Re: [asterisk-users] Dial Plan Application(main-menu)
Do you have the main-menu sound file in the correct format? Goksie On 11/20/09, Steve Edwards asterisk@sedwards.com wrote: On Fri, 20 Nov 2009, aster...@opensourcesolution.in wrote: the problem is that when call comes it answers but backgroup main menu dosent play,for test purpose i had done The problem is that you do not have (or have not provided) sufficient information to solve today's problem. You should bump up logging (logger.conf, console = debug,dtmf,error,event,notice,verbose,warning) and contemplate (for a very long time) the meaning of the messages. There are resources available on the Internet (google.com, voip-info.org) where you can find answers faster and without annoying the hell out of the list as you attempt to have others write your dialplan line-by-line, day-by-day. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Development on top of freePbx Gui and AsteriskNow
Dear all, I have to develop and integrate an own application with AsteriskNOW. So create table, access them, do some action from asterisk freepbx GUI and use my data inside dialplan (e.g: to choice if a number can be dialed or no) Can someone suggest which technologies are available or link some documentation ? Thanks in advance. -- Giancarlo Lombardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Development on top of freePbx Gui and AsteriskNow
On 22 Nov 2009, at 10:46, giancarlo lombardo wrote: I have to develop and integrate an own application with AsteriskNOW. So create table, access them, do some action from asterisk freepbx GUI and use my data inside dialplan (e.g: to choice if a number can be dialed or no) Can someone suggest which technologies are available or link some documentation ? Thanks in advance. http://www.freepbx.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent Dial if any extension is busy
Magnus Benngård skrev:
Hi!
Part of extensions.conf:
exten = 985,1,Dial(SIP/0317998985H323/00702221...@avaya,20)
exten = 985,2,Goto(985-${DIALSTATUS},1)
exten = 985-BUSY,1,VoiceMail(0317998...@inputinterior.se,b)
exten = 985-BUSY,2,PlayBack(vm-goodbye)
exten = 985-BUSY,3,HangUp()
exten = 985-NOANSWER,1,VoiceMail(0317998...@inputinterior.se,u)
exten = 985-NOANSWER,2,PlayBack(vm-goodbye)
exten = 985-NOANSWER,3,HangUp()
0317998985 is a direct connected SIP phone
0702221448 is a celluar phone.
When dialing 985 both phones rings, perfect
If none answer within 20 seconds,
VoiceMail(0317998...@inputinterior.se,u), perfect
But my problem comes when I speak on 0317998985 and someone calls on
985, the call
get to my celluar phone and ofc the other way around.
Is there a way to check if any extension is busy and in that case jump
to VoiceMail(0317998...@inputinterior.se,b)?
If both phones were directly connected sip, it could be done.
The problem is that you can't determine if the cellular is busy before
you call it.
If the cell was only called via asterisk, you could set a flag, when
asterisk called extension 985, and clear it, when hanging up, but I
guess the phone is used for call out via regular cell service, and also
called directly on its own number.
You don't own the cell-company, and can setup an API to get the status
of the cell, right? I didn't think so :-)
You could do this:
check if sip is busy, using ChanIsAvail
If so, go to voicemail.
Else, dial cell, timeout 20 sec
if busy go to voicemail
else dial sip, timeout 20 sec
if not answered. go to voicemail.
But this will give 20 seconds delay before sip rings, and 40 seconds
timeout for the caller before voicemail.
The other option is to modify the source, and add an option to the
dial-command, to exit if any extension dialled is busy.
After all, this is open source :-)
Leif
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Re: [asterisk-users] Prevent Dial if any extension is busy
On Sun, 22 Nov 2009 15:38:00 +0100, Leif Neland wrote: Magnus Benngård
skrev: Hi!
Part of extensions.conf:
exten = 985,1,Dial(SIP/0317998985H323/00702221...@avaya,20)
exten = 985,2,Goto(985-${DIALSTATUS},1)
exten = 985-BUSY,1,VoiceMail(0317998...@inputinterior.se,b [1])
exten = 985-BUSY,2,PlayBack(vm-goodbye)
exten = 985-BUSY,3,HangUp()
exten = 985-NOANSWER,1,VoiceMail(0317998...@inputinterior.se,u [2])
exten = 985-NOANSWER,2,PlayBack(vm-goodbye)
exten = 985-NOANSWER,3,HangUp()
0317998985 is a direct connected SIP phone
0702221448 is a celluar phone.
When dialing 985 both phones rings, perfect
If none answer within 20 seconds, VoiceMail(0317998...@inputinterior.se,u
[3]), perfect
But my problem comes when I speak on 0317998985 and someone calls on 985,
the call
get to my celluar phone and ofc the other way around.
Is there a way to check if any extension is busy and in that case jump to
VoiceMail(0317998...@inputinterior.se,b [4])?
If both
phones were directly connected sip, it could be done.
The problem is that you can't determine if the cellular is busy before
you call it.
If the cell was only called via asterisk, you could set a flag, when
asterisk called extension 985, and clear it, when hanging up, but I guess
the phone is used for call out via regular cell service, and also called
directly on its own number.
You don't own the cell-company, and can setup an API to get the status of
the cell, right? I didn't think so :-)
No i dont own the cell-company but they route the cell-call to my main
Avaya pbx and the Avaya route it back (with a new b-number) so I have
pretty much control over the cell-call.
Just have to route it to my Asterisk and set the flag there, will do some
reading and figure out how.
You could do this:
check if sip is busy, using ChanIsAvail
I am running Asterisk SVN-branch-1.6.2-r230384 so I thougt i can do
something like:
(For checking if I am talking on the SIP phone)
exten =
985,1,GotoIf($[${DEVICE_STATE(SIP/0317998985)}=BUSY]?11)
exten = 985,2,Dial(SIP/0317998985H323/00702221...@avaya,20)
exten = 985,3,Goto(985-${DIALSTATUS},21)
exten = 985,4,HangUp()
exten = 985-BUSY,11,VoiceMail(0317998...@inputinterior.se,b)
exten = 985-BUSY,12,PlayBack(vm-goodbye)
exten = 985-BUSY,13,HangUp()
exten = 985-NOANSWER,21,VoiceMail(0317998...@inputinterior.se,u)
exten = 985-NOANSWER,22,PlayBack(vm-goodbye)
exten = 985-NOANSWER,23,HangUp()
But there is something wrong with the first line, tried INUSE aswell.
When I place a call from 0317998985 and some1 call 985, the call goes to
the cell phone. :(
Can any1 se what I am doing wrong?
If so, go to voicemail.
Else, dial cell, timeout 20 sec
if busy go to voicemail
else dial sip, timeout 20 sec
if not answered. go to voicemail.
But this will give 20 seconds delay before sip rings, and 40 seconds
timeout for the caller before voicemail.
The other option is to modify the source, and add an option to
the
dial-command, to exit if any extension dialled is busy.
After all, this is open source :-)
Leif
Links:
--
[1] mailto:0317998...@inputinterior.se,b
[2] mailto:0317998...@inputinterior.se,u
[3] mailto:0317998...@inputinterior.se,u
[4] mailto:0317998...@inputinterior.se,b
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[asterisk-users] transferring SIP call: no voice
I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk
B. Both are behind NAT, but port forwarded. I get the connection, but no
voice - either in or out.
I can call on SIP from A to B (and from B to A). Do it all the time.
Asterisk A receives SIP calls from Junction and Teliax.
CLI on A looks right:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Executing [300838...@sipgate-test:1]
Answer(SIP/sipgate-0016, ) in new stack
-- Executing [300838...@sipgate-test:2]
Goto(SIP/sipgate-0016, home,447,1) in new stack
-- Goto (home,447,1)
-- Executing [...@home:1] NoOp(SIP/sipgate-0016, x)
in new stack
-- Executing [...@home:2] NoOp(SIP/sipgate-0016,
yyy x) in new stack
-- Executing [...@home:3] Dial(SIP/sipgate-0016,
SIP/nhi-riverside-sip) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called nhi-riverside-sip
-- SIP/nhi-riverside-sip-0017 answered SIP/sipgate-0016
-- Packet2Packet bridging SIP/sipgate-0016 and
SIP/nhi-riverside-sip-0017
And on B:
-- Executing [...@incoming:1]
Answer(SIP/nhi-riverside-sip-0009, ) in new stack
-- Executing [...@incoming:2] NoOp(SIP/nhi-riverside-sip-0009,
callerid: y x) in new stack
-- Executing [...@incoming:3] Dial(SIP/nhi-riverside-sip-0009,
DAHDI/g0,60) in new stack
-- Called g0
-- DAHDI/1-1 is ringing
Asterisk A sip.conf:
[sipgate]
type=friend
secret= ;;SIP_PASSWORD
insecure=port,invite
defaultuser= ;; SIP-ID
fromuser= ;;SIP-ID
context=sipgate-test
fromdomain=sipgate.com
host=sipgate.com
outboundproxy=proxy.live.sipgate.com
qualify=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=yes
canreinvite=no
Asterisk A extensions.conf:
[sipgate-test]
exten = _X.,1,Answer()
exten = _X.,n,GoTo(home,447,1)
[home]
exten =447,1,NoOp(${CALLERID(num)})
exten =447,n,NoOp(${CALLERID(all)})
exten=447,n,Dial(SIP/nhi-riverside-sip)
And iptables on the router for Asterisk A:
$IPT -t nat -A PREROUTING -i $EXTIF -p udp --dport 5060 -j DNAT --to
10.10.10.180:5060
$IPT -A FORWARD -p udp --dport 5060 -m state --state NEW -d 10.10.10.180
-j ACCEPT
# for sip, also port forward rtp ports
$IPT -t nat -A PREROUTING -i $EXTIF -p udp --dport 1:2 -j DNAT
--to 10.10.11.180 # sip rtp
$IPT -A FORWARD -i $EXTIF -p udp --dport 1:2 -j ACCEPT
What am I missing?
sean
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[asterisk-users] Wierd problem
I have asterisk setup in my house and I have a SPA-3102-NA and a PAP2T. This is probably just me not understanding what is going on, but I was playing around last night and I used the sip unregister extension command on the CLI. I thought the boxes would re-register when their registration interval was up. This is not what is happening. Now the devices are failing to register, even my softphones (yes, I was an idiot and unregistered them all). I don't know how to clear this and get my stuff to work again. I've turned Asterisk off and back on, restarted the whole machine, power cycled the devices and pulled out some hair. Nothing seems to work. Could a kind soul help me out? Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd problem
I have asterisk setup in my house and I have a SPA-3102-NA and a PAP2T. This is probably just me not understanding what is going on, but I was playing around last night and I used the sip unregister extension command on the CLI. I thought the boxes would re-register when their registration interval was up. This is not what is happening. Now the devices are failing to register, even my softphones (yes, I was an idiot and unregistered them all). I don't know how to clear this and get my stuff to work again. I've turned Asterisk off and back on, restarted the whole machine, power cycled the devices and pulled out some hair. Nothing seems to work. Could a kind soul help me out? Tim Finally figured it out. Somethings are too simple to notice. It was merely a DNS lookup problem. (Grumble) Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] transferring SIP call: no voice
sean darcy wrote:
I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk
B. Both are behind NAT, but port forwarded. I get the connection, but no
voice - either in or out.
I can call on SIP from A to B (and from B to A). Do it all the time.
Asterisk A receives SIP calls from Junction and Teliax.
CLI on A looks right:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Executing [300838...@sipgate-test:1]
Answer(SIP/sipgate-0016, ) in new stack
-- Executing [300838...@sipgate-test:2]
Goto(SIP/sipgate-0016, home,447,1) in new stack
-- Goto (home,447,1)
-- Executing [...@home:1] NoOp(SIP/sipgate-0016, x)
in new stack
-- Executing [...@home:2] NoOp(SIP/sipgate-0016,
yyy x) in new stack
-- Executing [...@home:3] Dial(SIP/sipgate-0016,
SIP/nhi-riverside-sip) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called nhi-riverside-sip
-- SIP/nhi-riverside-sip-0017 answered SIP/sipgate-0016
-- Packet2Packet bridging SIP/sipgate-0016 and
SIP/nhi-riverside-sip-0017
And on B:
-- Executing [...@incoming:1]
Answer(SIP/nhi-riverside-sip-0009, ) in new stack
-- Executing [...@incoming:2] NoOp(SIP/nhi-riverside-sip-0009,
callerid: y x) in new stack
-- Executing [...@incoming:3] Dial(SIP/nhi-riverside-sip-0009,
DAHDI/g0,60) in new stack
-- Called g0
-- DAHDI/1-1 is ringing
Asterisk A sip.conf:
[sipgate]
type=friend
secret= ;;SIP_PASSWORD
insecure=port,invite
defaultuser= ;; SIP-ID
fromuser= ;;SIP-ID
context=sipgate-test
fromdomain=sipgate.com
host=sipgate.com
outboundproxy=proxy.live.sipgate.com
qualify=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=yes
canreinvite=no
Asterisk A extensions.conf:
[sipgate-test]
exten = _X.,1,Answer()
exten = _X.,n,GoTo(home,447,1)
[home]
exten =447,1,NoOp(${CALLERID(num)})
exten =447,n,NoOp(${CALLERID(all)})
exten=447,n,Dial(SIP/nhi-riverside-sip)
And iptables on the router for Asterisk A:
$IPT -t nat -A PREROUTING -i $EXTIF -p udp --dport 5060 -j DNAT --to
10.10.10.180:5060
$IPT -A FORWARD -p udp --dport 5060 -m state --state NEW -d 10.10.10.180
-j ACCEPT
# for sip, also port forward rtp ports
$IPT -t nat -A PREROUTING -i $EXTIF -p udp --dport 1:2 -j DNAT
--to 10.10.11.180 # sip rtp
$IPT -A FORWARD -i $EXTIF -p udp --dport 1:2 -j ACCEPT
What am I missing?
sean
FWIW, asterisk A is 1.6.0.18, B is 1.6.1.10.
sean
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[asterisk-users] Sending call information to handset
I have asterisk and linksys spa 942 phones. Normally If there are missed calls they display on the phones screen. I want to write a script that sends all missed calls to the phones screen, and email for theat extension. I need advice on where to start especially on how to send the informatio to the handset Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Portec - feedback wanted
I am thinking of buying a Portec MV370 (single channel VoIP/GSM gateway) I am after feedback reports both good and otherwise please. Thanks, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Portec - feedback wanted
Hi Michael, It does what it is announced/supposed to do. I have checked and know well all the Portech GSM/SIP family. But, be carefull, because under the same reference you can buy/receive different hardware versions : - 2, 3 or 4 GSM frequencies bands - Siemens or Simcom GSM modules So, the audio quality is best with Siemens module, and, depending of the GSM base near from your site, you must have the correct radio band ! To avoid bad surprise or issue, buy the quadband and Siemens version only. Check with your provider, because it's never well explained and you risk to have the bad one, because it's always easier to propose the lower price with the most poor version to win sales... Best Regards, Francois France -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]de la part de Michael Envoye : dimanche 22 novembre 2009 20:16 A : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Portec - feedback wanted I am thinking of buying a Portec MV370 (single channel VoIP/GSM gateway) I am after feedback reports both good and otherwise please. Thanks, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ce message entrant est certifie sans virus connu. Analyse effectuee par AVG - www.avg.fr Version: 8.5.425 / Base de donnees virale: 270.14.76/2519 - Date: 11/22/09 07:38:00 Ce message sortant est certifie sans virus connu. Analyse effectuee par AVG - www.avg.fr Version: 8.5.425 / Base de donnees virale: 270.14.76/2519 - Date: 11/22/09 07:38:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Portec - feedback wanted
On Mon, 23 Nov 2009 08:54:34 F6HQZ wrote: Hi Michael, It does what it is announced/supposed to do. I have checked and know well all the Portech GSM/SIP family. But, be carefull, because under the same reference you can buy/receive different hardware versions : - 2, 3 or 4 GSM frequencies bands - Siemens or Simcom GSM modules So, the audio quality is best with Siemens module, and, depending of the GSM base near from your site, you must have the correct radio band ! To avoid bad surprise or issue, buy the quadband and Siemens version only. Check with your provider, because it's never well explained and you risk to have the bad one, because it's always easier to propose the lower price with the most poor version to win sales... Thanks. New Zealand is 900/1800 for Vodafone and 900 for 2degrees GSM networks. I am of course buying the quad band version (In case I want to take it to Australia where they use 850 as well AFAIK). I will write back to Portech and enquire about the GSM module brand. Thanks for that. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold
thx a lot friend. On Sat, 21 Nov 2009 20:08:45 -0500, C F shma...@gmail.com wrote: On Thu, Nov 19, 2009 at 10:31 PM, aster...@opensourcesolution.in wrote: hello friends i want very simple thing in my dial plan. 1.When ever calls come at exten 2000 and if it is not answered with in 60 secs it should hangup. Set absolute timeout to 60 seconds. 2.when ever call comes at exten 2000 and if it is answered within 60 secs and if person who receives the call, puts the call on hold than music on hold should begins. Setup music on hold: http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf 3.if music on hold is placed for more than 60 secs call should hangup. As far as I know, that is impossible to do with current code, since asterisk sees an answered call the same way a call thats place on hold, therefore asterisk has no way to distinguish between being on hold or actively talking on the phone. my extention.conf is like this vi /etc/asterisk/extentions.conf exten = 2000,1,Answer() exten = 2000,n,Dial(SIP/2000,60) exten = 2000,n,Dial(SIP/2000,60,m) exten = 2000,n,Hangup the output of this is that when call is coming at exten 2000 call is answered and another call comes n first call is on hold after 60 secs music on hold starts but if i receive call before 60 secs even than MOH starts even i dont put call on hold. thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verification number / code
Un-top-posting...
On Sat, 21 Nov 2009, Thomas Perron wrote:
I want to distribute a random number to each of the first 100 callers
to my IVR. This random number will be matched to their telephone
number. Where in Asterisk can I do this? And, how?
Logic.
Call arrives. Context for announcement begins. You will receive a
authentication code at the end of the message. Then, if they press a
certain digit to confirm then I simply pass a code to them. These
codes are distributed to the first 100. The 101st call does not get a
code.
On Sat, Nov 21, 2009 at 7:20 PM, Steve Edwards
asterisk@sedwards.comwrote:
I'm guessing you really don't want a random number since a random
number generator can generate duplicates.
Matching the number to their ANI also has issues. What if my ANI is
blocked? What if I spoof my ANI? What if I call from a SIP phone?
I would pre-compute the random numbers and store them in a database.
When a call arrives, I would invoke an AGI that would lock the table,
read the first value with a null ANI, update the row with the caller's
ANI, and unlock the table.
You could do it in dialplan, but I find database access in dialplan
ugly.
Alternatively, you could mung UNIQUEID (number of seconds since
Epoch.number of channels created by this instance of Asterisk) to
appear to the caller as random and then store that and their ANI in a
database.
What happens if Asterisk is restarted in the middle of your campaign?
On Sat, 21 Nov 2009, Thomas Perron wrote:
that is a bit heavy for me. how about some simple way to announce a
random number. using RAND. and saydigit
exten = s,1,Set(junky=${RAND(1,8)})
Um. OK. Use RAND and saydigit. And use some sort of counter to know when
you've issued 100 numbers.
But...
) You may have duplicate random numbers.
) You may have issues counting the callers if you have more than 1 call
arrive at close to the same time.
) If Asterisk restarts, how will you know how many numbers have been
issued?
) You still have to handle the matching requirement.
I still think a databased* approach is the best approach. I'm sorry it's
not within your current skill-set. Maybe this means you should invest some
time learning the skills or hire someone who has them.
If this is a toy (Guess a number...), use RAND and saydigit. If this is
a contest with some sort of prize or involves anything of value, use a
database.
*) MySQL has a really neat generic lock facility that will come in handy.
See get_lock(lock-name, timeout).
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] End to End delay calculation
Hi! I am looking to calculate the end-to-end delay between two soft phone/hard phone. I have asterisk server and configured ntp server on the same machine and synchronized it with ntp pool. I have seen that Wireshark can be used to check the jitter. But I am not sure how can i calculate the end to end. May be this is not related to the mailing list topic but please help me if anyone has some information. Regards, _ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Yealink SIP-T22P Auto Provisioning via HTTP ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, I have come across the above handset a few times in the UK, They are quite cheap over here (~£80) Not the best handset in the world but works well enough. I have been asked to setup a central config server for a large collection of these handsets. I know they can do Auto provisioning via FTP/HTTP/TFTP I have got an example of the generic Firmware .cfg file that allows you to upgrade all handsets firmwares, All of the SIP-T22P handsets look for a file called y0005.cfg @ the URL configured on the device. This was easy to find an example of the file. via the Docs on the Yealink site and Google. The issue is that I have not been able to find an example of the per handset .cfg file... Like the Cisco handsets the file should be named after the MAC address of the handset, and should be in the same location as the y0005.cfg file, I just cant find any examples... I have also tried to E-mail Yealink support without success.. Any help/pointers/how-tos or Examples would be brilliant. Thank You for your help and time. - -- Gavin Spurgeon. AKA Da Geek - -- The happiest of people don't necessarily have the best of everything, they just make the most of everything that comes along their way.. -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.12 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAksJ0fQACgkQvp6arS3vDiq/5gCfZ5zxDQDyIVTCKTcbdfrKLo9C oIkAnAg2B7Vj8Od0XuJz5oMS6sJr1jiM =n0ST -END PGP SIGNATURE- -- This message was scanned by DaGeek Spam Filter and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yealink SIP-T22P Auto Provisioning via HTTP ? (Solved)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List (Again) On 23/11/2009 00:06, Gavin Spurgeon wrote: I have come across the above handset a few times in the UK, They are quite cheap over here (~£80) Not the best handset in the world but works well enough. I have been asked to setup a central config server for a large collection of these handsets. I know they can do Auto provisioning via FTP/HTTP/TFTP I have got an example of the generic Firmware .cfg file that allows you to upgrade all handsets firmwares, All of the SIP-T22P handsets look for a file called y0005.cfg @ the URL configured on the device. This was easy to find an example of the file. via the Docs on the Yealink site and Google. The issue is that I have not been able to find an example of the per handset .cfg file... Like the Cisco handsets the file should be named after the MAC address of the handset, and should be in the same location as the y0005.cfg file, I just cant find any examples... I have also tried to E-mail Yealink support without success.. Any help/pointers/how-tos or Examples would be brilliant. Thank You for your help and time. To answer my own Question... The Yealink UK site (http://www.yealink.co.uk) has some very useful stuff in teh downloads section @ http://www.yealink.co.uk/downloads/ Like an 'Auto Provisioning Overview - Technical Document' @ (http://www.yealink.co.uk/assets/Document-Downloads/Auto%20Provision%20Manual%20V1.2.1.zip) with Docs Example .cfg files The yealink.com is not as useful... - -- Gavin Spurgeon. AKA Da Geek - -- The happiest of people don't necessarily have the best of everything, they just make the most of everything that comes along their way.. -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.12 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAksJ3OoACgkQvp6arS3vDioyYQCgh7Ug20FeCcQa0b0Q4r0yjfmN UDkAoLFki81ocYE7H0GDz5H5ipA4qO9B =GneW -END PGP SIGNATURE- -- This message was scanned by DaGeek Spam Filter and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7961 - can't place calls
snip Thanks for the reply. I am not getting any output from the Asterisk CLI when I place the call. The phone give busy signal as soon as I push the first digit of the extension #. When I call the 7961 from another extension I get the following on the CLI - that works fine. /snip If the phone gives a fast busy AS SOON as you type a digit, the problem is likely that you need to edit your dialplan.xml file on your TFTP server, so that the phone knows not to send digits immediately after you start typing: Contents of dialplan.xml (customize to fit your situation): DIALTEMPLATE TEMPLATE MATCH=91.. TIMEOUT=0/ TEMPLATE MATCH=9[2-9].. TIMEOUT=0/ TEMPLATE MATCH=10. TIMEOUT=0/ TEMPLATE MATCH=5.. TIMEOUT=0/ TEMPLATE MATCH=605 TIMEOUT=0/ TEMPLATE MATCH=* TIMEOUT=10/ /DIALTEMPLATE -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] End to End delay calculation
On Sun, 22 Nov 2009, capricorn 80 wrote: I am looking to calculate the end-to-end delay between two soft phone/hard phone. I have asterisk server and configured ntp server on the same machine and synchronized it with ntp pool. I have seen that Wireshark can be used to check the jitter. But I am not sure how can i calculate the end to end. May be this is not related to the mailing list topic but please help me if anyone has some information. A very long time ago, I made the mistake of letting a client listen (with a handset on each side of his head) to end-to-end delay. This all of a sudden became a quest for the Holy Grail to quantify and reduce the delay. I got a couple of RadioShack telephone recording interfaces, connected one to each endpoint. Then I connected the outputs to the left and right channels on a PC and recorded tapping on one of the handsets using Audacity. When I selected the interval between the tap and the ping, Audacity would show the time in ms. All very old-school but it worked and the client never questioned the pretty pictures on the computer screen. Wireshark may be able to tell you how long it takes a packet to travel across your network, but what about the time from the network interface on the host until sound comes out the earpiece? How long does it take a SIP phone to take a packet off it's network interface, wiggle it through it's jitter buffer, transcode it, convert it to analog and deliver it to the earpiece? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7961 - can't place calls
Thanks for the reply. I am not getting any output from the Asterisk CLI when I place the call. The phone give busy signal as soon as I push the first digit of the extension #. When I call the 7961 from another extension I get the following on the CLI - that works fine. On Sun, 22 Nov 2009, David Gibbons wrote: If the phone gives a fast busy AS SOON as you type a digit, the problem is likely that you need to edit your dialplan.xml file on your TFTP server, so that the phone knows not to send digits immediately after you start typing: Contents of dialplan.xml (customize to fit your situation): DIALTEMPLATE TEMPLATE MATCH=91.. TIMEOUT=0/ TEMPLATE MATCH=9[2-9].. TIMEOUT=0/ TEMPLATE MATCH=10. TIMEOUT=0/ TEMPLATE MATCH=5.. TIMEOUT=0/ TEMPLATE MATCH=605 TIMEOUT=0/ TEMPLATE MATCH=* TIMEOUT=10/ /DIALTEMPLATE My dialplan.xml looks like: DIALTEMPLATE TEMPLATE MATCH=#... TIMEOUT=5 USER=Phone / TEMPLATE MATCH=* TIMEOUT=5 USER=Phone / TEMPLATE MATCH=1.. TIMEOUT=0 TONE=Bellcore-Alerting USER=Phone / /DIALTEMPLATE It seems to work OK with my Asterisk server. Telnet into the phone and enter show config Do defaultgw, outbound_proxy, outbound_proxy_port, proxy1_address, proxy1_port, proxy_register, timer_register_expires seem reasonable? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] End to End delay calculation
Hi ! Yea lot of things to look but what in case of sip phone to sip phone ? Is there anyway we can do it with some open source tool ? I have to do it for my experiment and I am really worried about it. Regards, Date: Sun, 22 Nov 2009 17:12:22 -0800 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] End to End delay calculation On Sun, 22 Nov 2009, capricorn 80 wrote: I am looking to calculate the end-to-end delay between two soft phone/hard phone. I have asterisk server and configured ntp server on the same machine and synchronized it with ntp pool. I have seen that Wireshark can be used to check the jitter. But I am not sure how can i calculate the end to end. May be this is not related to the mailing list topic but please help me if anyone has some information. A very long time ago, I made the mistake of letting a client listen (with a handset on each side of his head) to end-to-end delay. This all of a sudden became a quest for the Holy Grail to quantify and reduce the delay. I got a couple of RadioShack telephone recording interfaces, connected one to each endpoint. Then I connected the outputs to the left and right channels on a PC and recorded tapping on one of the handsets using Audacity. When I selected the interval between the tap and the ping, Audacity would show the time in ms. All very old-school but it worked and the client never questioned the pretty pictures on the computer screen. Wireshark may be able to tell you how long it takes a packet to travel across your network, but what about the time from the network interface on the host until sound comes out the earpiece? How long does it take a SIP phone to take a packet off it's network interface, wiggle it through it's jitter buffer, transcode it, convert it to analog and deliver it to the earpiece? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Keep your friends updated—even when you’re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do I take out one office out of the call stream?
Hi, I have two locations A and B. I have calls coming in into both locations but they are answered only by location A, location B forwards all calls to location A to be answered. Now when I have a call coming into location B then the call gets transferred to Location A then transferred to location B again it seem like the location A is still in the stream. Is there a way of taking it out of the stream? Thanks, robert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hardware echo cancellation
I got a few newbie questions. If I get an echo cancellation module for my Digium TE121 card, will I need to do any adjustments/configuration in Asterisk? Is the hardware better than the software version? TIA! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme 'o' - what actually it does..??
Hi Can someone explain me what is the purpose for MeetMe Option 'o'.. If I defined 'o' with MeetMe option or If not defined with MeetMe option... What is the difference between these two if defined or not defined MeetMe 'o' option... -- Regards, Chandrakant Solanki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
