Re: [asterisk-users] Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted
use 'ulimit' to set a higher value on max open file descriptors On 2 July 2011 02:00, Eric Wieling wrote: > > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > Kaushal Shriyan > > Sent: Friday, July 01, 2011 8:28 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Starting asterisk: > > /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot > > modify limit: Operation not permitted > > > > On Fri, Jul 1, 2011 at 11:13 AM, Kaushal Shriyan > > wrote: > > > Hi > > > > > > Please help me understand about the below issue ? > > > > > > [root@asterisk1 ~]# /etc/init.d/asterisk restart Stopping > > > safe_asterisk:[ OK ] Shutting > > > down asterisk:[ OK ] Starting > > > asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open > > > files: cannot modify limit: Operation not permitted > > > [ OK ] > > > (reverse-i-search)`d': /etc/init.d/asterisk restart > > > [root@asterisk1 ~]# rpm -qa | grep asterisk > > > asterisk-sounds-core-en-gsm-1.4.21-1_centos5 > > > asterisk18-1.8.4.4-1_centos5 > > > asterisk18-core-1.8.4.4-1_centos5 > > > asterisk18-doc-1.8.4.4-1_centos5 > > > asterisk18-dahdi-1.8.4.4-1_centos5 > > > asterisk18-configs-1.8.4.4-1_centos5 > > > asterisk18-voicemail-1.8.4.4-1_centos5 > > > [root@asterisk1 ~]# uname -a > > > Linux asterisk1 2.6.18-238.el5 #1 SMP Thu Jan 13 15:51:15 EST 2011 > > > x86_64 x86_64 x86_64 GNU/Linux > > > [root@asterisk1 ~]# cat /proc/version > > > Linux version 2.6.18-238.el5 (mockbu...@builder10.centos.org) (gcc > > > version 4.1.2 20080704 (Red Hat 4.1.2-48)) #1 SMP Thu Jan > > 13 15:51:15 > > > EST 2011 > > > [root@asterisk1 ~]# cat /etc/redhat-release CentOS release > > 5.6 (Final) > > > [root@asterisk1 ~]# > > > > > > Regards > > > > > > Kaushal > > > > > > > Hi Again, > > > > Can someone please reply on my earlier post to this emailing list. > > This is an operating system question. The link is for core size, but the > basic concept should work for open files as well. > > > http://superuser.com/questions/79717/bash-ulimit-core-file-size-cannot-modify-limit-operation-not-permitted > > -- > _ > -- 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] Distributing the incoming calls and the huntgroup
Hi All; To be able to distribute the incoming calls on a group of extensions, is there huntgroup in Asterisk? Or what I have to use? I need first call to be send for extension 500 and second call to be send for extension 501 and third call to be send for extension 502 and fourth call to be send again for extension 501 and so on .. I searched for huntgroup in Asterisk, but did not find any thing related to huntgroup in asterisk ! It look like there is not huntgroup in asterisk?! So how to distribute the calls? Regards Bilal -- _ -- 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] Load Balance Trunks
On Fri, 1 Jul 2011, A J Stiles wrote: But you'll need to contact me off-list, as the rules here forbid the discussion of services in respect of which money is going to be changing hands. Love it :) -- 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
Re: [asterisk-users] Distributing the incoming calls and the huntgroup
FreeBPX calls them Ring Groups, you can look in to that. Or you could use a small ACD group. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Saturday, July 02, 2011 12:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Distributing the incoming calls and the huntgroup Hi All; To be able to distribute the incoming calls on a group of extensions, is there huntgroup in Asterisk? Or what I have to use? I need first call to be send for extension 500 and second call to be send for extension 501 and third call to be send for extension 502 and fourth call to be send again for extension 501 and so on .. I searched for huntgroup in Asterisk, but did not find any thing related to huntgroup in asterisk ! It look like there is not huntgroup in asterisk?! So how to distribute the calls? Regards Bilal -- _ -- 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] chanspy spies on wrong channel
asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use
flash operator panel < 2.0
(from extensions.conf)
exten=> 304,1,ChanSpy(Zap/4|q)
exten=> 304,2,hangup
There is no entry ChanSpy(Zap/41) in extensions.conf
On dialing 304 and Zap/41 is in use this happens:
[Jul 1 18:24:47] VERBOSE[14447] logger.c: -- Executing
[304@flash:1] ChanSpy("Zap/31-1", "Zap/4|q") in new stack
[Jul 1 18:24:47] VERBOSE[14447] logger.c: == Spying on channel Zap/41-1
[Jul 1 18:24:47] NOTICE[14447] app_chanspy.c: Attaching Zap/31-1 to
Zap/41-1
If while spying on Zap/41 that channel is hung up:
[Jul 1 19:06:48] VERBOSE[15242] logger.c: == Done Spying on channel
Zap/41-1
[Jul 1 19:06:48] VERBOSE[15242] logger.c: == Spying on channel Zap/4-1
[Jul 1 19:06:48] NOTICE[15242] app_chanspy.c: Attaching Zap/31-1 to Zap/4-1
thanks list
Steve
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Re: [asterisk-users] SIP Peer Name Variable
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Dan Journo
> Sent: Saturday, July 02, 2011 8:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] SIP Peer Name Variable
>
> Hi,
>
>
>
> Is there a variable that contains the Sip Peer name?
>
> I was using ${CALLERID(num)} for outgoing calls, but when a
> call is being transferred, that variable contains something else.
>
>
>
> I need a variable that is always set to the SIP Peer's name.
pbx*CLI> core show function CHANNEL
-= Info about function 'CHANNEL' =-
[Synopsis]
Gets/sets various pieces of information about the channel.
[Description]
Gets/sets various pieces of information about the channel, additional
may be available from the channel driver; see its documentation for details.
Any requested that is not available on the current channel will return
an empty string.
[Syntax]
CHANNEL(item)
[Arguments]
item
Standard items (provided by all channel technologies) are:
audioreadformat - R/O format currently being read.
audionativeformat - R/O format used natively for audio.
audiowriteformat - R/O format currently being written.
callgroup - R/W call groups for call pickup.
channeltype - R/O technology used for channel.
checkhangup - R/O Whether the channel is hanging up (1/0)
language - R/W language for sounds played.
musicclass - R/W class (from musiconhold.conf) for hold music.
name - The name of the channel
parkinglot - R/W parkinglot for parking.
rxgain - R/W set rxgain level on channel drivers that support it.
secure_bridge_signaling - Whether or not channels bridged to this
channel require secure signaling
secure_bridge_media - Whether or not channels bridged to this channel
require secure media
state - R/O state for channel
tonezone - R/W zone for indications played
transfercapability - R/W ISDN Transfer Capability, one of:
SPEECH
DIGITAL
RESTRICTED_DIGITAL
3K1AUDIO
DIGITAL_W_TONES
VIDEO
txgain - R/W set txgain level on channel drivers that support it.
videonativeformat - R/O format used natively for video
trace - R/W whether or not context tracing is enabled, only available
*if CHANNEL_TRACE is defined*.
*chan_sip* provides the following additional options:
peerip - R/O Get the IP address of the peer.
recvip - R/O Get the source IP address of the peer.
from - R/O Get the URI from the From: header.
uri - R/O Get the URI from the Contact: header.
useragent - R/O Get the useragent.
peername - R/O Get the name of the peer.
t38passthrough - R/O '1' if T38 is offered or enabled in this channel,
otherwise '0'
rtpqos - R/O Get QOS information about the RTP stream
This option takes two additional arguments:
Argument 1:
'audio' Get data about the audio stream
'video' Get data about the video stream
'text' Get data about the text stream
Argument 2:
'local_ssrc'Local SSRC (stream ID)
'local_lostpackets' Local lost packets
'local_jitter' Local calculated jitter
'local_maxjitter' Local calculated jitter (maximum)
'local_minjitter' Local calculated jitter (minimum)
'local_normdevjitter'Local calculated jitter (normal
deviation)
'local_stdevjitter' Local calculated jitter (standard
deviation)
'local_count' Number of received packets
'remote_ssrc' Remote SSRC (stream ID)
'remote_lostpackets'Remote lost packets
'remote_jitter' Remote reported jitter
'remote_maxjitter' Remote calculated jitter (maximum)
'remote_minjitter' Remote calculated jitter (minimum)
'remote_normdevjitter'Remote calculated jitter (normal
deviation)
'remote_stdevjitter'Remote calculated jitter (standard
deviation)
'remote_count' Number of transmitted packets
'rtt' Round trip time
'maxrtt'Round trip time (maximum)
'minrtt'Round trip time (minimum)
'normdevrtt'Round trip time (normal deviation)
'stdevrtt' Round trip time (standard deviation)
'all' All statistics (in a form suited to
logging, but not for parsing)
rtpdest - R/O Get remote RTP destination information.
This option takes one additional argument:
Argument 1:
'audio' Get audio destination
'video' Get video destination
'text' Get text destination
*chan_iax2* provides the following additional options:
peerip - R/O Get the peer's ip address.
peername - R/O Get the peer's username.
*chan_dahdi*
Re: [asterisk-users] chanspy spies on wrong channel
The argument to chanspy is a pattern and not an exact match.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 2, 2011, at 3:48 PM, steve casto wrote:
> asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use
> flash operator panel < 2.0
>
> (from extensions.conf)
> exten=> 304,1,ChanSpy(Zap/4|q)
> exten=> 304,2,hangup
> There is no entry ChanSpy(Zap/41) in extensions.conf
>
> On dialing 304 and Zap/41 is in use this happens:
> [Jul 1 18:24:47] VERBOSE[14447] logger.c: -- Executing
> [304@flash:1] ChanSpy("Zap/31-1", "Zap/4|q") in new stack
> [Jul 1 18:24:47] VERBOSE[14447] logger.c: == Spying on channel Zap/41-1
> [Jul 1 18:24:47] NOTICE[14447] app_chanspy.c: Attaching Zap/31-1 to
> Zap/41-1
>
> If while spying on Zap/41 that channel is hung up:
> [Jul 1 19:06:48] VERBOSE[15242] logger.c: == Done Spying on channel
> Zap/41-1
> [Jul 1 19:06:48] VERBOSE[15242] logger.c: == Spying on channel Zap/4-1
> [Jul 1 19:06:48] NOTICE[15242] app_chanspy.c: Attaching Zap/31-1 to Zap/4-1
>
> thanks list
> 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
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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