[asterisk-users] sip calls not going through
Hello,
i've recently configured my asterisk for internal sip calls.
while testing, i noticed that 1 out of 10 calls works..
at first i thought my router dropping packets around the way as it were a
bottle neck..
so i've added a switch.
once i tested again same prob occurs...
im using xlite as a softphone on clients pc
and centos server on a dedicated machine.
at times the phone call goes through and voice is perfect..
and at others one side can hear me yet i cant hear them.. and at others neither
one of us can hear the other end..
i've checked my logs and havent found anything relevant.. but yet again maybe
you could as i'm a newbie..
Registered SIP '101' at 192.168.75.192 port 22162
-- Saved useragent "X-Lite release 1014k stamp 47051" for peer 101
-- Executing [...@spa:1] Dial("SIP/100-0967ad88", "SIP/101|15") in new stack
-- Called 101
-- SIP/101-09683690 is ringing
-- SIP/101-09683690 answered SIP/100-0967ad88
-- Native bridging SIP/100-0967ad88 and SIP/101-09683690
== Spawn extension (spa, 101, 1) exited non-zero on 'SIP/100-0967ad88'
-- Executing [...@spa:1] Dial("SIP/101-09676378", "SIP/100|15") in new stack
-- Called 100
-- SIP/100-0967ad88 is ringing
-- SIP/100-0967ad88 answered SIP/101-09676378
-- Native bridging SIP/101-09676378 and SIP/100-0967ad88
== Spawn extension (spa, 100, 1) exited non-zero on 'SIP/101-09676378'
-- Executing [...@spa:1] Dial("SIP/100-09676378", "SIP/101|15") in new stack
-- Called 101
-- SIP/101-09677b10 is ringing
-- SIP/101-09677b10 answered SIP/100-09676378
-- Native bridging SIP/100-09676378 and SIP/101-09677b10
== Spawn extension (spa, 101, 1) exited non-zero on 'SIP/100-09676378'
-- Unregistered SIP '100'
-- Registered SIP '100' at 192.168.75.139 port 14226
-- Saved useragent "X-Lite release 1014k stamp 47051" for peer 100
-- Executing [...@spa:1] Dial("SIP/100-096792c8", "SIP/101|15") in new stack
-- Called 101
-- SIP/101-09683690 is ringing
-- SIP/101-09683690 answered SIP/100-096792c8
-- Native bridging SIP/100-096792c8 and SIP/101-09683690
== Spawn extension (spa, 101, 1) exited non-zero on 'SIP/100-096792c8'
-- Unregistered SIP '100'
-- Registered SIP '100' at 192.168.75.139 port 41372
-- Saved useragent "X-Lite release 1014k stamp 47051" for peer 100
-- Executing [...@spa:1] Dial("SIP/100-096792c8", "SIP/101|15") in new stack
-- Called 101
-- SIP/101-09683690 is ringing
-- SIP/101-09683690 answered SIP/100-096792c8
-- Native bridging SIP/100-096792c8 and SIP/101-09683690
== Spawn extension (spa, 101, 1) exited non-zero on 'SIP/100-096792c8'
-- Unregistered SIP '100'
-- Executing [...@spa:1] Dial("SIP/101-09677b10", "SIP/100|15") in new stack
[Jun 10 09:37:54] WARNING[7880]: app_dial.c:1237 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [...@spa:2] VoiceMail("SIP/101-09677b10", "1...@default") in
new stack
-- Playing 'vm-intro' (language 'en')
-- Registered SIP '100' at 192.168.75.139 port 58704
-- Saved useragent "X-Lite release 1014k stamp 47051" for peer 100
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/100/tmp/zWOOql
format: wav, 0x967d7b0
-- User hung up
== Spawn extension (spa, 100, 2) exited non-zero on 'SIP/101-09677b10'
-- Executing [...@spa:1] Dial("SIP/101-09677b10", "SIP/100|15") in new stack
-- Called 100
-- SIP/100-0967ad88 is ringing
-- SIP/100-0967ad88 answered SIP/101-09677b10
-- Native bridging SIP/101-09677b10 and SIP/100-0967ad88
== Spawn extension (spa, 100, 1) exited non-zero on 'SIP/101-09677b10'
-- Unregistered SIP '100'
-- Registered SIP '100' at 192.168.75.139 port 14744
-- Saved useragent "X-Lite release 1014k stamp 47051" for peer 100
-- Unregistered SIP '101'
-- Executing [...@spa:1] Dial("SIP/100-096792c8", "SIP/101|15") in new stack
[Jun 10 09:39:23] WARNING[7885]: app_dial.c:1237 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [...@spa:2] VoiceMail("SIP/100-096792c8", "1...@default") in
new stack
-- Playing 'vm-intro' (language 'en')
== Spawn extension (spa, 101, 2) exited non-zero on 'SIP/100-096792c8'
-- Registered SIP '101' at 192.168.75.192 port 34518
-- Saved useragent "X-Lite release 1014k stamp 47051" for peer 101
-- Executing [...@spa:1] Dial("SIP/100-0967f670", "SIP/101|15") in new stack
-- Called 101
-- SIP/101-09684cb0 is ringing
-- SIP/101-09684cb0 answered SIP/100-0967f670
-- Native bridging SIP/100-0967f670 and SIP/101-09684cb0
== Spawn extension (spa, 101, 1) exited non-zero on 'SIP/100-0967f670'
-- Executing [...@spa:1] Dial("SIP/101-0967f670", "SIP/100|15") in new stack
[asterisk-users] Advice
Hi all, a few month ago I got the task of setting up asterisk for my company. I had 94 employee to set this up for ... I never heard of asterisk before to b honest, so after researching a bit.. I started with a digium card with ZAP though that didn't work out as the card were flawed.. so ended up setting up SIP for everyone using a SIP callcentric accounts as well as sipura for pstn lines.. now it's working at it's minimal state.. but as am out of the heat of pressure from management.. so now It's time to learn about asterisk the right way as I had lots of help from this mailing list as well as the IRC channel that I'm not sure I could do it again on my own.. so not to add more to my email, I'm seeking advice about the proper way to learn about asterisk from A to Z if possible... any advice would be appreciated thanks in advance, Roland<>___ -- 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] tcpdum
Hi Michel, how's beirut's weather with ya! anyway, TTL stands for TIME To LIVE. it's encapsulated on layer three of the OSI layer to each packet going out that specific interface. by default routers has a 16 TTL that means each time the designated packet reaches a router (gets decapsulated) it gets a -1... this helps in preventing loops which would eventually lead to congestion. now latency wise, for VOIP to operate correctly it needs a latency of under 200 ms. (I currently have a microwave link , and unfortunately im not getting that a latency less than 280 to my SIP provider) if your asterisk server is hosted online, you could simply traceroute it and check the highest latency, point. and depending on where that bottle neck would be, youll troubleshoot from there.. mine were on my ISP's international link, after having a meeting with my account manager, I got my link routed through a different international path which drastically decreased my latency. now on a different approach, you absolutly have to talk to your ISP/network administrator to provide you QOS for that specific IP whether it's public or private. depending on your network's traffic QOS would surely help with no doubt.. this would decrease latency as well hope I've shed some light about this, if not well the more knowledge the betteR best, Roland From: michel freiha Sent: Monday, December 15, 2008 10:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users-boun...@lists.digium.com Subject: Re: [asterisk-users] tcpdum Dear Sir, There is no relation between TTL and the latency on asterisk server? Regards On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere wrote: TTL is part of the UDP header (Time To Live). It isn't really about the voice at all. Length 345 is the number of bytes in the packet. j On Mon, 15 Dec 2008, michel freiha wrote: > *Dear All, > I run the below tcp dump on my asterisk server > > tcpdump -i eth0 -n -s0 -v udp port 5060 > > I got the following result > > 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, > length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 > > What i need to know please what TTL means specifically and what is the best > value og TTL and what is the lengh vale mean > > Regards* > ___ -- 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
[asterisk-users] sip to sip unplanned conference! help!!
first of all my topology is as such:Softphones<<-->> asterisk <<-->
sipurasoftphone with peer number 100, calls another softphone with peer number
as 200. (both has asterisk as gateway)relevant extensions.conf:
exten => _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten => _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten => _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line
is busy or unavailable
exten => _2XX,3,HangUp()
relevant sip.conf:
[200]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[200]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
in the meantime, an incoming call comes through Sipura which is directed to:
[incoming-samer]
exten => 201,1,Answer() ; Answer inbound calls
exten => 201,2,Playback(silence/1)
exten => 201,3,Background(joyce) ; input an extension
exten => 201,4,WaitExten(8)
exten => 201,5,Dial(SIP/220,15)
exten => 201,4,Wait(8)
include => spa
exten => 201,n,Hangup()
suddenly, the first conversation between 100 and 200, hears the attendant audio
message "joyce" welcoming the caller(the one calling sipura in a completely
different call) and listens to the entire conversation that the incoming caller
is having..
_
Invite your mail contacts to join your friends list with Windows Live Spaces.
It's easy!
http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip conversations overlapping!!!!
i appologize for not making myself clear.. i have my asterisk box, connexted to 4 sipura3102.. these sipuras has 4 PSTN lines connected to them through one cable, which has 8 lines inside of it (2 connected to an RJ11 and plugged into its respecitve fxs port in the sipura) on the other side, i have 20 softphones.. these softphones has asterisk as their gateway.. where they could call eachother! or call/recieve calls through any of the sipuras... my prob is as such: when i call from softphone#1 to sipura #1, sound is pretty good and everything is working perfectly.. though if asterisk recieves a call from another sipura.. lets say its sipura #2, then! i could hear the attendnat answering the incoming phone in my current conversation, and i could hear some1 picking up and answerinfg the call..! if i ask them to hang up! my line breaks as well.. Date: Fri, 29 Aug 2008 10:40:57 +0200 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sip conversations overlapping >Every one PSTN line connected to the FXS port of sipura.. >Though these 4 lines comes in one cable if that has to do with anything! Not clear for me, develop some more you topology. _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip conversations overlapping!!!!
Every one PSTN line connected to the FXS port of sipura.. Though these 4 lines comes in one cable if that has to do with anything! > Date: Thu, 28 Aug 2008 14:10:53 -0400 > From: [EMAIL PROTECTED] > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] sip conversations overlapping > > RoLaNd RoLaNd wrote: > > > > Hi all, > > > > i'm facing this weird prob...my topology is as such: > > > > > > > > - > > - > > > > when am on a call, sometimes when some1 else tries to call out.. i > > hear the actual tones which ends up preventing the other end from > > talking to me.. > > moroever, when some1 calls me through one sipura, while im talking on > > another... i can hear the attendant welcoming message, then i hear the > > voice of whoever have picked tht line up..! and if i ask that person > > to hang up... my line breaks as well..! > > can any1 help me with this issue! > > below is my config: > > > > How are the analogue phones wired? One phone plugged directly to one > 3102 FXS port? or is there common wiring ? > Are all the FXO ports connected to telco lines? > > > regards, > > Drew > > NOTE: Holding the key down whilst typing the first person, > singular, pronoun will produce stunningly readable results. Either > key will do, you can even use the key if both of > those are broken/can't locate them. You can also use this procedure for > the first letter of each sentence, it makes everything much easier to read. > > -- > Drew Gibson > > Systems Administrator > OANDA Corporation > www.oanda.com > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip conversations overlapping!!!!
Hi all,
i'm facing this weird prob...my topology is as such:
-
-
when am on a call, sometimes when some1 else tries to call out.. i hear the
actual tones which ends up preventing the other end from talking to me..
moroever, when some1 calls me through one sipura, while im talking on another...
i can hear the attendant welcoming message, then i hear the voice of whoever
have picked tht line up..! and if i ask that person to hang up... my line
breaks as well..!
can any1 help me with this issue!
below is my config:
extensions.conf
[incoming-conference]
exten => 333,1,Answer() ; Answer inbound calls
exten => 333,2,Playback(silence/1)
exten => 333,3,Background(joyce) ; input an extension
exten => 333,4,WaitExten(8)
exten => 333,5,Dial(SIP/310,15)
exten => 333,4,Wait(8)
include => spa
exten => 333,n,Hangup()
[incoming-samer]
exten => 334,1,Answer() ; Answer inbound calls
exten => 334,2,Playback(silence/1)
exten => 334,3,Background(joyce) ; input an extension
exten => 334,4,WaitExten(8)
exten => 334,5,Dial(SIP/330,15)
exten => 334,4,Wait(8)
include => spa
exten => 334,n,Hangup()
[incoming-gilberte]
exten => 335,1,Answer() ; Answer inbound calls
exten => 335,2,Playback(silence/1)
exten => 335,3,Background(joyce) ; input an extension
exten => 335,4,WaitExten(8)
exten => 335,5,Dial(SIP/350,15)
exten => 335,4,Wait(8)
include => spa
exten => 335,n,Hangup()
[incoming-line4]
exten => 336,1,Answer() ; Answer inbound calls
exten => 336,2,Playback(silence/1)
exten => 336,3,Background(joyce) ; input an extension
exten => 336,4,WaitExten(8)
exten => 336,5,Dial(SIP/340,15)
exten => 336,4,Wait(8)
include => spa
exten => 336,n,Hangup()
[sipura-line]
exten => 301,1,Answer() ; Answer inbound calls
exten => 301,2,Playback(silence/1)
exten => 301,3,Background(simzy1) ; input an extension
exten => 301,4,WaitExten(8)
exten => 301,5,Dial(SIP/100,15) ; goes to operator
exten => 301,4,Wait(8)
include => spa
[spa]
exten =>_301,1,GoTo(sipura-line,${EXTEN},1)
exten =>_333,1,GoTo(incoming-conference,${EXTEN},1)
exten => _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten => _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
will ring 3 times
exten => _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
line is busy or unavailable
exten => _2XX,3,HangUp()
exten => _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten => _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
line is busy or unavailable
exten => _3XX,3,HangUp()
exten =>_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =>_01,2,Set(TIMEOUT(absolute)=5)
exten =>_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =>_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =>_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =>_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =>_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten => 303,1,VoicemailMain ; voicemail box to be redirected to
sip.conf:
[300]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
canreinvite=yes
[301]
type=friend
host=dynamic
secret=1234
context=sipura-line
[EMAIL PROTECTED]
[304]
type=friend
host=dynamic
secret=1234
context=sipura-line2
[EMAIL PROTECTED]
[305]
type=friend
host=dynamic
secret=1234
context=incoming-samer
[EMAIL PROTECTED]
[306]
type=friend
host=dynamic
secret=1234
context=incoming-gilberte
[EMAIL PROTECTED]
[333]
type=friend
host=dynamic
secret=1234
context=incoming-conference
[EMAIL PROTECTED]
[334]
type=friend
host=dynamic
secret=1234
context=incoming-samer
[EMAIL PROTECTED]
[335]
type=friend
host=dynamic
secret=1234
context=incoming-gilberte
[EMAIL PROTECTED]
[336]
type=friend
host=dynamic
secret=1234
context=incoming-line4
[EMAIL PROTECTED]
[307]
type=friend
host=dynamic
secret=1234
context=incoming-conference
[EMAIL PROTECTED]
[308]
type=friend
host=dynamic
secret=1234
context=incoming-line4
[EMAIL PROTECTED]
[310]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[320]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[330]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[340]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[350]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[107]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[150]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[100]
type=friend
host=dynamic
secret=1234
contex
Re: [asterisk-users] Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura
i kinda have a relevant prob! my sipura 3102 wont pass CID to asterisk even though ive enabled such a feature in its web gui! > Date: Wed, 27 Aug 2008 12:07:51 -0600 > From: [EMAIL PROTECTED] > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Off-Hook (type II) CID passing to Asterisk via > Linsys/Sipura > > Does anybody have an idea how to pass Off Hook caller ID to Asterisk via > Linksys ? > I'm getting caller ID type I OK but when another customer rings the phone > (when I'm on line) the CID off hook is not coming through. > I think Off-Hook CID is called CID type II, isn't it? > > -- > #Joseph > GPG KeyID: ED0E1FB7 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] entering a password to have access to a sip account?!
Hello Steve,
thanks for the advice :)
though one prob! if i add the authenticate line itll require all callers to
enter 1234 to access *ANY* sip account..
even though this would come in handy at some point but at the moment i just
want to deny the extension 300 from being able to call "01" unless the caller
entered a password..
find below wht i did so far..
[sipura-line]
exten => 301,1,Answer() ; Answer inbound calls
exten => 301,2,Playback(silence/1)
exten => 301,3,Background(simzy1) ; input an extension
exten => 301,4,authenticate(1234)
exten => 301,5,WaitExten(8)
exten => 301,6,Dial(SIP/100,15) ; goes to operator
exten => 301,3,Wait(8)
include => spa
exten => _XXX,6,VoiceMail([EMAIL PROTECTED])
exten => 301,n,Hangup()
[spa]
exten =>_301,1,GoTo(sipura-line,${EXTEN},1)
exten => _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten => _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten => _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line
is busy or unavailable
exten => _2XX,3,HangUp()
exten => _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten => _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if line
is busy or unavailable
exten => _3XX,3,HangUp()
exten =>_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =>_01,2,Set(TIMEOUT(absolute)=5)
exten =>_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =>_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten => 303,1,VoicemailMain ; voicemail box to be redirected to
> Date: Sun, 24 Aug 2008 12:05:02 -0400
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] entering a password to have access to a sip
> account?!
>
> You want to use Authenticate() between answer and dial.
>
> http://www.google.com/search?q=asterisk+authenticate&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a
>
> Thanks,
> Steve Totaro
>
> On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd <[EMAIL PROTECTED]> wrote:
> >
> >
> > Hi all,
> >
> > i;m obviously a newbie, its been 2 days that im trying to figure out a way
> > to deny a specific extension (300) from calling another specific extensions
> > (03) except if the caller punch a specified password.. sorry if im not
> > explaining myself well.. heres an example:
> >
> > i called my pstn line(with 300 as its sip account), an attendant answers and
> > asks me to punch in an extension number right now if i dial "03" it rings at
> > the other end! though i dont want that to happen! i want to set asterisk up
> > in a way tht if i dial "03" from "300" to ask me for a password... or it
> > wont let the line go through!
> >
> >
> > can anyone guide me through this issue! im really going crazy to get this
> > done! any help would truly and utterly be appreciated:)
> >
> >
> >
> > ps: find below my extensions.conf
> >
> >
> > [sipura-line]
> > exten => 301,1,Answer() ; Answer inbound calls
> > exten => 301,2,Playback(silence/1)
> > exten => 301,3,Background(simzy1) ; input an extension
> > exten => 301,4,WaitExten(8)
> > exten => 301,5,Dial(SIP/100,15) ; goes to operator
> > exten => 301,4,Wait(8)
> > include => spa
> > exten => _XXX,6,VoiceMail([EMAIL PROTECTED])
> > exten => 301,n,Hangup()
> >
> >
> >
> >
> > [spa]
> > exten =>_301,1,GoTo(sipura-line,${EXTEN},1)
> > exten => _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
> > will ring 3 times
> > exten => _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
> > line
> > is busy or unavailable
> > exten => _1XX,3,HangUp()
> > exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it
> > will ring 3 times
> > exten => _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
> > line is busy or unavailable
> > exten => _2XX,3,HangUp()
> > exten => _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
> > will ring 3 times
> > exten => _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
> > line is busy or unavailable
> > exten => _3XX,3,HangUp()
> > exten =>_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
> > ;exten =>_01,2,Set(TIMEOUT(absolute)=5)
> > exten =>_02,1,Dial(SI
[asterisk-users] entering a password to have access to a sip account?!
Hi all,
i;m obviously a newbie, its been 2 days that im trying to figure out a way to
deny a specific extension (300) from calling another specific extensions (03)
except if the caller punch a specified password.. sorry if im not explaining
myself well.. heres an example:
i called my pstn line(with 300 as its sip account), an attendant answers and
asks me to punch in an extension number right now if i dial "03" it rings at
the other end! though i dont want that to happen! i want to set asterisk up in
a way tht if i dial "03" from "300" to ask me for a password... or it wont let
the line go through!
can anyone guide me through this issue! im really going crazy to get this done!
any help would truly and utterly be appreciated:)
ps: find below my extensions.conf
[sipura-line]
exten => 301,1,Answer() ; Answer inbound calls
exten => 301,2,Playback(silence/1)
exten => 301,3,Background(simzy1) ; input an extension
exten => 301,4,WaitExten(8)
exten => 301,5,Dial(SIP/100,15) ; goes to operator
exten => 301,4,Wait(8)
include => spa
exten => _XXX,6,VoiceMail([EMAIL PROTECTED])
exten => 301,n,Hangup()
[spa]
exten =>_301,1,GoTo(sipura-line,${EXTEN},1)
exten => _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten => _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten => _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line
is busy or unavailable
exten => _2XX,3,HangUp()
exten => _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it
will ring 3 times
exten => _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if line
is busy or unavailable
exten => _3XX,3,HangUp()
exten =>_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =>_01,2,Set(TIMEOUT(absolute)=5)
exten =>_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =>_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =>_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =>_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =>_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten => 303,1,VoicemailMain ; voicemail box to be redirected to
_
News, entertainment and everything you care about at Live.com. Get it now!
http://www.live.com/getstarted.aspx___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5 min limitation on phone calls! how to!
i tried that before.. it didnt actually work! it the call kept on going well
beyound the allowed test seconds...
heres my extensions.conf:
[sipura-line]
exten => 301,1,Answer() ; Answer inbound calls
exten => 301,2,Playback(silence/1)
exten => 301,3,Background(simzy1) ; input an extension
exten => 301,4,WaitExten(8)
exten => 301,5,Dial(SIP/100,15) ; goes to operator
exten => 301,4,Wait(8)
include => spa
exten => _XXX,6,VoiceMail([EMAIL PROTECTED])
exten => 301,n,Hangup()
[spa]
exten =>_301,1,GoTo(sipura-line,${EXTEN},1)
exten => _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so it will
ring 3 times
exten => _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so it will
ring 3 times
exten => _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line
is busy or unavailable
exten => _2XX,3,HangUp()
exten => _3XX,1,Dial(SIP/${EXTEN},15) ; each ring equals to 5 seconds so it
will ring 3 times
exten => _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if line
is busy or unavailable
exten => _3XX,3,HangUp()
exten =>_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
exten =>_01,2,Set(TIMEOUT(absolute)=5)
exten =>_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =>_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =>_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =>_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =>_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten => 303,1,VoicemailMain ; voicemail box to be redirected to
> Date: Thu, 21 Aug 2008 20:26:48 +0300
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!
>
> RoLaNd RoLaNd schrieb:
> > Hello all!
> >
> > my last month's phone bill sky rocketed after i setup asterisk with
> > softphones all over the house!
> >
> > could someone help me set up a limitation for my wife and kids not to be
> > able to talk for more than 5 min at a time!
> > or like 20 min per week! or whtever limitation i could set for this!
>
> Set(TIMEOUT(absolute)=seconds)
>
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout
>
>
> Terve,
> Stefan
>
> --
> Last words of a stormchaser:
> "Where is that rotation on the radar?!"
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
_
Discover the new Windows Vista
http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 5 min limitation on phone calls! how to!
Hello all! my last month's phone bill sky rocketed after i setup asterisk with softphones all over the house! could someone help me set up a limitation for my wife and kids not to be able to talk for more than 5 min at a time! or like 20 min per week! or whtever limitation i could set for this! any help would trull be appreciated:) _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] opening Doors with Asterisk!?
Hello all, i read a few articles online about the possibility to setup a "buzzer" door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] list of minutes spent on SIP phone calls?! any advice?!
Hi All, i have asterisk with 9 SIP accounts on it. i was wondering if theres a way to setup asterisk, to send the amount of minutes each SIP account have spent incoming as well as outgoing and if possible the number it called! any advice?! any help would truly be appreciated..! thanks in advance and best regards, _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] removing == Parsing '/etc/asterisk/manager.conf': Found from CLI!
hi all, is there any way of removing this line from showing on the console? my verbosity level is 3. and this is the following output on cli 24/7 unless its interrupted by the msgs tht really counts like connected sip and so on.. [Jul 4 10:32:38] NOTICE[18542]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:32:48] NOTICE[18543]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:32:58] NOTICE[18544]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:33:08] NOTICE[18545]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' == Connect attempt from '127.0.0.1' unable to authenticate -- Registered SIP '179' at 192.168.0.2 port 27780 expires 3600 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Jul 4 10:33:18] NOTICE[18547]: manager.c:1015 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'admin' _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?
Hey Matt!! thanks for the advice! appreciate it.. just installed it and everything worked fine ( i got internet calling in my menu) though i cant seem to access the editing tool.. keeps on giving me some error even after soft reseting.. any idea?! From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Thu, 3 Jul 2008 13:11:40 -0400 Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client? Hi Roland, Did you try: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windows-mobile-6x-for-free-voip-calls-using-asterisk/ We have this successfully working on a Touch (ELF), and a HTC Tilt (Tytn II) Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd Sent: Thursday, July 03, 2008 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client? Hey! i'm facing the same prob.. i bought an HTC vox (s710) 2 weeks ago, and im still looking for a sip client..! so far i found these 3: AGEphone mobile: http://www.ageet.com/ SJphone: http://www.sjlabs.com/sjp.html Bria Mobile: http://www.counterpath.com/enterprise-mobility-gateway.html so far i just tried AgePhone (trial mode) sound is great though im facing 2 problems with it: 1: u can only use handsfree option, tht mean its a privacy killer. 2: you cant get a dial tone on it, tht means if you got 2 stage dialing on your asterisk (if u call an extension, and wait to hear a dialtone) it wont work. as for the other 2 i didnt try them yet... ps: if you found out anything else bout this matter id appreciate if you could let me know :) > Date: Mon, 30 Jun 2008 21:51:57 +0200 > From: [EMAIL PROTECTED] > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Windows Mobile 6 IAX/SIP client? > > I just bought a HTC TyTn II phone, but unfortunately it doesn't even have > a SIP client in it. > > I tried the wiki searching for a SIP or IAX client but only found some > PocketPC stuff (Windows Mobile 2003). > > Does anyone know of a good quality SIP or IAX softphone that will run on > Windows Mobile 6? > > I only have a data subscription, no voice so the quality should be > sufficient to be used constantly. > > Thanks!! > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users Discover the new Windows Vista Learn more! _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?
Hey! i'm facing the same prob.. i bought an HTC vox (s710) 2 weeks ago, and im still looking for a sip client..! so far i found these 3: AGEphone mobile: http://www.ageet.com/ SJphone: http://www.sjlabs.com/sjp.html Bria Mobile: http://www.counterpath.com/enterprise-mobility-gateway.html so far i just tried AgePhone (trial mode) sound is great though im facing 2 problems with it: 1: u can only use handsfree option, tht mean its a privacy killer. 2: you cant get a dial tone on it, tht means if you got 2 stage dialing on your asterisk (if u call an extension, and wait to hear a dialtone) it wont work. as for the other 2 i didnt try them yet... ps: if you found out anything else bout this matter id appreciate if you could let me know :) > Date: Mon, 30 Jun 2008 21:51:57 +0200 > From: [EMAIL PROTECTED] > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Windows Mobile 6 IAX/SIP client? > > I just bought a HTC TyTn II phone, but unfortunately it doesn't even have > a SIP client in it. > > I tried the wiki searching for a SIP or IAX client but only found some > PocketPC stuff (Windows Mobile 2003). > > Does anyone know of a good quality SIP or IAX softphone that will run on > Windows Mobile 6? > > I only have a data subscription, no voice so the quality should be > sufficient to be used constantly. > > Thanks!! > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to setup one stage dialing plan, instead of two! help!!!
Hello all, i recently finished setting up my asterisk with sipura 3102 using PSTN. this is my dial plan relevant to wht i want: exten =>_01,1,Dial(SIP/$(EXTEN)@200) right now as u see i made my dial plan on a 2 stage dialing mode. tht means i dial 01, i get the pstn dial tone, and then i call whichever number i want through it. i want to have the option for my call to directly go through pstn without having to wait for the pstn dial tone. any help would be appreciated.. :) _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming calls through callcentric sip account!!
Hi all, i've recently acquired a callcentric account. i've perfectly setup my sip.conf and extensions.conf to make outgoing calls. but the problem is with incoming calls! when i call in, asterisk doesnt even see the incoming call! how is tht possible! please see the following my config: sip.conf: [general] dtmfmode = rfc2833 context=from-callcentric srvlookup=yes register => username:[EMAIL PROTECTED]/username [callcentric] type=peer context=from-callcentric host=callcentric.com username=username secret=password fromuser=username fromdomain=callcentric.com insecure=very [107] context=to-callcentric type=friend username=107 secret=secret host=dynamic etensions.conf: [from-callcentric] exten => s,1,Dial(SIP/107) [to-callcentric] exten => _0.,1,Dial(SIP/[EMAIL PROTECTED]) _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extensions help!
hello all,
was wondering if some1 could help me to add an option to my incoming operator
menu.
currently, when some1 calls in, he gets a recorded msg asking for him to punch
in an extension or dial 100 for operator assistance wht i want is to add 2
other things;
firstly, if in a period of time the person didnt punch in an extension i want
him to b directed atomaticly to the operator.
2ndly, to add an option of lets say, press 2 to listen to availabe extensions
this is my extensions.conf
[sipura-line]
exten => 201,1,Answer() ; Answer inbound calls
exten => 201,2,Playback(silence/1)
exten => 201,3,Background(simzy1) ; input an extension
exten => 201,4,Wait(8)
include => spa
exten => 201,n,Hangup()
[spa]
exten =>_201,1,GoTo(sipura-line,${EXTEN},1)
exten => _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 t\
imes
exten => _1XX,2,VoiceMail([EMAIL PROTECTED])
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 t\
imes
exten => _2XX,2,VoiceMail([EMAIL PROTECTED])
exten => _2XX,3,HangUp()
exten =>_01,1,Dial(SIP/200)
exten => 203,1,VoicemailMain
exten => _2XX,1,Dial(SIP/${EXTEN},15)
_
Discover the new Windows Vista
http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE___
-- 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] extensions.conf HELP with dial plan!!
hello all,
im looking for a way to do the following:
when a SPECIFIC call comes through to asterisk through sip, i want it to b
directed to a pool of specific sip extensions (9 extensions) where asterisk
tries one after the other till lhe finds one of them thats actually on.i want
to add a step for asterisk to follow which is, when a sip extension doesn't
answer or its offline, instead of immediately transferring to voice mail, i
want it to dial that sip holder's number so it transfers the call to his
cellphone for example. and if he didn't answer his cellphone its then that i
want it to direct it to voice mail.i want to add another item to the operator
menu, instead of just receiving the call and telling the caller to either dial
extension or 100 for operator, i want asterisk to offer the caller an
additional option like for example pressing 2, would direct you to a list of
key personnels with their respective extensions.please find below my
extensions.conf:
[sipura-line]
exten => 201,1,Answer() ; Answer inbound calls
exten => 201,2,Playback(silence/1)
exten => 201,3,Background(simzy1) ; input an extension
exten => 201,4,Wait(8)
include => spa
exten => 201,n,Hangup()
[spa]
exten =>_201,1,GoTo(sipura-line,${EXTEN},1)
exten => _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten => _1XX,2,VoiceMail([EMAIL PROTECTED])
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten => _2XX,2,VoiceMail([EMAIL PROTECTED])
exten => _2XX,3,HangUp()
exten =>_01,1,Dial(SIP/200)
exten => 203,1,VoicemailMain
exten => _2XX,1,Dial(SIP/${EXTEN},15)
_
News, entertainment and everything you care about at Live.com. Get it now!
http://www.live.com/getstarted.aspx___
-- 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] adding funcionatlity to asterisk?! is it possible?!
hello all,
im looking for a way to do the following:
when a SPECIFIC call comes through to asterisk through sip, i want it to b
directed to a pool of specific sip extensions (9 extensions) where asterisk
tries one after the other till lhe finds one of them thats actually on.i want
to add a step for asterisk to follow which is, when a sip extension doesn't
answer or its offline, instead of immediately transferring to voice mail, i
want it to dial that sip holder's number so it transfers the call to his
cellphone for example. and if he didn't answer his cellphone its then that i
want it to direct it to voice mail.i want to add another item to the operator
menu, instead of just receiving the call and telling the caller to either dial
extension or 100 for operator, i want asterisk to offer the caller an
additional option like for example pressing 2, would direct you to a list of
key personnels with their respective extensions.please find below my
extensions.conf:
[sipura-line]
exten => 201,1,Answer() ; Answer inbound calls
exten => 201,2,Playback(silence/1)
exten => 201,3,Background(simzy1) ; input an extension
exten => 201,4,Wait(8)
include => spa
exten => 201,n,Hangup()
[spa]
exten =>_201,1,GoTo(sipura-line,${EXTEN},1)
exten => _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten => _1XX,2,VoiceMail([EMAIL PROTECTED])
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten => _2XX,2,VoiceMail([EMAIL PROTECTED])
exten => _2XX,3,HangUp()
exten =>_01,1,Dial(SIP/200)
exten => 203,1,VoicemailMain
exten => _2XX,1,Dial(SIP/${EXTEN},15)
_
Invite your mail contacts to join your friends list with Windows Live Spaces.
It's easy!
http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us___
-- 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] Incoming calls not being answered by asterisk
Hey thanks for the help :) though i already did that, and the sip debugging info shows me tht its ringing on the respective sip extension (1002) but nothing is happening.. so i guess its true it IS a diala plan issue tht i am yet to figure it out ... > Date: Sat, 24 May 2008 14:20:45 +0100 > From: [EMAIL PROTECTED] > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Incoming calls not being answered by asterisk > > The first thing to do is type "sip debug" on the console and place the > call from the Sipura. If you get a bunch of SIP messages flashing down > your console you know the call is reaching Asterisk and it's most > likely going to be an issue authenticating the call or a problem in > your dial plan. > > If no SIP messages flash up then the call is not reaching your Asterisk > server. > > Regards, > > Greyman. > > ___ > -- 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 _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- 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] Incoming calls not being answered by asterisk
Hello all, ive got the following setup currently: __Sipura 3102-PSTN | Lan | | |__asterisk i configured both asterisk and pstn to be able to receive/make calls through each other using sip of course.. but the thing is i want asterisk that when it receives an incoming call from sipura, to answer it, play msg that i recorded and wait for the caller to dial in an extension, where it would transfer the caller to that exntension, and in case the extension owner isnt available to answer it would direct him to his voicemail(tht i dont know how to set yet), and in case the caller didnt dial any extension in a certain amount of time, it automaticly directs it to a specific extensions i'd specify.. i tried the examples given in lots of forums and so on none of them worked, the phone keeps on ringing with every incomign dial plan ive specified without asterisk answering it.. the thing i did is that sipura directs incoming calls to 1002, so ive set the context of 1002 in sip.conf to a dial plan of [incoming-sipura] and ive set the commands i mentioned earlier tht i took out of those forums.. but theyre not working!!! anyone has an example i could go on with ? any help would be apreciated:) _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE___ -- 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] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hi Roberto,
i added this exntesions.conf
[spa]
Exten => _1XXX,1,Dial(SIP/${EXTEN})
exten => _0.,1,Dial(SIP/1009/${EXTEN:1})
and in sip.conf:
[1009]
username=1009
type=friend
secret=1234
host=dynamic
canreinvite=yes
context=spa
disallow=all
allow=alaw
dtmfmode=info
qualify=yes
callgroup=1
pickupgroup=1
which i have the extension 1009 in sip.conf directed to it..
and then tried calling out but it still gave me the same error!
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f0b840",
"SIP/1009/0144") in new stack
-- Called 1009/0144
-- Got SIP response 503 "Service Unavailable" back from 192.168.0.111
-- SIP/1009-0821d888 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/1003-b5f0b840' status is 'CONGESTION'
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Wed, 21 May 2008 10:29:21 -0700
Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to
pstn calls)
Ciao Roland
your dialplan:Exten => _1XX,1,Dial(SIP/${EXTEN})
_1XX is a three (3) digit number starting with 1, I'm not sure what happens if
you dial 1009 but it seems that it is dialing.
Anyway the ${EXTEN} is 1009 so asterisk is trying to dial that extension which
doesn't exist.
your dial out should look something like:
[outgoing]
exten => _9.,1,Dial(SIP/100/${EXTEN:1})
where you're specifying that all the calls that starts with 9 will go to
extension 100 (assuming that is your spa-3102) and there you dial the number
dialed from the caller stripped by the 9 (that is the :1 after EXTEN)Now 9 is
standard in USA for outside line, in some other countries is 0, you choose
CiaoRoberto
On May 21, 2008, at 7:52 AM, RoLaNd RoLaNd wrote:
Hello Roberto,
first of all, id like to thank you for your help with this..
secondly, i tried the configuration you gave me but it still gave me the same
error..!
but just to b sure ill tell u wht im doing..
after following ur advice to the letter.. i kept my asterisk configuration the
same the only thing i edited in sip.conf is adding the port for the pstn
extension to match the one in sipura 3102.. and gave the PSTN line interface on
sipura the user id of " 1009"
then i called from my softphone 1009 so i could dial out..
and it gave me this error in asterisk cli:
Connect attempt from '127.0.0.1' unable to authenticate
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f0e828", "SIP/1009") in
new stack
-- Called 1009
-- Got SIP response 503 "Service Unavailable" back from 192.168.0.111
-- SIP/1009-0821d888 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/1003-b5f0e828' status is 'CONGESTION'
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager.d/op-panel.conf': Found
== Parsing '/etc/asterisk/users.conf': Found
is that the right way of doing this?! do i call 1009 (pstn line user id) or
wht!
ps: could us hare with me ur sip.conf and extensions.conf please just to
compare mine with urs maybe something is missing!
once again thanks for ur help :)
> Message: 22
> Date: Wed, 21 May 2008 06:49:39 -0700
> From: Roberto Milani <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to
> sip/sip to pstn calls)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="windows-1252"
>
> Hi Roland
>
> I have 2 linksys spa-3102 working pretty good both dialing in and out
> and I followed this instructions to set it up:
>
>
> update to the latest firmware then:
>
> ..Go to the first tab ?Voice? and sixth sub-tab ?Line 1?
> SIP Settings:
> ..SIP Port: Notice that it is set to 5060 for line 1 and 5061 for
> PSTN Line (next tab). These port values must be correctly transferred
> to the correct contexts in sip.conf.
> Proxy and registration:
> ..Proxy: 192.168.5.70 < The IP address of your Asterisk server
> Subscriber Information:
> ..Display Name: LivingRoom < This will be the test phone, but any
> name would do as lone as it is used in the configuration files.
> ..User ID: LivingRoom
> ..Password: SomePassword
> ..Auth ID: LivingRoom < probably not needed
> Dial Plan:
> ..Dial Plan: (*xx|[3469]11|0|00|[2-9]x|
> 1xxx[2-9]xS0|.) < We have 10 digit local dialing.
> The default is set for seven digit local dialing. Adjust as needed.
> ..Emergency Number: < Hmmm, I don?t know what to do here: it?s
> probably important, but it is poor form to dial 911 just
Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hi Jose, i just did that, doesnt seem to work.. its still giving me the same error Date: Wed, 21 May 2008 11:02:36 -0500 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls) I was seeing your print screen images, and the observation is. You are not doing any sip registration on the server since your Register option in the Tab PSTN Line is set to NO. you should change it to yes. (or add in the sip.conf the host=SPA_ip instead of dynamic). regards. -- Jose Flores Galicia <<[EMAIL PROTECTED]>> BriefCode && Code Based Training _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us___ -- 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] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
yes thats the only thing i have in extensions.conf
should there be anything else?!
Message: 21Date: Wed, 21 May 2008 09:40:26 -0400From: Matt Watson <[EMAIL
PROTECTED]>Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to
sip/sip to pstn calls)To: Asterisk Users Mailing List - Non-Commercial
Discussion Message-ID:<[EMAIL
PROTECTED]>Content-Type: text/plain; charset="us-ascii" Does your
extensions.conf have any more configuration than what you've shown? If not,
then you are lacking dialplan for anything but internal calls. --Matt From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNdSent:
Wednesday, May 21, 2008 9:01 AMTo: [EMAIL PROTECTED]: [asterisk-users] asterisk
and sipura 3102 (pstn to sip/sip to pstn calls) Hello all, its been a while im
trying to setup my asterisk/sipura 3102 to recieve/make calls from softphones
on pcs in my home..i've set up 5 SIP extensions in sip.conf and made the
dialing plan in extensions.conf..i could make calls from 1 sip phone to another
in my home.. but i cant call out using pstn line interface nor recieve
calls..please find below my topology as well as config info:
(192.168.0.0) LAN__ |
| |softphone asterisk
sipura-PSTN LINE Configuration: ASTERISK: sip.conf
[101]type=peerport=5062host=dynamicsecret=1234context=spa
[103]type=peerport=5061host=dynamicsecret=1234context=spa
[100]type=peerport=5061host=dynamicsecret=1234context=spa
[111]type=peerport=5060host=dynamicsecret=1234context=spa
== === EXTENSIONS.CONF
[spa]Exten => _1XX,1,Dial(SIP/${EXTEN})
== === and this is the
settings i have right now for sipura 3102 in my PSTN LINE:
http://img84.imageshack.us/my.php?image=40541922um2.jpg
http://img98.imageshack.us/my.php?image=55448347ss9.jpg
http://img262.imageshack.us/my.php?imag ...
472qz3.jpg ps: i
read so many tutorials and none seems to help..lately whenever i try to call
out using my sipphone.. it gives me "503 service unavailable" and this is wht
shows on the CLI of asterisk when i set sip debug on..
ubuntu-pbx-desktop*CLI> == Connect attempt from '127.0.0.1' unable to
authenticate-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f05600",
"SIP/1009") in new stack-- Called 1009*CLI>-- Got SIP response 410
"Gone" back from 192.168.0.111-- SIP/1009-081741d0 is circuit-busy ==
Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel
'SIP/1003-b5f05600' status is 'CONGESTION'
_
Invite your mail contacts to join your friends list with Windows Live Spaces.
It's easy!
http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us___
-- 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] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
k hands
> out.> PSTN-To-VoIP Gateway Setup:> ..PSTN Ring Thru Line 1: no < When
> this is ?yes?, an incoming call > goes directly through to Line 1. We only
> want line 1 to ring when > Asterisk routs a call to it.> ..PSTN CID for
> VoIP CID: yes < capture the Caller ID provided by > the incoming call and
> pass it through to Asterisk to display on your > internal phones.> ..PSTN
> Caller Default DP: 2 < Change to 2. The incoming call will > be passed to
> your extensions.conf file with extension 's' as defined > in Dial Plan 2
> (above).> ..Off Hook While Calling VoIP: no < I read this in some Google
> > search. I don?t know what it does, but stuff seems to work. Help?> FXO
> Timer Values (sec):> ..PSTN Answer Delay: 5 < Delay so that you can get
> the CID data. > NghtShd at
> http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html
> > claims that 5 seconds is long enough.> Click Submit All Changes> >
> Ciao> > Roberto> > On May 21, 2008, at 6:00 AM, RoLaNd RoLaNd wrote:> > >
> Hello all,> >> > its been a while im trying to setup my asterisk/sipura 3102
> to > > recieve/make calls from softphones on pcs in my home..> > i've set up
> 5 SIP extensions in sip.conf and made the dialing plan > > in
> extensions.conf..> > i could make calls from 1 sip phone to another in my
> home.. but i > > cant call out using pstn line interface nor recieve calls..>
> > please find below my topology as well as config info:> >> > (192.168.0.0)>
> > LAN__> > | | |> > softphone asterisk
> sipura-PSTN LINE> >> >> >> > Configuration:> >> > ASTERISK:> >> >
> sip.conf> >> > [101]> > type=peer> > port=5062> > host=dynamic> >
> secret=1234> > context=spa> >> >> > [103]> > type=peer> > port=5061> >
> host=dynamic> > secret=1234> > context=spa> >> > [100]> > type=peer> >
> port=5061> > host=dynamic> > secret=1234> > context=spa> >> > [111]> >
> type=peer> > port=5060> > host=dynamic> > secret=1234> > context=spa> >> >
> == ===> >> >
> EXTENSIONS.CONF> >> > [spa]> > Exten => _1XX,1,Dial(SIP/${EXTEN})> >> >
> == ===> >> >> > and
> this is the settings i have right now for sipura 3102 in my PSTN > > LINE:>
> >> >> > http://img84.imageshack.us/my.php?image=40541922um2.jpg> >> >
> http://img98.imageshack.us/my.php?image=55448347ss9.jpg> >> >
> http://img262.imageshack.us/my.php?imag ... 472qz3.jpg> >> > ps: i read so
> many tutorials and none seems to help..> > lately whenever i try to call out
> using my sipphone.. it gives me > > "503 service unavailable" and this is wht
> shows on the CLI of > > asterisk when i set sip debug on..> >> >> >> >> >
> ubuntu-pbx-desktop*CLI>> > == Connect attempt from '127.0.0.1' unable to
> authenticate> > -- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f05600",
> "SIP/1009") > > in new stack> > -- Called 1009*CLI>> > -- Got SIP response
> 410 "Gone" back from 192.168.0.111> > -- SIP/1009-081741d0 is circuit-busy> >
> == Everyone is busy/congested at this time (1:0/1/0)> > == Auto fallthrough,
> channel 'SIP/1003-b5f05600' status is > > 'CONGESTION'> >> >> >> > Invite
> your mail contacts to join your friends list with Windows > > Live Spaces.
> It's easy! Try it! > > ___> > --
> 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> > --
> next part --> An HTML attachment was scrubbed...> URL:
> http://lists.digium.com/pipermail/asterisk-users/attachments/20080521/7c9ef721/attachment.htm
> > > --> >
> ___> --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> > End of
> asterisk-users Digest, Vol 46, Issue 69>
> **
_
News, entertainment and everything you care about at Live.com. Get it now!
http://www.live.com/getstarted.aspx___
-- 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] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
Hello all,
its been a while im trying to setup my asterisk/sipura 3102 to recieve/make
calls from softphones on pcs in my home..
i've set up 5 SIP extensions in sip.conf and made the dialing plan in
extensions.conf..
i could make calls from 1 sip phone to another in my home.. but i cant call out
using pstn line interface nor recieve calls..
please find below my topology as well as config info:
(192.168.0.0)
LAN__
|| |
softphone asterisk sipura-PSTN LINE
Configuration:
ASTERISK: sip.conf [101] type=peer port=5062 host=dynamic secret=1234
context=spa [103] type=peer port=5061 host=dynamic secret=1234 context=spa
[100] type=peer port=5061 host=dynamic secret=1234 context=spa [111] type=peer
port=5060 host=dynamic secret=1234 context=spa
== === EXTENSIONS.CONF
[spa] Exten => _1XX,1,Dial(SIP/${EXTEN})
== === and this is the
settings i have right now for sipura 3102 in my PSTN LINE:
http://img84.imageshack.us/my.php?image=40541922um2.jpg
http://img98.imageshack.us/my.php?image=55448347ss9.jpg
http://img262.imageshack.us/my.php?imag ... 472qz3.jpg
ps: i read so many tutorials and none seems to help..
lately whenever i try to call out using my sipphone.. it gives me "503 service
unavailable" and this is wht shows on the CLI of asterisk when i set sip debug
on..
ubuntu-pbx-desktop*CLI> == Connect attempt from '127.0.0.1' unable to
authenticate-- Executing [EMAIL PROTECTED]:1] Dial("SIP/1003-b5f05600",
"SIP/1009") in new stack-- Called 1009*CLI>-- Got SIP response 410
"Gone" back from 192.168.0.111-- SIP/1009-081741d0 is circuit-busy ==
Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel
'SIP/1003-b5f05600' status is 'CONGESTION'
_
Invite your mail contacts to join your friends list with Windows Live Spaces.
It's easy!
http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us___
-- 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
