe the IP5000, I haven't checked) also support G.722.1 and G.722.1C. The IP6000 is actually model Polycom recommended for testing when we implemented G.722.1. One of the annoying things about the Polycoms is trying to work out what they can do. You have to search quite hard to find which codecs each model supports. Steve -- _ -- 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] Polarity Reverseal....with analog line
Can I reverse the polarity from Asterisk to get the call ?
I have 5 days with this and I dont know what to do.
I changed zaptel for dAHDI now I have Dahdi 2.4 and asterisk 1.4.30
TIA
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-Soporte PostgreSQL
*-www.jqmicrosistemas.com
*-809-849-8087
*---*
> Date: Wed, 5 Jan 2011 17:04:02 -0500
> From: markm-li...@intellasoft.net
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Polarity Reversealwith analog line
>
> Looks like your telco is sending you polarity reversal on sending you a
> call. Which is one of the types of setups for analog lines.l
>
> From your console output it looks like the call was handled just fine
> other than the 'weird event' notification, which I'm not familiar with.
>
>
>
> On 01/05/2011 11:50 AM, Edwin Quijada wrote:
> > Hi !
> > I ma having trouble with my PTSN line. When I call to my asterisk I get
> > this..
> >
> > -- Executing [...@from-pstn:3] Hangup("Zap/5-1", "") in new stack
> > == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1'
> > -- Hungup 'Zap/5-1'
> > -- Starting simple switch on 'Zap/5-1'
> > [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17
> > (Polarity Reversal)...
> > [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17
> > (Polarity Reversal)...
> > [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17
> > (Polarity Reversal)...
> > -- Executing [...@from-pstn:1] Answer("Zap/5-1", "") in new stack
> > -- Executing [...@from-pstn:2] Playback("Zap/5-1", "vm-intro") in new stack
> > -- Playing 'vm-intro' (language 'en')
> > [Jan 5 12:45:08] WARNING[2893]: chan_dahdi.c:4550 dahdi_handle_event:
> > Ring/Off-hook in strange state 6 on channel 5
> > -- Executing [...@from-pstn:3] Hangup("Zap/5-1", "") in new stack
> > == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1'
> > -- Hungup 'Zap/5-1'
> >
> > I am using 1.4.30 and zaptel 1.12.
> >
> > Any cluess?
> > *---*
> > *-Edwin Quijada
> > *-Developer DataBase
> > *-JQ Microsistemas
> > *-Soporte PostgreSQL
> > *-www.jqmicrosistemas.com
> > *-809-849-8087
> > *---*
> >
> >
> >
> >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> > http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Definations of READ/WRITE parameters of manager.conf contexts?
On Fri, 7 Jan 2011, Bruce B wrote: I want to know each and every parameter's detail that can be included in the read= write= in manager.conf Where can I find this? 0) Try and spell check the subject a bit better. It will make it easier for the 'next guy' to search for. 1) Google. 2) The Asterisk source code. Even if you aren't a C programmer grepping through the source code can be very productive. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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
