[asterisk-users] Talk detection in meetme
Hi, I create Chat room with MEETME and now I have a problem. I want that the host of the room could identify the participants in the room by their speech, so that if a participant uses language the host could kick him from the room. Is there a way to do it? thanks. Eyal Mahalal -- _ -- 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] Populate CDR issues
Hello Everyone,
I didn't get a reply to my problem below so I'm posting again just in case
someone who might be able to help missed my previous post.
Thank You...
*
Hello list,
I'm trying to populate my CDR logs with values which are available after the
call has started (e.g. signalling IP of remote user, media IP, codec etc.).
While CHANNEL function give me all I need for the incoming leg (leg A), I can't
get the relevant values for the outgoing channel. I've tried using the option
'U' with my dial command (execute subroutine for called channel after called
channel answered but before the call is bridged). While this throws the correct
information to the console it does not populate the CDRs accordingly.
Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC
and the table therein contains the relevant fields.
This is the console with 'very-verbose' output for the 'Dial' application where
office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP
192.168.20.226. My comments added prefixed by ** and on separate line:
** channel here is source channel: SIP/office_Admin2-0015
[Dec 1 12:14:31] -- Executing [316@InternalDP:5]
Dial(SIP/office_Admin2-0015, SIP/office_ServerRoom,,FgU(jump2SetVar))
in new stack
[Dec 1 12:14:31] == Using UDPTL CoS mark 5
[Dec 1 12:14:31] == Using SIP RTP CoS mark 5
[Dec 1 12:14:31] -- Called SIP/office_ServerRoom
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:33] -- SIP/office_ServerRoom-0016 answered
SIP/office_Admin2-0015
** from here the channel is the destination channel:
SIP/office_ServerRoom-0016
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:1]
Gosub(SIP/office_ServerRoom-0016, SetVar,postdial,1) in new stack
** This is how I obtain channel information:
** exten =
postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peername)},port)})
; resulting format: a.b.c.d:port
** same = n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same = n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec 1 12:14:33] -- Executing [postdial@SetVar:1]
Set(SIP/office_ServerRoom-0016, CDR(chanoutsigip)=192.168.20.226:5065)
in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:2]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutmediaip)=192.168.20.226:23008) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:3]
Set(SIP/office_ServerRoom-0016, CDR(chanoutcodec)=g729) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:4]
Goto(SIP/office_ServerRoom-0016, endsub,1) in new stack
[Dec 1 12:14:33] -- Goto (SetVar,endsub,1)
[Dec 1 12:14:33] -- Executing [endsub@SetVar:1]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:2]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1]
NoOp(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Auto fallthrough, channel
'SIP/office_ServerRoom-0016' status is 'UNKNOWN'
[Dec 1 12:14:33] -- Remotely bridging SIP/office_Admin2-0015 and
SIP/office_ServerRoom-0016
When call is terminated the relevant fields in the database for
CDR(chanoutsigip), CDR(chanoutmediaip) and CDR(chanoutcodec) are populated with
their default values (typically blank or '-') and NOT with the values above.
Am I doing something wrong or is there a different way to populate CDR's with
info from called channel (leg B)?
Thank you for your replies...
Harel
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[asterisk-users] help please (D 300 JCT)
i need to know how can i configure a D 300JCT with asterisk, i want to connect two PBX where each one have this card on it,i really need your help as soon as possible. i already done some file configuration system.conf and in chan_dahdi.conf and i have installed the DAHDI and the LibPri modules. -- * HARAZ Tahar * *Engineering Student at the National Institute for Posts and Telecommunications (INPT) * * Phone: +212 6 78030050 E-mail: harazta...@gmail.com ouabimedcha...@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] Populate CDR issues
Le 06/12/2011 10:16, Harel Cohen a écrit : Hello Everyone, Hi Harel I didn’t get a reply to my problem below so I’m posting again just in case someone who might be able to help missed my previous post. Thank You… Please take a look at issue ASTERISK-18875 https://issues.asterisk.org/jira/browse/ASTERISK-18875 [...] -- Daniel -- _ -- 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] Simple Generic IVR to get us up an running Quick
On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote: Hello Everyone, Are there any descent generic IVR recordings, that we can use to quickly get our PBX up and running? It will obviously not include the company name. It's easy enough to make your own recordings. Word of caution though. It might be advisable to ask somebody outside the company to record the phrases, Wonder why? At home i did it my self, and i still hear people stating that they have been talking at me, totaly unaware that it was just the voicemail anouncements. Peope just hear a voice, but seldom listen. And not just 90-old aunts, But people from helpdesks and even CEO's. Sometimes i wonder, if i should ask/test the callers I.Q. , And adapt the IVR's accordingly ;=) hw -- _ -- 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 use Hints in asterisk
Hi All, I read about the *Hint* in asterisk. I want to implements into my server for testing purpose. How to use it ? please help me... -- Thanks and regards Virendra Bhati +91-8885268942 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
[asterisk-users] Data provided by pri show spans and dahdi_tool do not match
Hello, On a 4 BRI Euro-ISDN TE/PtmP production system (Asterisk 1.6.1.18, Dahdi 2.5.0, Libpri 1.4.12), I sudently got the console cluttered with messages like this: PRI got event: HDLC Abort(6) on Primary D-Channel of span foo Then a flow of messages (twice per second) like this: PRI got event: HDLC Overrun(7) on Primary D-Channel of span 2 Contrary to the first one which concerned all of the 4 spans, the second message type only concerned one span (span2). So I stopped dahdi and asterisk, disabled span 2 and restarted everything. No I see : PRI span 1/0: Provisioned, Up, Active PRI span 3/0: Provisioned, In Alarm, Up, Active PRI span 4/0: Provisioned, Up, Active With dahdi_tool main screen, I can exactly read Alarms Span OK HA8- UNCONFIGURED HA8- OK HA8- REDHA8- With pri intensive debug span 3, I'm getting: [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: t203_expire [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: TEI: 75 State 7(Multi-frame established) [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: V(A)=3, V(S)=3, V(R)=1 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: T200_id=0, N200=3, T203_id=0 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: [ 00 97 01 03 ] [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: Supervisory frame: [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: SAPI: 00 C/R: 0 EA: 0 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: TEI: 075EA: 1 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: Zero: 0 S: 0 01: 1 [ RR (receive ready) ] [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: N(R): 001 P/F: 1 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: 0 bytes of data [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: -- Starting T200 timer [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: TEI: 75 State 8(Timer recovery) [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: V(A)=3, V(S)=3, V(R)=1 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: T200_id=16384, N200=3, T203_id=0 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: [ 02 97 01 07 ] [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: Supervisory frame: [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: SAPI: 00 C/R: 1 EA: 0 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: TEI: 075EA: 1 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: Zero: 0 S: 0 01: 1 [ RR (receive ready) ] [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: N(R): 003 P/F: 1 [Dec 6 10:32:58] VERBOSE[11461] chan_dahdi.c: 0 bytes of data With dahdi_scan : dahdi_scan [1] active=yes alarms=OK description=HA8- name=WCBRI/0/0 manufacturer=Digium devicetype=HA8- location=PCI Bus 04 Slot 07 basechan=1 totchans=3 irq=20 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS [2] active=yes alarms=UNCONFIGURED description=HA8- name=WCBRI/0/1 manufacturer=Digium devicetype=HA8- location=PCI Bus 04 Slot 07 basechan=4 totchans=3 irq=20 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding= framing=CAS [3] active=yes alarms=OK description=HA8- name=WCBRI/0/2 manufacturer=Digium devicetype=HA8- location=PCI Bus 04 Slot 07 basechan=7 totchans=3 irq=20 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS [4] active=yes alarms=RED description=HA8- name=WCBRI/0/3 manufacturer=Digium devicetype=HA8- location=PCI Bus 04 Slot 07 basechan=10 totchans=3 irq=20 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS I'm trying to understand what is going on. My questions are: 1. Am I correct observing span status provided by pri show spans and dahdi_tool, differ: pri show spans says span 3 is in Alarm while dahdi_tool (and dahdi_scan) say that span 4 is in Alarm. 2. Which tool would give me the most accurate reason why a Dahdi span is in RED Alarm ? 3. Does the pri intensive debug span 3 output above really belongs to span 3 and to an In Alarm span ? 4. Suggestions ? Comments ? Regards -- _ -- 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] Hint'ing with XMPP?
2011/12/5 Jamie A. Stapleton jstaple...@computer-business.com I have not ever done what you are talking about. ** ** However, I can tell you that our Openfire XMPP server has similar functionality because of their Asterisk-IM Plugin. Are you currently using it ? With which asterisk version ? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jay R. Worthington *Sent:* Saturday, December 03, 2011 8:11 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Hint'ing with XMPP? ** ** Hiya, can i use an XMPP Client to see the presence of a hint? I have configured asterisk in component-mode, seem's to work, but all users ( xmpp:1...@asterisk.dohmain.com are online, even if 123 isn't a configured hint). Any good howto's out there, all the stuff on voip-info.org is completely outdated, i'm using asterisk 10... Regards Jay -- _ -- 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] A new hack?
On Mon, 2011-12-05 at 18:51 -0800, Steve Edwards wrote: snip Your security needs depends on your environment. At this point in time, all of the hosts I manage for my clients exist in very limited environments and have very small attack surfaces. They are racked in secure data centers. They only accept SIP from clients with static IP addresses that we have an existing business relationship with. They only accept SSH connections from me. They only accept HTTP connections from me and my boss. That's about it. I don't see where F2B adds much value for me. *) Lots of admins think they can't limit access to servers because they have 'mobile' users. Your users probably don't need to access your servers from every single place on the Internet. If your users don't come from China, North Korea, Iran, etc, you can block entire regions with a few rules and eliminate 80% of probes and attacks from reaching your servers in the first place. Apologies in advance if you happen to live in some of these regions -- feel free to `s/China, North Korea, Iran/United States, Canada, England/g` Perhaps an other suggestion. If they are true road warriors, i presume they are capable of setting up an vpn to the company. In that case, only allow registrations/calls through the secured tunnel. Then it's not any concern to asterisk. And if they can breach your tunnel, you have something else to worry about. hw -- _ -- 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 use Hints in asterisk
Hi All, I did some google and found some documents on that and finally I got some response from asterisk . Below is the CLI output of my google. *haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:IdleWatchers 0 1 hint matching extension 2218 == Using SIP RTP CoS mark 5 -- Executing [222@bhati-test:1] NoOp(SIP/2218-02c3, Call from Gtalk ) in new stack -- Executing [222@bhati-test:2] NoOp(SIP/2218-02c3, Hint for Extension 2218 is ) in new stack -- Executing [222@bhati-test:3] Set(SIP/2218-02c3, CALLERID(name)=From Google Talk) in new stack -- Executing [222@bhati-test:4] Wait(SIP/2218-02c3, 10) in new stack haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:InUse Watchers 0 1 hint matching extension 2218 -- Executing [222@bhati-test:5] Dial(SIP/2218-02c3, SIP/my_sip_phones) in new stack == Using SIP RTP CoS mark 5 [Dec 6 17:39:38] WARNING[6689]: chan_sip.c:5331 create_addr: No such host: my_sip_phones [Dec 6 17:39:38] WARNING[6689]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/2218-02c3' status is 'CHANUNAVAIL' -- Executing [h@bhati-test:1] NoOp(SIP/2218-02c3, hangup the call now) in new stack haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:Idle Watchers 0 1 hint matching extension 2218 * *Is this the right way to use HINT of asterisk ?* On Tue, Dec 6, 2011 at 3:38 PM, virendra bhati virbh...@gmail.com wrote: Hi All, I read about the *Hint* in asterisk. I want to implements into my server for testing purpose. How to use it ? please help me... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- Thanks and regards Virendra Bhati +91-8885268942 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 use Hints in asterisk
Hello, AFAIK Hints are used for looking out for a device state before actually doing anything. Hints can be used from Dial-plan, AMI, or AGI. An example can be to look for state of a SIP user. Read these links for better understanding. http://www.smartvox.co.uk/astfaq_subscribe_hints.htm http://www.voip-info.org/wiki/view/Asterisk+standard+extensions Regards, Sammy. On Tue, Dec 6, 2011 at 5:14 PM, virendra bhati virbh...@gmail.com wrote: Hi All, I did some google and found some documents on that and finally I got some response from asterisk . Below is the CLI output of my google. *haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:IdleWatchers 0 1 hint matching extension 2218 == Using SIP RTP CoS mark 5 -- Executing [222@bhati-test:1] NoOp(SIP/2218-02c3, Call from Gtalk ) in new stack -- Executing [222@bhati-test:2] NoOp(SIP/2218-02c3, Hint for Extension 2218 is ) in new stack -- Executing [222@bhati-test:3] Set(SIP/2218-02c3, CALLERID(name)=From Google Talk) in new stack -- Executing [222@bhati-test:4] Wait(SIP/2218-02c3, 10) in new stack haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:InUse Watchers 0 1 hint matching extension 2218 -- Executing [222@bhati-test:5] Dial(SIP/2218-02c3, SIP/my_sip_phones) in new stack == Using SIP RTP CoS mark 5 [Dec 6 17:39:38] WARNING[6689]: chan_sip.c:5331 create_addr: No such host: my_sip_phones [Dec 6 17:39:38] WARNING[6689]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/2218-02c3' status is 'CHANUNAVAIL' -- Executing [h@bhati-test:1] NoOp(SIP/2218-02c3, hangup the call now) in new stack haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:Idle Watchers 0 1 hint matching extension 2218 * *Is this the right way to use HINT of asterisk ?* On Tue, Dec 6, 2011 at 3:38 PM, virendra bhati virbh...@gmail.com wrote: Hi All, I read about the *Hint* in asterisk. I want to implements into my server for testing purpose. How to use it ? please help me... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- Thanks and regards Virendra Bhati +91-8885268942 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 use Hints in asterisk
Hi All,
Below bold application gives the correct information with asterisk
*HINT*function.
exten = 222,1,NoOp( Call from Gtalk )
*same = n,NoOp(My phone state is currently
${DEVICE_STATE(SIP/2218)})*
same = n,Set(CALLERID(name)=From Google Talk)
same = n,Wait(10)
same = n,Dial(SIP/my_sip_phones)
Spatially thanks for Sammy who give me the way to get success on that way.
On Tue, Dec 6, 2011 at 5:58 PM, Sammy Govind govoi...@gmail.com wrote:
Hello,
AFAIK Hints are used for looking out for a device state before actually
doing anything. Hints can be used from Dial-plan, AMI, or AGI. An example
can be to look for state of a SIP user.
Read these links for better understanding.
http://www.smartvox.co.uk/astfaq_subscribe_hints.htm
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
Regards,
Sammy.
On Tue, Dec 6, 2011 at 5:14 PM, virendra bhati virbh...@gmail.com wrote:
Hi All,
I did some google and found some documents on that and finally I got some
response from asterisk . Below is the CLI output of my google.
*haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:IdleWatchers 0
1 hint matching extension 2218
== Using SIP RTP CoS mark 5
-- Executing [222@bhati-test:1] NoOp(SIP/2218-02c3, Call
from Gtalk ) in new stack
-- Executing [222@bhati-test:2] NoOp(SIP/2218-02c3, Hint for
Extension 2218 is ) in new stack
-- Executing [222@bhati-test:3] Set(SIP/2218-02c3,
CALLERID(name)=From Google Talk) in new stack
-- Executing [222@bhati-test:4] Wait(SIP/2218-02c3, 10) in
new stack
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:InUse Watchers 0
1 hint matching extension 2218
-- Executing [222@bhati-test:5] Dial(SIP/2218-02c3,
SIP/my_sip_phones) in new stack
== Using SIP RTP CoS mark 5
[Dec 6 17:39:38] WARNING[6689]: chan_sip.c:5331 create_addr: No such
host: my_sip_phones
[Dec 6 17:39:38] WARNING[6689]: app_dial.c:1747 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/2218-02c3' status is
'CHANUNAVAIL'
-- Executing [h@bhati-test:1] NoOp(SIP/2218-02c3, hangup the
call now) in new stack
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218 State:Idle
Watchers 0
1 hint matching extension 2218
*
*Is this the right way to use HINT of asterisk ?*
On Tue, Dec 6, 2011 at 3:38 PM, virendra bhati virbh...@gmail.comwrote:
Hi All,
I read about the *Hint* in asterisk. I want to implements into my
server for testing purpose. How to use it ? please help me...
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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Re: [asterisk-users] How to use Hints in asterisk
Hi All,
If you used *DEVICE_STATE *function then there is no need to used *HINT* it
work independently.
It's not become to confusion for me how to when to used *HINT *and
when *DEVICE_STATE
?
*
On Tue, Dec 6, 2011 at 6:20 PM, virendra bhati virbh...@gmail.com wrote:
Hi All,
Below bold application gives the correct information with asterisk
*HINT*function.
exten = 222,1,NoOp( Call from Gtalk )
*same = n,NoOp(My phone state is currently
${DEVICE_STATE(SIP/2218)})*
same = n,Set(CALLERID(name)=From Google Talk)
same = n,Wait(10)
same = n,Dial(SIP/my_sip_phones)
Spatially thanks for Sammy who give me the way to get success on that way.
On Tue, Dec 6, 2011 at 5:58 PM, Sammy Govind govoi...@gmail.com wrote:
Hello,
AFAIK Hints are used for looking out for a device state before actually
doing anything. Hints can be used from Dial-plan, AMI, or AGI. An example
can be to look for state of a SIP user.
Read these links for better understanding.
http://www.smartvox.co.uk/astfaq_subscribe_hints.htm
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
Regards,
Sammy.
On Tue, Dec 6, 2011 at 5:14 PM, virendra bhati virbh...@gmail.comwrote:
Hi All,
I did some google and found some documents on that and finally I got
some response from asterisk . Below is the CLI output of my google.
*haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:IdleWatchers 0
1 hint matching extension 2218
== Using SIP RTP CoS mark 5
-- Executing [222@bhati-test:1] NoOp(SIP/2218-02c3, Call
from Gtalk ) in new stack
-- Executing [222@bhati-test:2] NoOp(SIP/2218-02c3, Hint for
Extension 2218 is ) in new stack
-- Executing [222@bhati-test:3] Set(SIP/2218-02c3,
CALLERID(name)=From Google Talk) in new stack
-- Executing [222@bhati-test:4] Wait(SIP/2218-02c3, 10) in
new stack
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:InUse Watchers 0
1 hint matching extension 2218
-- Executing [222@bhati-test:5] Dial(SIP/2218-02c3,
SIP/my_sip_phones) in new stack
== Using SIP RTP CoS mark 5
[Dec 6 17:39:38] WARNING[6689]: chan_sip.c:5331 create_addr: No such
host: my_sip_phones
[Dec 6 17:39:38] WARNING[6689]: app_dial.c:1747 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/2218-02c3' status is
'CHANUNAVAIL'
-- Executing [h@bhati-test:1] NoOp(SIP/2218-02c3, hangup the
call now) in new stack
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:Idle
Watchers 0
1 hint matching extension 2218
*
*Is this the right way to use HINT of asterisk ?*
On Tue, Dec 6, 2011 at 3:38 PM, virendra bhati virbh...@gmail.comwrote:
Hi All,
I read about the *Hint* in asterisk. I want to implements into my
server for testing purpose. How to use it ? please help me...
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] How to use Hints in asterisk
Try this link - I think they describe it better than I would
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
From: virendra bhati [mailto:virbh...@gmail.com]
Sent: Tuesday, December 06, 2011 7:28 AM
To: Danny Nicholas
Subject: Re: How to use Hints in asterisk
thank you Danny, I want to know how to configure HINT in dialplan. please
give me some clue about it
On Tue, Dec 6, 2011 at 6:52 PM, Danny Nicholas da...@debsinc.com wrote:
I'm sure this is overly generic but AFAIK DEVICE_STATE is for use with the
active dialplan and HINT can be used in the dialplan or externally with
AMI/AGI, etc.
From: virendra bhati [mailto:virbh...@gmail.com]
Sent: Tuesday, December 06, 2011 6:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas
Cc: Sammy Govind
Subject: Re: How to use Hints in asterisk
Hi All,
If you used DEVICE_STATE function then there is no need to used HINT it work
independently.
It's not become to confusion for me how to when to used HINT and when
DEVICE_STATE ?
On Tue, Dec 6, 2011 at 6:20 PM, virendra bhati virbh...@gmail.com wrote:
Hi All,
Below bold application gives the correct information with asterisk HINT
function.
exten = 222,1,NoOp( Call from Gtalk )
same = n,NoOp(My phone state is currently
${DEVICE_STATE(SIP/2218)})
same = n,Set(CALLERID(name)=From Google Talk)
same = n,Wait(10)
same = n,Dial(SIP/my_sip_phones)
Spatially thanks for Sammy who give me the way to get success on that way.
On Tue, Dec 6, 2011 at 5:58 PM, Sammy Govind govoi...@gmail.com wrote:
Hello,
AFAIK Hints are used for looking out for a device state before actually
doing anything. Hints can be used from Dial-plan, AMI, or AGI. An example
can be to look for state of a SIP user.
Read these links for better understanding.
http://www.smartvox.co.uk/astfaq_subscribe_hints.htm
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
Regards,
Sammy.
On Tue, Dec 6, 2011 at 5:14 PM, virendra bhati virbh...@gmail.com wrote:
Hi All,
I did some google and found some documents on that and finally I got some
response from asterisk . Below is the CLI output of my google.
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:IdleWatchers 0
1 hint matching extension 2218
== Using SIP RTP CoS mark 5
-- Executing [222@bhati-test:1] NoOp(SIP/2218-02c3, Call from
Gtalk ) in new stack
-- Executing [222@bhati-test:2] NoOp(SIP/2218-02c3, Hint for
Extension 2218 is ) in new stack
-- Executing [222@bhati-test:3] Set(SIP/2218-02c3,
CALLERID(name)=From Google Talk) in new stack
-- Executing [222@bhati-test:4] Wait(SIP/2218-02c3, 10) in new
stack
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:InUse Watchers 0
1 hint matching extension 2218
-- Executing [222@bhati-test:5] Dial(SIP/2218-02c3,
SIP/my_sip_phones) in new stack
== Using SIP RTP CoS mark 5
[Dec 6 17:39:38] WARNING[6689]: chan_sip.c:5331 create_addr: No such host:
my_sip_phones
[Dec 6 17:39:38] WARNING[6689]: app_dial.c:1747 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/2218-02c3' status is 'CHANUNAVAIL'
-- Executing [h@bhati-test:1] NoOp(SIP/2218-02c3, hangup the call
now) in new stack
haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:IdleWatchers 0
1 hint matching extension 2218
Is this the right way to use HINT of asterisk ?
On Tue, Dec 6, 2011 at 3:38 PM, virendra bhati virbh...@gmail.com wrote:
Hi All,
I read about the Hint in asterisk. I want to implements into my server for
testing purpose. How to use it ? please help me...
--
Thanks and regards
Virendra Bhati
+91-8885268942 tel:%2B91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942 tel:%2B91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942 tel:%2B91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942 tel:%2B91-8885268942
Software Engineer
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Populate CDR issues
IMO you are trying to circumvent basic Asterisk functionality. It's your
CDR so you can do what you want with it - I think the answer to this is to
populate another DB with the live call data, then update the CDR from that
after the call has ended (perhaps a daemon).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Harel Cohen
Sent: Tuesday, December 06, 2011 3:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Populate CDR issues
Hello Everyone,
I didn't get a reply to my problem below so I'm posting again just in case
someone who might be able to help missed my previous post.
Thank You.
*
Hello list,
I'm trying to populate my CDR logs with values which are available after the
call has started (e.g. signalling IP of remote user, media IP, codec etc.).
While CHANNEL function give me all I need for the incoming leg (leg A), I
can't get the relevant values for the outgoing channel. I've tried using the
option 'U' with my dial command (execute subroutine for called channel after
called channel answered but before the call is bridged). While this throws
the correct information to the console it does not populate the CDRs
accordingly.
Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive
ODBC and the table therein contains the relevant fields.
This is the console with 'very-verbose' output for the 'Dial' application
where office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP
192.168.20.226. My comments added prefixed by ** and on separate line:
** channel here is source channel: SIP/office_Admin2-0015
[Dec 1 12:14:31] -- Executing [316@InternalDP:5]
Dial(SIP/office_Admin2-0015,
SIP/office_ServerRoom,,FgU(jump2SetVar)) in new stack
[Dec 1 12:14:31] == Using UDPTL CoS mark 5
[Dec 1 12:14:31] == Using SIP RTP CoS mark 5
[Dec 1 12:14:31] -- Called SIP/office_ServerRoom
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:33] -- SIP/office_ServerRoom-0016 answered
SIP/office_Admin2-0015
** from here the channel is the destination channel:
SIP/office_ServerRoom-0016
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:1]
Gosub(SIP/office_ServerRoom-0016, SetVar,postdial,1) in new stack
** This is how I obtain channel information:
** exten =
postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peer
name)},port)}) ; resulting format: a.b.c.d:port
** same = n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same = n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec 1 12:14:33] -- Executing [postdial@SetVar:1]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutsigip)=192.168.20.226:5065) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:2]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutmediaip)=192.168.20.226:23008) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:3]
Set(SIP/office_ServerRoom-0016, CDR(chanoutcodec)=g729) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:4]
Goto(SIP/office_ServerRoom-0016, endsub,1) in new stack
[Dec 1 12:14:33] -- Goto (SetVar,endsub,1)
[Dec 1 12:14:33] -- Executing [endsub@SetVar:1]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:2]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1]
NoOp(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Auto fallthrough, channel
'SIP/office_ServerRoom-0016' status is 'UNKNOWN'
[Dec 1 12:14:33] -- Remotely bridging SIP/office_Admin2-0015 and
SIP/office_ServerRoom-0016
When call is terminated the relevant fields in the database for
CDR(chanoutsigip), CDR(chanoutmediaip) and CDR(chanoutcodec) are populated
with their default values (typically blank or '-') and NOT with the
values above.
Am I doing something wrong or is there a different way to populate CDR's
with info from called channel (leg B)?
Thank you for your replies.
Harel
--
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Re: [asterisk-users] Simple Generic IVR to get us up an running Quick
IVR = Idiot Verify and Recognize? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Tuesday, December 06, 2011 4:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Simple Generic IVR to get us up an running Quick On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote: Hello Everyone, Are there any descent generic IVR recordings, that we can use to quickly get our PBX up and running? It will obviously not include the company name. It's easy enough to make your own recordings. Word of caution though. It might be advisable to ask somebody outside the company to record the phrases, Wonder why? At home i did it my self, and i still hear people stating that they have been talking at me, totaly unaware that it was just the voicemail anouncements. Peope just hear a voice, but seldom listen. And not just 90-old aunts, But people from helpdesks and even CEO's. Sometimes i wonder, if i should ask/test the callers I.Q. , And adapt the IVR's accordingly ;=) hw -- _ -- 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] Simple Generic IVR to get us up an running Quick
That's too funny! What are some tricks to make it sound professional. What I mean is, what are some of the typcial things people do to the recording, to make it sound kind-of robotic? I have no idea how to explain it. Maybe those of you that have done ivr recordings for corporations could share what tools and tricks you use to get that professional look? Thanks in Advance, Nick. On Tue, Dec 6, 2011 at 5:07 AM, Hans Witvliet aster...@a-domani.nl wrote: On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote: Hello Everyone, Are there any descent generic IVR recordings, that we can use to quickly get our PBX up and running? It will obviously not include the company name. It's easy enough to make your own recordings. Word of caution though. It might be advisable to ask somebody outside the company to record the phrases, Wonder why? At home i did it my self, and i still hear people stating that they have been talking at me, totaly unaware that it was just the voicemail anouncements. Peope just hear a voice, but seldom listen. And not just 90-old aunts, But people from helpdesks and even CEO's. Sometimes i wonder, if i should ask/test the callers I.Q. , And adapt the IVR's accordingly ;=) hw -- _ -- 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] Simple Generic IVR to get us up an running Quick
There are some Allison Smith Speaks blogs out there with good IVR hints. Some hints from my experience 1. The recommended volume adjustment for asterisk is -3 DB (that's -3 if you look at the wav in Audaciity). This will vary depending on your flavor of Asterisk and your input (SIP/DAHDI/etc). 2. Use a metronome - regular speech has a large variance in tempo. If you say leave your message after the tone normally you are more likely to get Nick's 90 year old aunt than leave-your-message-after-the-tone. 3. If you aren't going to buy a high quality microphone and software, you're just as well off recording using the normal record function. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Tuesday, December 06, 2011 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Simple Generic IVR to get us up an running Quick That's too funny! What are some tricks to make it sound professional. What I mean is, what are some of the typcial things people do to the recording, to make it sound kind-of robotic? I have no idea how to explain it. Maybe those of you that have done ivr recordings for corporations could share what tools and tricks you use to get that professional look? Thanks in Advance, Nick. On Tue, Dec 6, 2011 at 5:07 AM, Hans Witvliet aster...@a-domani.nl wrote: On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote: Hello Everyone, Are there any descent generic IVR recordings, that we can use to quickly get our PBX up and running? It will obviously not include the company name. It's easy enough to make your own recordings. Word of caution though. It might be advisable to ask somebody outside the company to record the phrases, Wonder why? At home i did it my self, and i still hear people stating that they have been talking at me, totaly unaware that it was just the voicemail anouncements. Peope just hear a voice, but seldom listen. And not just 90-old aunts, But people from helpdesks and even CEO's. Sometimes i wonder, if i should ask/test the callers I.Q. , And adapt the IVR's accordingly ;=) hw -- _ -- 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 -- _ -- 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] Simple Generic IVR to get us up an running Quick
What I would like to know is. How could you have possibly known I have a 90 year old aunt?!?! Sorry for the Noise! Merry Christmas/Happy Holidays, Nick. On Tue, Dec 6, 2011 at 9:29 AM, Danny Nicholas da...@debsinc.com wrote: There are some Allison Smith Speaks blogs out there with good IVR hints. Some hints from my experience 1. The recommended volume adjustment for asterisk is -3 DB (that's -3 if you look at the wav in Audaciity). This will vary depending on your flavor of Asterisk and your input (SIP/DAHDI/etc). 2. Use a metronome - regular speech has a large variance in tempo. If you say leave your message after the tone normally you are more likely to get Nick's 90 year old aunt than leave-your-message-after-the-tone. 3. If you aren't going to buy a high quality microphone and software, you're just as well off recording using the normal record function. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Tuesday, December 06, 2011 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Simple Generic IVR to get us up an running Quick That's too funny! What are some tricks to make it sound professional. What I mean is, what are some of the typcial things people do to the recording, to make it sound kind-of robotic? I have no idea how to explain it. Maybe those of you that have done ivr recordings for corporations could share what tools and tricks you use to get that professional look? Thanks in Advance, Nick. On Tue, Dec 6, 2011 at 5:07 AM, Hans Witvliet aster...@a-domani.nl wrote: On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote: Hello Everyone, Are there any descent generic IVR recordings, that we can use to quickly get our PBX up and running? It will obviously not include the company name. It's easy enough to make your own recordings. Word of caution though. It might be advisable to ask somebody outside the company to record the phrases, Wonder why? At home i did it my self, and i still hear people stating that they have been talking at me, totaly unaware that it was just the voicemail anouncements. Peope just hear a voice, but seldom listen. And not just 90-old aunts, But people from helpdesks and even CEO's. Sometimes i wonder, if i should ask/test the callers I.Q. , And adapt the IVR's accordingly ;=) hw -- _ -- 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 -- _ -- 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] A new hack?
On Tue, Dec 6, 2011 at 5:19 AM, Hans Witvliet aster...@a-domani.nl wrote: On Mon, 2011-12-05 at 18:51 -0800, Steve Edwards wrote: snip Your security needs depends on your environment. At this point in time, all of the hosts I manage for my clients exist in very limited environments and have very small attack surfaces. They are racked in secure data centers. They only accept SIP from clients with static IP addresses that we have an existing business relationship with. They only accept SSH connections from me. They only accept HTTP connections from me and my boss. That's about it. I don't see where F2B adds much value for me. *) Lots of admins think they can't limit access to servers because they have 'mobile' users. Your users probably don't need to access your servers from every single place on the Internet. If your users don't come from China, North Korea, Iran, etc, you can block entire regions with a few rules and eliminate 80% of probes and attacks from reaching your servers in the first place. Apologies in advance if you happen to live in some of these regions -- feel free to `s/China, North Korea, Iran/United States, Canada, England/g` Perhaps an other suggestion. If they are true road warriors, i presume they are capable of setting up an vpn to the company. In that case, only allow registrations/calls through the secured tunnel. Then it's not any concern to asterisk. And if they can breach your tunnel, you have something else to worry about. Well, that means opening up VPN connections from everywhere. Thats why I suggested turning off the server completely. -- _ -- 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] google voice calling dial plan question.
On Sat, Dec 3, 2011 at 12:59 AM, white hat whitehat...@gmail.com wrote: When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I just need a little advice on how to write the dial plan. I still have much to learn about asterisk, and appreciate any advice. Geez, Maybe I am just brute forcing it, but, the following dialplan seems to work (at least, most of the time!): [gtalk_incoming] exten = s,1,Answer() exten = s,n,Wait(5) exten = s,n,SendDTMF(1) exten = s,n,Dial(SIP/Ciscofficephone,10) exten = s,n,Playback(vm-nobodyavail) exten = s,n,Playback(vm-pls-try-again) same = n,Hangup() HTH, dwa dai...@pervasivetelcom.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] Asterisk 1.4 - Help/Doc for Park() application [SOLVED]
2011/12/5 Olivier oza_4...@yahoo.fr Hi, Porting a dialplan from 1.6.1 to and old 1.4 install, it seems Park() application uses different arguments. The only doc I could get a hand on is (core show application Park) this one : [Synopsis] Park yourself [Description] Park():Used to park yourself (typically in combination with a supervised transfer to know the parking space). This application is always registered internally and does not need to be explicitly added into the dialplan, although you should include the 'parkedcalls' context (or the context specified in features.conf). If you set the PARKINGEXTEN variable to an extension in your parking context, park() will park the call on that extension, unless it already exists. In that case, execution will continue at next priority. More specifically, I'm getting this : -- Executing [9200@autopark:49] Park(SIP/9140-0991dd30, 1000*30|9200|local|s) in new stack == Parked SIP/9140-0991dd30 on 701@parkedcalls. Will timeout back to extension [autopark] s, 1 in 45 seconds Above that, silent option 's' is ignored (parking position is read to incoming channel). So it seems, my timeout, return context and feedback options are not correctly understood. Suggestions ? Cheers Hi, Replying to myself, I worked around this using ParkAndAnnounce app instead (of Park). Too bad I could find by myself what was missing in documentation. Regards -- _ -- 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] rasterisk not knowing config path?
Hello I have machine running a couple of instances of asterisk. Each instance create own control pipe (asterisk.ctl). How I can remotely connect into asterisk which own pipe I know? I know I can do it if path to pipe specified in asterisk.conf, but I have not any asterisk.conf accessible, only control pipe. -- _ -- 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] A new hack?
Well, that means opening up VPN connections from everywhere. Thats why I suggested turning off the server completely. hmmm - I thought that was the point of a vpn -- _ -- 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] rasterisk not knowing config path?
You don't state your Asterisk version, but this sounds like a task for chan_skinny perhaps? Or it might just be as simple as hitting an RTP range. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yaroslav Panych Sent: Tuesday, December 06, 2011 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] rasterisk not knowing config path? Hello I have machine running a couple of instances of asterisk. Each instance create own control pipe (asterisk.ctl). How I can remotely connect into asterisk which own pipe I know? I know I can do it if path to pipe specified in asterisk.conf, but I have not any asterisk.conf accessible, only control pipe. -- _ -- 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] rasterisk not knowing config path?
2011/12/6 Danny Nicholas da...@debsinc.com: You don't state your Asterisk version, but this sounds like a task for chan_skinny perhaps? Or it might just be as simple as hitting an RTP range. Asterisk =1.8 No, I want manually connect to asterisk via ssh console. I.e. like: ssh user@ast-host /sbin/rasterisk -switch_to_point_asterisk.ctl path_to_asterisk.ctl I know path_to_asterisk.ctl but did not found any switch_to_point_asterisk.ctl in manuals. -- _ -- 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] Proper sip.conf and extensions.conf for Exchange 2010 U.M.
Hi All, I'm using Exchange as our voicemail system. Everything works fine until the 1 week mark when Exchange changes the port number used, then Asterisk 1.8 seg faults and I have no phones (unless I restart the U.M. service before the 1 week period is up). Since that is a hack, I'm hoping someone can post their working configs that accomodates the port change. The documentation I've seen is still a little unclear to me. I'm not using secured mode, so just using ports 5065/5067. TIA -- _ -- 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] google voice calling dial plan question.
If I understand correctly, turning off Call Screening in your Google
Voice configuration should directly connect incoming calls and eliminate
the need to press one.
JF
On 12/2/2011 11:59 PM, white hat wrote:
When a caller calls my google voice phone number, I must answer, wait
and press one to accept. Sometimes even that does not work.
I have tried a few different things to get asterisk to place the call
in an answered state and send the DTMF 1 with the Dial macro.
I found Malcom Davenports wiki page regarding Google calling which has
been very helpful in troubleshooting the issue.
https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google?focusedCommentId=18415969#comment-18415969
I'm sure that I'm close to getting things working properly.
Here's my config.
##jabber.conf##
[general]
debug=no
autoprune=no
autoregister=yes
[whitehat238]
type=client
serverhost=talk.google.com http://talk.google.com
username=whitehat...@gmail.com/Talk http://whitehat...@gmail.com/Talk
secret=password
port=5222
usetls=yes
usesasl=yes
status=Available
statusmessage=No Information Available
timeout=100
keepalive=yes
##gtalk.conf##
[general]
allowguest=yes
context=googlein
stunaddr=stun01.sipphone.com http://stun01.sipphone.com
[guest]
disallow=all
allow=ulaw
connection=whitehat238
context=googlein
##extensions_custom.conf##
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,1,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)})
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,n,GotoIf($[${CALLERID(name):0:2} !=
+1]?notrim)
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,n,Set(CALLERID(name)=${CALLERID(name):2})
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,n(notrim),Set(CALLERID(number)=${CALLERID(name)})
exten = whitehat...@gmail.com mailto:whitehat...@gmail.com,n,Answer
exten = whitehat...@gmail.com mailto:whitehat...@gmail.com,n,Wait(1)
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,n,SendDTMF(1)
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,n,Goto(from-trunk,5025551212,1)
[gvoice-whitehat238]
exten = _X.,1,Dial(Gtalk/whitehat238/+${EXTEN}@voice.google.com
mailto:exten...@voice.google.com)
exten = _X.,n,Noop(GVoice Call to ${EXTEN} failed)
exten = h,1,Macro(hangupcall,)
I have a working inbound route which rings an internal extension
(7008) when calling the GV number. I can also make outbound calls to
any number using the GV trunk.
I found this page (Link to Michigan telephone blog) which helped me
get everything setup initially and included a shell script that made
it easy to generate the configuration.
http://michigantelephone.wordpress.com/2011/01/20/a-bash-script-to-assist-asterisk-1-8freepbx-2-8-users-in-adding-new-google-voice-accounts/
The author explains the config in more detail and why he choose to
write it the way he did.
I have tried using the alternative method of sending the DTMF 1 tone
by changing the last block as follows:
[gvoice-whitehat238]
exten = _X.,1,Dial(Gtalk/whitehat238/+${EXTEN}@voice.google.com
mailto:exten...@voice.google.com,D(:1))
exten = _X.,n,Noop(GVoice Call to ${EXTEN} failed)
exten = h,1,Macro(hangupcall,)|
However, that did not work.
I just need a little advice on how to write the dial plan. I still
have much to learn about asterisk, and appreciate any advice.
Thanks,
|
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Re: [asterisk-users] rasterisk not knowing config path?
On 12/6/11 9:18 AM, Yaroslav Panych wrote: Asterisk=1.8 No, I want manually connect to asterisk via ssh console. I.e. like: ssh user@ast-host /sbin/rasterisk -switch_to_point_asterisk.ctl path_to_asterisk.ctl I knowpath_to_asterisk.ctl but did not found any switch_to_point_asterisk.ctl in manuals. /usr/sbin/asterisk -r -s path_to_asterisk.ctl and you'll need both rw permission to that socket. (and minimum of x permission for the directory tree it resides in) -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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] google voice calling dial plan question.
Hey Josh, I've messed with the google voice account settings extensively. As of now, in Google voice account settings I have. Voice tab: forward calls to Google chat checked. Nothing else is checked. Calls tab: call screening is off. On incoming call, display callers number. On Caller ID outing. Don't change anything is selected. Do not disturb is disabled. Nothing else is checked (enabled) The behavior is that the call comes in, and asterisk rings extension 7008, but I never here the prompt by Google to press one to accept the call. It either isn't played, isn't recognized, by Google when asterisk sends the DTMF 1, or it's played before I answer the extension and I don't hear it because the audio streams were not connected when it was played. If I answer extension 7008, and then press 1 (full one second press of the button) then most of the time it will connect the call. Sometimes I have to press 1 two or three times before it will connect, and rarely, it won't connect at all, even with the key presses. As part of the troubleshooting I have removed all other Google voice accounts in extensions_additional.conf, and left only the whitehat238 gvoice connection. Now the prompt is never played but the key press is still required as if it were. On Tue, Dec 6, 2011 at 10:36 AM, Dave Aibel dai...@pervasivetelecom.comwrote: On Sat, Dec 3, 2011 at 12:59 AM, white hat whitehat...@gmail.com wrote: When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I just need a little advice on how to write the dial plan. I still have much to learn about asterisk, and appreciate any advice. Geez, Maybe I am just brute forcing it, but, the following dialplan seems to work (at least, most of the time!): [gtalk_incoming] exten = s,1,Answer() exten = s,n,Wait(5) exten = s,n,SendDTMF(1) exten = s,n,Dial(SIP/Ciscofficephone,10) exten = s,n,Playback(vm-nobodyavail) exten = s,n,Playback(vm-pls-try-again) same = n,Hangup() HTH, dwa dai...@pervasivetelcom.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] google voice calling dial plan question.
dwa As part of the troubleshooting I updated all of the asterisk packages from the repo with yum. I'm using freepbx distro (centos based) with asterisk 1.8 There were several newer asterisk 1.8 packages available. I'm not using any custom modules in freepbx. After the updates, I restarted asterisk with core restart now but this hasn't helped. I'm sure it's a dial plan configuration issue. Would you be willing to post sanitized versions of your jabber.conf, gtalk.conf and details regarding the context you're using and how your inbound route is configured in your dial plan? Are you using STUN? Is Asterisk behind a NAT device or on a public IP? Thanks On Tue, Dec 6, 2011 at 10:36 AM, Dave Aibel dai...@pervasivetelecom.comwrote: On Sat, Dec 3, 2011 at 12:59 AM, white hat whitehat...@gmail.com wrote: When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I just need a little advice on how to write the dial plan. I still have much to learn about asterisk, and appreciate any advice. Geez, Maybe I am just brute forcing it, but, the following dialplan seems to work (at least, most of the time!): [gtalk_incoming] exten = s,1,Answer() exten = s,n,Wait(5) exten = s,n,SendDTMF(1) exten = s,n,Dial(SIP/Ciscofficephone,10) exten = s,n,Playback(vm-nobodyavail) exten = s,n,Playback(vm-pls-try-again) same = n,Hangup() HTH, dwa dai...@pervasivetelcom.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] google voice calling dial plan question.
You could also try putting a Progress() statement between Answer and Wait. I know there is a latency issue with DAHDI calls; 5 seconds may or may not be enough for googlevoice. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of white hat Sent: Tuesday, December 06, 2011 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] google voice calling dial plan question. dwa As part of the troubleshooting I updated all of the asterisk packages from the repo with yum. I'm using freepbx distro (centos based) with asterisk 1.8 There were several newer asterisk 1.8 packages available. I'm not using any custom modules in freepbx. After the updates, I restarted asterisk with core restart now but this hasn't helped. I'm sure it's a dial plan configuration issue. Would you be willing to post sanitized versions of your jabber.conf, gtalk.conf and details regarding the context you're using and how your inbound route is configured in your dial plan? Are you using STUN? Is Asterisk behind a NAT device or on a public IP? Thanks On Tue, Dec 6, 2011 at 10:36 AM, Dave Aibel dai...@pervasivetelecom.com wrote: On Sat, Dec 3, 2011 at 12:59 AM, white hat whitehat...@gmail.com wrote: When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I just need a little advice on how to write the dial plan. I still have much to learn about asterisk, and appreciate any advice. Geez, Maybe I am just brute forcing it, but, the following dialplan seems to work (at least, most of the time!): [gtalk_incoming] exten = s,1,Answer() exten = s,n,Wait(5) exten = s,n,SendDTMF(1) exten = s,n,Dial(SIP/Ciscofficephone,10) exten = s,n,Playback(vm-nobodyavail) exten = s,n,Playback(vm-pls-try-again) same = n,Hangup() HTH, dwa dai...@pervasivetelcom.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
[asterisk-users] IRC Client
Hello all, I am new to the Asterisk IRC users group. I was wondering if it would be possible to use an IRC client when reading through the posts. If so, can someone recommend one, and how I should go about configuring the client. Thank you for your time and assistance. Peter -- _ -- 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] Hint'ing with XMPP?
Yes, we are using it. Most of the docs on the Internet are for 1.4. However, we now have it working with 1.8 (after some work). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Tuesday, December 06, 2011 5:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hint'ing with XMPP? 2011/12/5 Jamie A. Stapleton jstaple...@computer-business.commailto:jstaple...@computer-business.com I have not ever done what you are talking about. However, I can tell you that our Openfire XMPP server has similar functionality because of their Asterisk-IM Plugin. Are you currently using it ? With which asterisk version ? From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jay R. Worthington Sent: Saturday, December 03, 2011 8:11 AM To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Hint'ing with XMPP? Hiya, can i use an XMPP Client to see the presence of a hint? I have configured asterisk in component-mode, seem's to work, but all users (xmpp:1...@asterisk.dohmain.commailto:xmpp%3a...@asterisk.dohmain.com are online, even if 123 isn't a configured hint). Any good howto's out there, all the stuff on voip-info.orghttp://voip-info.org is completely outdated, i'm using asterisk 10... Regards Jay -- _ -- 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
