Re: [asterisk-users] dahdi and digium debian package
On Wednesday 22 February 2012, Paul Belanger wrote: On 12-02-22 01:36 PM, A J Stiles wrote: My best suggestion? Uninstall and purge any Debian packages you installed (backup needed config files first .) and just install Dahdi and Asterisk from the Source Code. I don't know how to reply to this. I can only assume you only manage a single box. What if OP is running more then 1 asterisk box, manually compiling asterisk and installing it each time would not be the best solution. Then you build your own .deb package, and install that on all the boxes you manage (assuming the same architecture). -- AJS Answers come *after* questions. -- _ -- 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] Phone Inventory
Does anyone know if Snom phones can be set to do this as well? Thanks Ish On Thu, 2012-02-23 at 09:31 -0500, Eric Wieling wrote: Polycom phones can be set to include their MAC in the User Agent string. Useragent: PolycomSoundPointIP-SPIP_550-UA/3.3.4.0085_0004f2233929 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, February 23, 2012 2:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phone Inventory Usually phones transmit all these informations as User Agent, so you can just write in asterisk cli: sip show peer peername and you'll get something like : Useragent: Yealink SIP-T28P 2.60.0.40 Reg. Contact : sip:peername@192.168.25.102:5062 Qualify Freq : 6 ms Leandro 2012/2/23 Muro, Sam resea...@businesstz.com: Hi there I have just took a support of a customer with hundreds of IP phones, mostly Polycom with mixed models. Is there a way to query asterisk or any other command to retrieve the inventory of all connected phones. i.e. Phone Type and Phone Model, say Polycom SPIP331 or so Thanks Sam -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Phone Inventory
On 02/23/2012 08:49 AM, Danny Nicholas wrote:
Here is a snippet that somebody smarter than I am can improve upon
for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx sip
show peer $a;done|grep Useragent
for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx sip
show peer $a;done|grep Contact
Thanks for the inspiration!!
Here is my version, done with a single loop and gets Useragent and
Contact together with a visual separation between peers.
asterisk -rx sip show peers|
cut -f1 -d/ | grep -P '\d\d\d\d' |
grep -vP '(UNKNOWN|Unmonitored)' |
while read PEER
do
asterisk -rx sip show peer ${PEER} |
grep -P (Useragent|Contact)
echo
done
I hope others find it useful.
Dale
PS. I by no means claim to be smarter than thou. I just happen to
really like grep and the -P option ;-)
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Re: [asterisk-users] Phone Inventory
Thank you all
You are a life saver
Sam
Dale Noll wrote:
On 02/23/2012 08:49 AM, Danny Nicholas wrote:
Here is a snippet that somebody smarter than I am can improve upon
for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx
sip
show peer $a;done|grep Useragent
for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx
sip
show peer $a;done|grep Contact
Thanks for the inspiration!!
Here is my version, done with a single loop and gets Useragent and
Contact together with a visual separation between peers.
asterisk -rx sip show peers|
cut -f1 -d/ | grep -P '\d\d\d\d' |
grep -vP '(UNKNOWN|Unmonitored)' |
while read PEER
do
asterisk -rx sip show peer ${PEER} |
grep -P (Useragent|Contact)
echo
done
I hope others find it useful.
Dale
PS. I by no means claim to be smarter than thou. I just happen to
really like grep and the -P option ;-)
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Re: [asterisk-users] Phone Inventory
+1 Dale - p.s. the grep -P '\d\d\d\d' killed the output on my 1.4 box.
P.P.S if you change grep -P (Useragent|Contact) to grep -P
(Username|Contact|Username) it produces a nice 4 line report like this:
Def. Username: Danny Nicholas
Useragent: PolycomSoundPointIP-SPIP_501-UA/3.1.2.0392
Reg. Contact : sip:104@192.168.23.114
=
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dale Noll
Sent: Thursday, February 23, 2012 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Phone Inventory
On 02/23/2012 08:49 AM, Danny Nicholas wrote:
Here is a snippet that somebody smarter than I am can improve upon for
a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx sip
show peer $a;done|grep Useragent for a in `asterisk -rx sip show
peers|cut -f1 -d/` ;do asterisk -rx sip show peer $a;done|grep
Contact
Thanks for the inspiration!!
Here is my version, done with a single loop and gets Useragent and Contact
together with a visual separation between peers.
asterisk -rx sip show peers|
cut -f1 -d/ | grep -P '\d\d\d\d' |
grep -vP '(UNKNOWN|Unmonitored)' |
while read PEER
do
asterisk -rx sip show peer ${PEER} |
grep -P (Useragent|Contact)
echo
done
I hope others find it useful.
Dale
PS. I by no means claim to be smarter than thou. I just happen to
really like grep and the -P option ;-)
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Re: [asterisk-users] Block Collect Calls on ISDN trunk
I tried to leave these fields empty. It did not work. Making the trunkgroups section empty was to eliminate a different configuration problem you have. It is not related to the reverse charging issue. Not appear IE74 in the debug. The possibility exists that the carrier does not be sending? Yes. It does look like the carrier is not sending the Reverse Charging Indication ie. Richard Att, Rafael Saraiva 2012/2/17 Richard Mudgett rmudg...@digium.com I had not noticed that you switched to direct email earlier. - Original Message - Switchtype: euroisdn This is my chan_dadhi.conf [trunkgroups] trunkgroup=1,16 spanmap=1,1 The trunkgroups section is only needed if you are using NFAS and its presence may cause issues for switches that do not support it. NFAS is not supported by euroisdn or qsig. After looking again at your traces, the STATUS messages are complaining about the channel id indicated by Asterisk. The trunk groups setup you have here is causing it. Just make this an empty section. [channels] language=pt_BR hidecallerid=no usecallerid=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no callerid=asreceived switchtype=euroisdn ;qsig signalling=pri_cpe context=PABX facilityenable=yes hidecalleridname=no ;aoc_enable=s,d,e imediate=yes ;busydetect=yes overlapdial=yes inbanddisconnect=yes priindication=inband ;outofband nsf=none ;qsigchannelmapping=logical pridialplan=unknown prilocaldialplan=unknown ;internationalprefix=00 ;nationalprefix=55 ;localprefix=5551 ;privateprefix=55513205 ;unknownprefix= service_message_support=yes mohinterpret=default discardremoteholdretrieval=yes group=1 channel=1-15,17-31 And system.conf: loadzone=br defaultzone=br #span=1,1,0,cas,hdb3 #cas=1-15,17-31:1101 span=1,0,0,ccs,hdb3 dchan=16 bchan=1-15,17-31 Att, Rafael Saraiva 2012/2/17 Richard Mudgett rmudg...@digium.com I could not send attachments to the list. Att, Rafael Saraiva You could have put the traces inline like you did the console capture. However, the attached files were oddly garbled in that lines were duplicated. It looks like your network switch does not like the PROCEEDING or ALERTING messages. The SETUP message did not have any differences between a normal call and a collect call that I could see. So there does not appear to be a way to tell the calls apart. What is the configured switch type? Richard -- _ -- 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] Phone Inventory
My short google-fu session says no - It's not universal for Polycom either (the phone has to support microbrowser). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Thursday, February 23, 2012 9:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phone Inventory Does anyone know if Snom phones can be set to do this as well? Thanks Ish On Thu, 2012-02-23 at 09:31 -0500, Eric Wieling wrote: Polycom phones can be set to include their MAC in the User Agent string. Useragent: PolycomSoundPointIP-SPIP_550-UA/3.3.4.0085_0004f2233929 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, February 23, 2012 2:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phone Inventory Usually phones transmit all these informations as User Agent, so you can just write in asterisk cli: sip show peer peername and you'll get something like : Useragent: Yealink SIP-T28P 2.60.0.40 Reg. Contact : sip:peername@192.168.25.102:5062 Qualify Freq : 6 ms Leandro 2012/2/23 Muro, Sam resea...@businesstz.com: Hi there I have just took a support of a customer with hundreds of IP phones, mostly Polycom with mixed models. Is there a way to query asterisk or any other command to retrieve the inventory of all connected phones. i.e. Phone Type and Phone Model, say Polycom SPIP331 or so Thanks Sam -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Is Asterisk 10 available in Digium Repository? I doesn't show up
Hi, I have followed instruction on https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisites to add Digium Asterisk repositories but doing a, yum search asterisk only shows me Asterisk 1.4, 1.6, and 1.8. There is no Asterisk 10 and yum install asterisk10 fails. Am I missing something? or Asterisk 10 is just no available in binary? Thanks, -- _ -- 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] Is Asterisk 10 available in Digium Repository? I doesn't show up
On 02/23/2012 10:09 AM, Ast Coder wrote: Hi, I have followed instruction on https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisites to add Digium Asterisk repositories but doing a, yum search asterisk only shows me Asterisk 1.4, 1.6, and 1.8. There is no Asterisk 10 and yum install asterisk10 fails. Am I missing something? or Asterisk 10 is just no available in binary? Thanks, There are now repositories for each major version of Asterisk, which have to be explicitly enabled to use them. `yum update` to get to the latest of everything, then do `yum update --enablerepo=asterisk-10`. Asterisk 10 will be installed, and that repository will be enabled permanently. I'll add that information to the wiki shortly. -- _ -- 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] Streaming musiconhold via mpg123
On 02/21/2012 11:30 PM, Jason Parker wrote: On 02/21/2012 05:34 PM, Stephen Brown wrote: application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000 /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3 Probably unrelated to your issue, but you're going to want to quote that filename. And while I am not a lawyer, nor do I play one on TV, you should be aware that using copyrighted music as music-on-hold can constitute 'broadcast' of that music, and might result in you being obligated to pay royalties to the copyright holders and their license managers. You need to investigate the situation in the place you plan to do this. I'll also state that while I very much enjoy that song, I do not want to *ever* hear it in 8kHz mono :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_rpt and chan_usbradio removal from trunk
Just to inform the list - App_rpt and chan_usbradio are still regularly used and maintained, but now live in a repository at ohnosec.org along with the forked-off builds of Asterisk 1.4 and Zaptel that are required to have them work properly. I'm told there is some fundamental incompatibility between canonical Zaptel/DAHDI and the radio application that can't be effectively worked around, or would take more effort than it would be worth to fix and keep up with DAHDI changes. This was the motivation for forking Asterisk and maintaining a separate codebase. Although I can't speak authoritatively for the app_rpt community, I'll say that I haven't seen too much concern over being stuck with 1.4. Those of us who really like the idea of integrating our app_rpt radio systems with more conventional Asterisk use cases find it much easier (and often more desirable anyway, from a system viewpoint) to just set up a second box with canonical 1.8 or 10 and trunk the two together. Josh Freeman On 02/23/2012 08:57 AM, Paul Belanger wrote: Good morning, There is a new patch up on reviewboard[1] right now for the removal of app_rpt and chan_usbradio from Asterisk trunk. As it stands right now these two modules do not appear to be maintained in this repository and have out-of-date code. Russellb's patch will see these to modules removed from asterisk trunk (asterisk 11). If a large part of the community wishes to help maintain this code, please speak up. As it stands right now, we'll likely wait a week or two remove committing the patch. [1] - https://reviewboard.asterisk.org/r/1764/ -- _ -- 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] Starting asterisk: Cannot find specified TTY
Hello, when trying to start Asterisk on a new server, I got the following problem : [root@sip asterisk]# /etc/init.d/asterisk status asterisk is stopped [root@sip asterisk]# /etc/init.d/asterisk start Starting asterisk: Cannot find specified TTY (9) [FAILED] I can start asterisk with /usr/sbin/asterisk -c but when I quit the CLI the asterisk proces is stopped also... What's wrong ?! I do not have this on another server. Kind regards, Jonas. -- _ -- 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] Rejecting transfers to in-use parking spaces
I'm trying to emulate the functionality of our existing phone system, which is
somewhat different than what Asterisk provides with a trivial parking
configuration. I'd like each user to have three park buttons, park 1, park 2,
park 3. The snom 870s I'm using have a Park+Orbit button, which best I can
determine, is a shortcut to transfer someone to an extension. So, I defined
some extensions:
exten = _*70[123],1,NoOp(parking in ${EXTEN:1})
same = n,Set(PARKINGEXTEN=${EXTEN:1})
same =
n,GotoIf(${DEVICE_STATE(park:${PARKINGEXTEN}@parkedcalls)}=INUSE?busy)
same = n,Park()
same = n(busy),Busy()
As you can see, I'm calling Busy() if someone is already parked in the space,
but this doesn't do what I'd like. What I'm hoping to accomplish is have
Asterisk respond to the Sip REFER to *701 with a 404 or similar response; if
Asterisk can do this, then the Snom will say transfer failed!. As it is, the
transfer is successful, and the caller hears a busy tone. Is there an
application that has the effect of Pretend this extension doesn't exist, or
can I somehow get the caller back to the person that tried to park them in this
space that's in use?
Also, if anyone has specific experience with the Snoms, I'd like to improve
this further. The Park+Orbit buttons seem to transfer the caller to an
extension, and I can use a BLF button to monitor the spaces and unpark calls.
It would be better if I could do this with just one button which parks the
caller if currently on a call, or which unparks a call if I'm not on a call.
Anyone have some idea how to accomplish this with the Snom 870?
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Re: [asterisk-users] Asterisk NOT in the media path
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls to one of the
2 available Asterisk-servers (B1 or B2) behind it.
So I want the first Asterisk-server A to accept the call, and based upon
some checks in the dialplan send the call through to one of the other
Asterisk-servers (B1 or B2) which further handle the call.
The first Asterisk-server A then needs to pull itself from the
media-path. There's no further need for this Asterisk to stay within the
audio-path.
1. Is this possible ?
2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes in the peer
definition of Asterisk B1 and Asterisk B2 ?
So I have :
Provider Asterisk A1 Asterisk B1 Asterisk B2
I want the audio to go directly from Provider to server B1 when the call
has been set up.
As long as there are no NATs involved, yes, this should work. You will
also need 'canreinvite' ('directmedia' in Asterisk 1.8 and later) in
the peer definition for the provider.
Hello again,
this is currently not really working.
I see on the Asterisk CLI that the call streams through my Asterisk A1
(which should stay out of the media path) :
[Feb 23 22:24:47] -- Called Mast/980419
[Feb 23 22:24:47] -- SIP/Mast-000e answered SIP/VOXBONEin-000d
[Feb 23 22:24:47] -- Native bridging SIP/VOXBONEin-000d and
SIP/Mast-000e
*CLI
*CLI core show channels
Channel Location State Application(Data)
SIP/Mast-00 (None) Up AppDial((Outgoing Line))
SIP/VOXBONEin-00 980419@VOXBONEin Up Dial(SIP/Mast/980419)
2 active channels
1 active call
Peer VoxBone and peer Mast should re-invite and leave this Asterisk out
of the media path on call answer.
These are my SIP peer definitions :
[VOXBONEin]
type=peer
host=XX.XX.XX.XX
context=VOXBONEin
disallow=all
allow=alaw
allow=gsm
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
[Mast]
type=peer
host=XX.XX.XX.XX
defaultuser=Mast
secret=guessme
disallow=all
allow=alaw
allow=gsm
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
Am I missing a setting ? Using Asterisk 1.6.2.22
Regards,
Jonas.
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Re: [asterisk-users] Starting asterisk: Cannot find specified TTY
On 02/23/2012 08:36 PM, Jonas Kellens wrote: Hello, when trying to start Asterisk on a new server, I got the following problem : [root@sip asterisk]# /etc/init.d/asterisk status asterisk is stopped [root@sip asterisk]# /etc/init.d/asterisk start Starting asterisk: Cannot find specified TTY (9) [FAILED] I can start asterisk with /usr/sbin/asterisk -c but when I quit the CLI the asterisk proces is stopped also... What's wrong ?! I do not have this on another server. Excuse me, I found the answer here : http://wiki.openvz.org/Asterisk_from_source -- _ -- 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] Transmit NOA (sss) to Dialogic IMG via SIP / Transmisión de NOA hacia Dialogic IMG por SIP
Spanish/Español: Hola a Tod@s, Estoy en la implementación de una solución a medida en la cual requerimos conectarnos vía SIP a un IMG de DIalogic y salir hacia otras centrales (MSC) vía SS7, la conexión entre Asterisk y el IMG es SIP, pero como requerimiento necesito enviar un NOA (Nature of Address) con valor 8 por requerimientos normativos, hasta ahora no he podido encontrar como puedo enviar dicho valor sobre SIP, asumo que existirá algún parámetro en las cabeceras del Invite, pero no me ha funcionado nada, ya que el NOA siempre lo recibe el IMG y las demás centrales con valor 3. Cualquier ayuda sobre como forzar el NOA a 8 sobre la troncal SIP se las agradezco mucho. El escenario: Asterisk SIP IMG Dialogic SS7 MSC Un Saludo! English/Ingles: Hi Everyone, I'm implementing a custom solution in which we require to connect via SIP a Dialogic IMG with Asterisk and make calls to other MSC via SS7, to connect Asterisk and IMG we have a SIP trunk, but now i need to send the NOA (Nature of Address ) with value 8 for regulatory requirements, so far I couldn't find how can i send the NOA value on SIP, I assume that there will be some parameters on the Invite headers, but nothing had worked so far, because the NOA always get to the IMG in 3. Thanks for any help on how to force the NOA value to 8 on the SIP trunk. The Scenario: Asterisk SIP IMG Dialogic SS7 MSC Regards! -- Luis Alejandro Beltrán Castañeda. Gerente General. SetColombia SAS. Bogota - Colombia. Tel/Fax: (571) 4756296 Movil: (57) 300-2721370 www.setcolombia.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
[asterisk-users] Trunking betweeb two Asterisk System
Hi guys, I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 1.6 but I cannt make it work, can any body help me plz? Thank you -- _ -- 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] Rejecting transfers to in-use parking spaces
I'm trying to emulate the functionality of our existing phone system,
which is somewhat different than what Asterisk provides with a
trivial parking configuration. I'd like each user to have three park
buttons, park 1, park 2, park 3. The snom 870s I'm using have a
Park+Orbit button, which best I can determine, is a shortcut to
transfer someone to an extension. So, I defined some extensions:
exten = _*70[123],1,NoOp(parking in ${EXTEN:1})
same = n,Set(PARKINGEXTEN=${EXTEN:1})
same =
n,GotoIf(${DEVICE_STATE(park:${PARKINGEXTEN}@parkedcalls)}=INUSE?busy)
same = n,Park()
same = n(busy),Busy()
As you can see, I'm calling Busy() if someone is already parked in
the space, but this doesn't do what I'd like. What I'm hoping to
accomplish is have Asterisk respond to the Sip REFER to *701 with a
404 or similar response; if Asterisk can do this, then the Snom will
say transfer failed!. As it is, the transfer is successful, and
the caller hears a busy tone. Is there an application that has the
effect of Pretend this extension doesn't exist, or can I somehow
get the caller back to the person that tried to park them in this
space that's in use?
The dialplan device state check above is not always going to work
because another call could park in that space between the check
and the actual park. The device state check in this case is
also unnecessary because Park will continue executing dialplan if
the park fails.
You could try three parkinglots with one parking space each. Each
phone park button would transfer the call to a different parkinglot.
Then all you need to do is include the parkinglot context(s) into your
dialplan context to have access to the generated parking extensions.
Please note that for Asterisk to detect an extension as a parking
extension, the first priority of the extension must be the park
application. If the park application is not the first priority of
the extension, then the transfer is treated as a normal transfer.
Richard
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Re: [asterisk-users] Rejecting transfers to in-use parking spaces
On Feb 23, 2012, at 16:32 , Richard Mudgett wrote:
exten = _*70[123],1,NoOp(parking in ${EXTEN:1})
same = n,Set(PARKINGEXTEN=${EXTEN:1})
same =
n,GotoIf(${DEVICE_STATE(park:${PARKINGEXTEN}@parkedcalls)}=INUSE?busy)
same = n,Park()
same = n(busy),Busy()
What I'm hoping to
accomplish is have Asterisk respond to the Sip REFER to *701 with a
404 or similar response; if Asterisk can do this, then the Snom will
say transfer failed!.
The dialplan device state check above is not always going to work
because another call could park in that space between the check
and the actual park. [...]
Please note that for Asterisk to detect an extension as a parking
extension, the first priority of the extension must be the park
application. If the park application is not the first priority of
the extension, then the transfer is treated as a normal transfer.
Hmm. Well, I could live with that race condition. Our call volume is low enough
that the odds of hitting it acceptably low.
I'm not entirely sure what it means for Asterisk to detect an extension as a
parking extension. Can you please explain how a parking transfer is different
than a normal transfer?
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Re: [asterisk-users] Hot desking and presence
One modification to my previous dialplan:
[hotdesk_outbound]
includes (via cascade) internal-calls
exten = .X,1,NoOp()
...snip do stuff to determine who's calling, set their extension number to
WHO...
same = n,Set(GROUP(activecalls)=${WHO})
same = n,Set(DEVICE_STATE(Custom:${WHO})=INUSE)
...snip make the call...
Needs to be:
[hotdesk_outbound]
includes (via cascade) internal-calls
exten = .X,1,NoOp()
...snip do stuff to determine who's calling, set their extension number to
WHO...
same = n,Set(GROUP(activecallers)=${WHO})
same = n,Set(DEVICE_STATE(Custom:${WHO})=INUSE)
...snip make the call...
And similarly, all checks or use of GROUP_COUNT(${WHO}@activecalls) should be
changed to GROUP_COUNT(${WHO}@activecallers). I missed the fact that setting
GROUP(activecalls)=${E} later on in my dialplan was overwriting the
GROUP(activecalls)=${WHO} for intra-office calls, and thus breaking my ability
to see that the person placing the intra-office call was on another line if a
second call rang their phone or what have you.
Noah Engelberth
MetaLINK Technologies
System Administration
The message does not contain any threats
AVG for MS Exchange Server (2012.0.1913 - 2114/4827)--
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Re: [asterisk-users] Rejecting transfers to in-use parking spaces
exten = _*70[123],1,NoOp(parking in ${EXTEN:1})
same = n,Set(PARKINGEXTEN=${EXTEN:1})
same =
n,GotoIf(${DEVICE_STATE(park:${PARKINGEXTEN}@parkedcalls)}=INUSE?busy)
same = n,Park()
same = n(busy),Busy()
What I'm hoping to
accomplish is have Asterisk respond to the Sip REFER to *701 with
a
404 or similar response; if Asterisk can do this, then the Snom
will
say transfer failed!.
The dialplan device state check above is not always going to work
because another call could park in that space between the check
and the actual park. [...]
Please note that for Asterisk to detect an extension as a parking
extension, the first priority of the extension must be the park
application. If the park application is not the first priority of
the extension, then the transfer is treated as a normal transfer.
Hmm. Well, I could live with that race condition. Our call volume is
low enough that the odds of hitting it acceptably low.
The next sentence also tried to point out that the check is not even
needed because if Park fails for whatever reason, it continues executing
dialplan. With that dialplan, it is mainly going to fail if the parking
space is already in use.
I'm not entirely sure what it means for Asterisk to detect an
extension as a parking extension. Can you please explain how a
parking transfer is different than a normal transfer?
If Asterisk detects the extension as a parking extension:
exten = 700,1,Park()
Asterisk can perform special processing dealing with parking the
call that may be needed for the channel driver. Also note that the
dialplan extension is *not* actually executed in this case.
For SIP, blind transferring to parking, the parking is done as part
of the transfer. If the park attempt fails, the transfer fails. A
normal blind transfer would complete the transfer and then execute
the dialplan extension on the transferred channel.
For DTMF transfers (features.conf), attended and blind transferring
to parking are identical. You will always hear the parking space
assigned. If the park attempt fails, the interrupted bridge will be
resumed.
Richard
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Re: [asterisk-users] Trunking betweeb two Asterisk System
On Thu, Feb 23, 2012 at 2:21 PM, Faraj Khasib fkha...@iconnecths.com wrote: Hi guys, I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 1.6 but I cannt make it work, can any body help me plz? Errors and other details might be helpful. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Trunking betweeb two Asterisk System
Hello are you using remotesecret on the trunk? regards - Original Message - From: Carlos Alvarez car...@televolve.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 23, 2012 6:08 PM Subject: Re: [asterisk-users] Trunking betweeb two Asterisk System On Thu, Feb 23, 2012 at 2:21 PM, Faraj Khasib fkha...@iconnecths.com wrote: Hi guys, I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 1.6 but I cannt make it work, can any body help me plz? Errors and other details might be helpful. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Rejecting transfers to in-use parking spaces
From: Phil Frost p...@macprofessionals.com
Sent: Thursday, February 23, 2012 4:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Rejecting transfers to in-use parking spaces
On Feb 23, 2012, at 16:32 , Richard Mudgett wrote:
exten = _*70[123],1,NoOp(parking in ${EXTEN:1})
same = n,Set(PARKINGEXTEN=${EXTEN:1})
same =
n,GotoIf(${DEVICE_STATE(park:${PARKINGEXTEN}@parkedcalls)}=INUSE?busy)
same = n,Park()
same = n(busy),Busy()
What I'm hoping to
accomplish is have Asterisk respond to the Sip REFER to *701 with a
404 or similar response; if Asterisk can do this, then the Snom will
say transfer failed!.
The dialplan device state check above is not always going to work
because another call could park in that space between the check
and the actual park. [...]
Please note that for Asterisk to detect an extension as a parking
extension, the first priority of the extension must be the park
application. If the park application is not the first priority of
the extension, then the transfer is treated as a normal transfer.
Hmm. Well, I could live with that race condition. Our call volume is low
enough that the odds of hitting it acceptably low.
I'm not entirely sure what it means for Asterisk to detect an extension as
a parking extension. Can you please explain how a parking transfer is
different than a normal transfer?
--
_
I was working on this today. I have it figured out but I don't have simple
dialplan code I can share as we are doing a lot of external db and script
calls to make ours work with our realtime stuff. We also are using the
Dynamic Parking stuff as well I pulled that out to simplify things.
[DoPark-Pickup-BlindPark]
exten = s,1,NoOp(Dynamic Parking Pickup)
exten = s,n,NoOp(Return Parked Call)
exten = s,n,GoToIf($[${LEN(${BLINDTRANSFER})} 0]?doParkAttempt,1)
exten = s,n,ParkedCall( PLACE DIALED EXTENSION VARIABLE HERE)
exten = doParkAttempt,1,NoOp(Attempt To Park)
Place logic to parse the ${BLINDTRANSFER} to get the return to extension
exten = doParkAttempt,n,Set(PARKINGLOT=Your Lot or var to handle the
lot)
exten = doParkAttempt,n,Set(PARKINGEXTEN=( PLACE DIALED EXTENSION VARIABLE
HERE))
exten = doParkAttempt,n,Park(time out, return_context, return_ext, return
priority, s)
If park fails return the call back to your return context, exten, priotiry
exten = doParkAttempt,n,Goto(return_context,return,ext, return priority)
Good luck
Bryant exten = doParkAttempt,n,Goto(return_context,return,ext, return
priority) Good luckBryant
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[asterisk-users] Asterisk 1.8.9.3 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.9.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix ACK routing for non-2xx responses. (Closes issue ASTERISK-19389. Reported by: Karsten Wemheuer) * --- Fix regressions with regards to route-set creation on early dialogs --- (Closes issue ASTERISK-19358. Reported-by: Karsten Wemheuer) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.3 Thank you for your continued support of Asterisk! -- _ -- 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] Asterisk 10.1.3 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.1.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.1.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix ACK routing for non-2xx responses. (Closes issue ASTERISK-19389. Reported by: Karsten Wemheuer) * --- Fix regressions with regards to route-set creation on early dialogs --- (Closes issue ASTERISK-19358. Reported-by: Karsten Wemheuer) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.3 Thank you for your continued support of Asterisk! -- _ -- 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] Where can I find some good examples of listening to AMI events via PHP how to listen to a specific event?
Hi everyone, I got HTTP AMI working fine here. For example this dials 1-415-999- and then sends to Extension @from-internal: http://192.168.0.100:8088/asterisk/manager?action=commandoriginateDAHDI/g0/1415999extension@from-internal However, I want to have some control over this call. I want to be notified the moment this call is hangup. I guess there would be a hangup event generated. I am not sure if that would be done through action:waitevent? or if there is another method. I am also looking for some php samples on listening for these events as I am trying to create a Web GUI for a dialer that will allow me to show status of a call in real-time like Call In Progress, Call Ended, etc... I see that too many events are generated and I am wondering if there is an easy way of listening for a particular event? Would that be ActionID? if so, how to use it? Thanks a lot -- _ -- 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] Replicating SIP registration Info between active to standby
I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a time. How can I copy sip registration information from Active Server to Standby Server Sam -- _ -- 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] Replicating SIP registration Info between active to standby
Hi. How about place backend DB on shared disk, or make replication between them? 2012/02/24 13:58 Muro, Sam resea...@businesstz.com: I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a time. How can I copy sip registration information from Active Server to Standby Server Sam -- _ -- 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] Replicating SIP registration Info between active to standby
Hi Takehiro Are you suggesting sharing the AstDB ? Sam Takehiro Matsushima wrote: Hi. How about place backend DB on shared disk, or make replication between them? 2012/02/24 13:58 Muro, Sam resea...@businesstz.com: I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a time. How can I copy sip registration information from Active Server to Standby Server Sam -- _ -- 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] Replicating SIP registration Info between active to standby
Hi, Sam Yes, I’m understanding that the backend of AstDB is bdb or sqlite(since asterisk10). So, I suggested to place files of them on shared disk (like DRBD). regards, takehiro 2012/2/24 Muro, Sam resea...@businesstz.com: Hi Takehiro Are you suggesting sharing the AstDB ? Sam Takehiro Matsushima wrote: Hi. How about place backend DB on shared disk, or make replication between them? 2012/02/24 13:58 Muro, Sam resea...@businesstz.com: I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a time. How can I copy sip registration information from Active Server to Standby Server Sam -- _ -- 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
