Re: [asterisk-users] how to set to make a call through a fixed ip on a 2 ips server?
Hi, I don't think there is a way for it inside asterisk but you achieve it by adding static route in Linux routing table and make interface having that IP as default route for the interested IPs traffic. Regards, Faisal From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cnasterisk Sent: Monday, July 04, 2011 10:40 AM To: asterisk-users Subject: [asterisk-users] how to set to make a call through a fixed ip on a 2 ips server? Hi all, I have a server runing asterisk 1.8, and the server has 2 different ip address if i want to make a call from a sip trunk with a fixed ip from the 2 ips, how to do? 2011-07-04 _ cnasterisk -- _ -- 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] SIP Peer Name Variable
When you make a call asterisk always create a channel named as below,
CheannelType/PeerName-uniquecode
Like
SIP/jon-312abf
So here jon is the peer name. This can help you to identify a peer as long
as A-Leg is active.
Regards,
Faisal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Sunday, July 03, 2011 6:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: 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 item
may be available from the channel driver; see its documentation for details.
Any item 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'
[asterisk-users] Mixmonitor concept's question
[RecordPrompts]
exten = ,1,Answer()
exten = ,n,NoOp(WelCome to conference section)
exten = ,n,Playback(ConfDemoWC)
exten =
,n,MixMonitor(tmp/00Record/-${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}.wav,ab)
exten = ,n,Konference(${EXTEN},ADRSV)
Hi
My basic doubt is that if 1 or more person call extension then
recording will be started by asterisk Mixmonitor application.
So basic question will come into mind that all calls will start recording it
means more then 1 thread will start by asterisk for only recording purpose
but finally 1 file will be save at disk.
Am I right ?
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
_
-- 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] how to set to make a call through a fixed ipon a 2 ips server?
thanks, Faisal Hanif! i will try it! 2011-07-04 cnasterisk 发件人： Faisal Hanif 发送时间： 2011-07-04 15:59:34 收件人： 'Asterisk Users Mailing List - Non-Commercial Discussion' 抄送： 主题： Re: [asterisk-users] how to set to make a call through a fixed ipon a 2 ips server? Hi, I don’t think there is a way for it inside asterisk but you achieve it by adding static route in Linux routing table and make interface having that IP as default route for the interested IPs traffic. Regards, Faisal From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cnasterisk Sent: Monday, July 04, 2011 10:40 AM To: asterisk-users Subject: [asterisk-users] how to set to make a call through a fixed ip on a 2 ips server? Hi all, I have a server runing asterisk 1.8, and the server has 2 different ip address if i want to make a call from a sip trunk with a fixed ip from the 2 ips, how to do? 2011-07-04 cnasterisk -- _ -- 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
Just to be sure that I am working in the right direction. To do ACD, then I have to configure queue.conf and agent.conf? One more question: if the agent needs to be in the NotReady state, then how this can be acheived? Regards Bilal - FreeBPX calls them Ring Groups, you can look in to that. Or you could use a small ACD group. 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
[asterisk-users] stream rtp from asterisk
Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus -- _ -- 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] stream rtp from asterisk
On 07/04/2011 06:58 AM, Marcus Kvarsell wrote: Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Can you clarify what you mean by streaming? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] stream rtp from asterisk
Sending the rtp-data to external server. One example which I have not gotten to work is this below: http://oreka.sourceforge.net/ September 02, 2009: Asterisk interception via Xorcom Asterisk patch Added support for recording of Asterisk voice calls (TDM and IP) using Xorcoms Asterisk patch. See here. If there is any folk out there that has knowledge of this or any similar software I would be very happy if you could help me get this to work. / Marcus -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Alex Balashov Skickat: den 4 juli 2011 14:49 Till: asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] stream rtp from asterisk On 07/04/2011 06:58 AM, Marcus Kvarsell wrote: Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Can you clarify what you mean by streaming? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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] Mixmonitor concept's question
On Monday, July 04, 2011 05:10:43 AM virendra bhati wrote:
[RecordPrompts]
exten = ,1,Answer()
exten = ,n,NoOp(WelCome to conference section)
exten = ,n,Playback(ConfDemoWC)
exten =
,n,MixMonitor(tmp/00Record/-${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}.w
av,ab) exten = ,n,Konference(${EXTEN},ADRSV)
Hi
My basic doubt is that if 1 or more person call extension then
recording will be started by asterisk Mixmonitor application.
So basic question will come into mind that all calls will start recording
it means more then 1 thread will start by asterisk for only recording
purpose but finally 1 file will be save at disk.
Am I right ?
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
Hi Virendra,
It has been my experience that if two or more calls happen in the same second
(which is the least identifier you are using to make the file unique) then the
result is that only one file is created. (the other(s) are not)
To solve that issue, you can use ${UNIQUEID}, something like:
exten = 6000,63,Set(CALLTIME=${STRFTIME(${EPOCH},,
%C%y%m%d%H%M%S)}_${UNIQUEID})
exten = 6000,64,Set(CALLFILENAME=6000-${CALLTIME})
exten = 6000,65,Set(CMD='/opt/lame/l.sh '${CALLFILENAME})
exten = 6000,66,MixMonitor(${CALLFILENAME}.wav|v(2)V(0)|${CMD})
Note that the 1st exten line wraps here in the email but it should be all on
one line.
earl
--
_
-- 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] Mixmonitor concept's question
Hi
Your suggestion is right if we want different recording for all channels.
But my problem is that I want to know if more user call the same conference
at different time gape(difference) then mixmonitor will take single asterisk
thread for recording or multiple thread for recoding.
On Mon, Jul 4, 2011 at 7:13 PM, Earl e...@micpc.com wrote:
On Monday, July 04, 2011 05:10:43 AM virendra bhati wrote:
[RecordPrompts]
exten = ,1,Answer()
exten = ,n,NoOp(WelCome to conference section)
exten = ,n,Playback(ConfDemoWC)
exten =
,n,MixMonitor(tmp/00Record/-${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}.w
av,ab) exten = ,n,Konference(${EXTEN},ADRSV)
Hi
My basic doubt is that if 1 or more person call extension then
recording will be started by asterisk Mixmonitor application.
So basic question will come into mind that all calls will start recording
it means more then 1 thread will start by asterisk for only recording
purpose but finally 1 file will be save at disk.
Am I right ?
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
Hi Virendra,
It has been my experience that if two or more calls happen in the same
second
(which is the least identifier you are using to make the file unique) then
the
result is that only one file is created. (the other(s) are not)
To solve that issue, you can use ${UNIQUEID}, something like:
exten = 6000,63,Set(CALLTIME=${STRFTIME(${EPOCH},,
%C%y%m%d%H%M%S)}_${UNIQUEID})
exten = 6000,64,Set(CALLFILENAME=6000-${CALLTIME})
exten = 6000,65,Set(CMD='/opt/lame/l.sh '${CALLFILENAME})
exten = 6000,66,MixMonitor(${CALLFILENAME}.wav|v(2)V(0)|${CMD})
Note that the 1st exten line wraps here in the email but it should be all
on
one line.
earl
--
_
-- 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
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
_
-- 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] stream rtp from asterisk
On 2011-07-04 15:07, Marcus Kvarsell wrote: Sending the rtp-data to external server. One example which I have not gotten to work is this below: http://oreka.sourceforge.net/ September 02, 2009: Asterisk interception via Xorcom Asterisk patch Added support for recording of Asterisk voice calls (TDM and IP) using Xorcoms Asterisk patch. See here. If there is any folk out there that has knowledge of this or any similar software I would be very happy if you could help me get this to work. / Marcus http://oreka.sourceforge.net/oreka-user-manual.html#gettingvoiptraffic Seems they propose setting the switch in mirror/monitoring-mode and sniff the traffic on another server. Normal managed and smart-switches support this option... Or you can install the software on the asterisk server. /Johan -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Alex Balashov Skickat: den 4 juli 2011 14:49 Till: asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] stream rtp from asterisk On 07/04/2011 06:58 AM, Marcus Kvarsell wrote: Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Can you clarify what you mean by streaming? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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 -- Med vänlig hälsning Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- 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] RINGNOANSWER events in queue log
Does anyone know why i would get this RINGNOANSWER events in queue_log when clearly the agent is busy and call-waiting is disabled. 1309550595|1309550570.399965|2253|Local/05@from-internal/n|CONNECT|2|1309550593.399966|0 1309550632|1309550533.399961|2253|Local/11@from-internal/n|COMPLETECALLER|1|74|1 1309550663|1309550640.399969|2253|NONE|ENTERQUEUE||zz 1309550666|1309550640.399969|2253|Local/01@from-internal/n|CONNECT|3|1309550663.399971|0 //here it looks like Agent01 got the call. 1309550671|1309550648.399970|2525|NONE|ENTERQUEUE||zzz 1309550671|1309550648.399970|2525|Local/05@from-internal/n|RINGNOANSWER|0 1309550671|1309550648.399970|2525|Local/01@from-internal/n|RINGNOANSWER|0 // why is the system trying that channel for agent01 again? -- _ -- 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] Testing Asterisk with media - sipp
I'm trying to get working SIPp with media but something is wrong (it's working well without media), please help: This is the command I send at SIPp server: ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err This is the result I see: Last Error: Aborting call on unexpected message for Call-Id '19-12768@12... What I see at sipp's logs: 2011-06-28 14:32:57:6241309289577.624809: Aborting call on unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100' (index 1), received 'SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 127.0.0.1:5061 ;branch=z9hG4bK-12768-1-0;received=192.168.1.253 From: sipp sip:sipp@127.0.0.1:5061;tag=12768SIPpTag091 To: sut sip:2005@192.168.1.18:5060;tag=as3614adc3 Call-ID: 1-12768@127.0.0.1 CSeq: 1 INVITE Server: Asterisk PBX 1.8.4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 This is my asterisk 1.8's configuration: *sip.conf* [sipp] type=friend context=sipp host=dynamic port=6000 user=sipp canreinvite=no disallow=all allow=ulaw * * *extensions.conf:* [sipp] exten = 2005,1,Answer same=n,Dial(SIP/intern,30) same=n,Hangup() exten = 2006,1,Answer() same= n,WaitMusicOnHold(4) same= n,Hangup() I'm using sipp.3.1.src.tar.gz and I have installed it this way: ..sip.svn# make pcapplay Thanks in advance. Elder -- _ -- 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] Testing Asterisk with media - sipp
488 means no mutually acceptable codecs were negotiated between the endpoints. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 4, 2011, at 3:29 PM, Daniel - Asterisk earohua...@gmail.com wrote: I'm trying to get working SIPp with media but something is wrong (it's working well without media), please help: This is the command I send at SIPp server: ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err This is the result I see: Last Error: Aborting call on unexpected message for Call-Id '19-12768@12... What I see at sipp's logs: 2011-06-28 14:32:57:6241309289577.624809: Aborting call on unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100' (index 1), received 'SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-12768-1-0;received=192.168.1.253 From: sipp sip:sipp@127.0.0.1:5061;tag=12768SIPpTag091 To: sut sip:2005@192.168.1.18:5060;tag=as3614adc3 Call-ID: 1-12768@127.0.0.1 CSeq: 1 INVITE Server: Asterisk PBX 1.8.4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 This is my asterisk 1.8's configuration: sip.conf [sipp] type=friend context=sipp host=dynamic port=6000 user=sipp canreinvite=no disallow=all allow=ulaw extensions.conf: [sipp] exten = 2005,1,Answer same=n,Dial(SIP/intern,30) same=n,Hangup() exten = 2006,1,Answer() same= n,WaitMusicOnHold(4) same= n,Hangup() I'm using sipp.3.1.src.tar.gz and I have installed it this way: ..sip.svn# make pcapplay Thanks in advance. Elder -- _ -- 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] Agent Login, Logout, Ready, Not Ready from the CTI application
Hi All; We know that agents can login and logout from the phone handset. But if we need the login, logout, ready and not ready to be from an application and to be integrated with the CRM, how to acheive this? Normally in Cisco and AVAYA, they use CTI integration and the CTI client (which is embded in the CRM application) will receive the the caller id or information via that CTI client. How this to be done in Asterisk? By the way: is the ready and not ready in Asterisk? 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] Agent Login, Logout, Ready, Not Ready from the CTI application
You need to use the AMI interface an deal with the events that are give to you. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 4, 2011, at 4:36 PM, bilal ghayyad wrote: Hi All; We know that agents can login and logout from the phone handset. But if we need the login, logout, ready and not ready to be from an application and to be integrated with the CRM, how to acheive this? Normally in Cisco and AVAYA, they use CTI integration and the CTI client (which is embded in the CRM application) will receive the the caller id or information via that CTI client. How this to be done in Asterisk? By the way: is the ready and not ready in Asterisk? 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
Re: [asterisk-users] Testing Asterisk with media - sipp
Thank you Alex, It's running without errors now and I can see the media flowing with 'rtp set debug on' but I can't still hear anything on the Asterisk's peers, any advice? Elder 2011/7/4, Alex Balashov abalas...@evaristesys.com: 488 means no mutually acceptable codecs were negotiated between the endpoints. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 4, 2011, at 3:29 PM, Daniel - Asterisk earohua...@gmail.com wrote: I'm trying to get working SIPp with media but something is wrong (it's working well without media), please help: This is the command I send at SIPp server: ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err This is the result I see: Last Error: Aborting call on unexpected message for Call-Id '19-12768@12... What I see at sipp's logs: 2011-06-28 14:32:57:6241309289577.624809: Aborting call on unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100' (index 1), received 'SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-12768-1-0;received=192.168.1.253 From: sipp sip:sipp@127.0.0.1:5061;tag=12768SIPpTag091 To: sut sip:2005@192.168.1.18:5060;tag=as3614adc3 Call-ID: 1-12768@127.0.0.1 CSeq: 1 INVITE Server: Asterisk PBX 1.8.4.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 This is my asterisk 1.8's configuration: sip.conf [sipp] type=friend context=sipp host=dynamic port=6000 user=sipp canreinvite=no disallow=all allow=ulaw extensions.conf: [sipp] exten = 2005,1,Answer same=n,Dial(SIP/intern,30) same=n,Hangup() exten = 2006,1,Answer() same= n,WaitMusicOnHold(4) same= n,Hangup() I'm using sipp.3.1.src.tar.gz and I have installed it this way: ..sip.svn# make pcapplay Thanks in advance. Elder -- _ -- 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 -- Enviado desde mi dispositivo móvil -- _ -- 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] DTMF between sip trunks and PRIs
Hi, I'm looking for some advice on how to solve DTMF issues. I have 2 boxes, one which is the connection to the PSTN (PSTN) through PRIs and SIP trunks, and a second (PBX) which has UAs registered to it. We have a customer that has an existing pbx that we trunk analog lines to using a GXW-4008. The GXW is set to dtmfmode inband. This seems to provide the best outbound DTMF. The issue I'm currently having is with inbound DTMF. PBX and PSTN are connected through a standard sip trunk. Both machines are on the same physical switch. Here are the results I've seen: PBX - PSTN using rfc2833 | Incoming call on PRI | DTMF on pbx voicemail system fails (dup/missing digits) PBX - PSTN using inband | Incoming call on PRI | DTMF on pbx voicemail system is correct PBX - PSTN using rfc2833 | Incoming call on SIP | DTMF on pbx voicemail system is correct PBX - PSTN using inband | Incoming call on SIP | DTMF on pbx voicemail system is correct All asterisk versions are 1.4.35. PRI card is a Sangoma A104 with HW DTMF detection. Does asterisk just have a problem converting the DTMF from the D-channel to rfc2833? The DTMF log looks ok (I dialed '642'), so I'm not sure where the issue is coming in. [Jul 4 21:05:44] DTMF[9769] channel.c: DTMF begin '6' received on Zap/15-1 [Jul 4 21:05:44] DTMF[9769] channel.c: DTMF begin passthrough '6' on Zap/15-1 [Jul 4 21:05:44] DTMF[9769] channel.c: DTMF end '6' received on Zap/15-1, duration 100 ms [Jul 4 21:05:44] DTMF[9769] channel.c: DTMF end accepted with begin '6' on Zap/15-1 [Jul 4 21:05:44] DTMF[9769] channel.c: DTMF end passthrough '6' on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF begin '4' received on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF begin passthrough '4' on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF end '4' received on Zap/15-1, duration 100 ms [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF end accepted with begin '4' on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF end passthrough '4' on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF begin '2' received on Zap/15-1 [Jul 4 21:05:45] DTMF[9769] channel.c: DTMF begin passthrough '2' on Zap/15-1 [Jul 4 21:05:46] DTMF[9769] channel.c: DTMF end '2' received on Zap/15-1, duration 100 ms [Jul 4 21:05:46] DTMF[9769] channel.c: DTMF end accepted with begin '2' on Zap/15-1 [Jul 4 21:05:46] DTMF[9769] channel.c: DTMF end passthrough '2' on Zap/15-1 Thanks. -- James -- _ -- 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] Blind Transfer Connected
Hi all In asterisk if blind transfer failed ,call is not connecting back . For Eg: A make call to B through asterisk,then B transfer the call to C. If C did not answer the call ,A and B Call should connect back.But this is not happening with asterisk(A and B call is disconnecting). Does anyone knows about this? 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
