RE: [Asterisk-Users] Linux Firewall Question
> > Hi, > > I am running Asterisks on Public IP with Fedora Core 3. > > > > What is the recommendation for making Linux secure on the > > Public IP since I am new to Linux. Which Firewall should I > > use? I am not intending to use Linux as router. > > > > Can any one provide some configuration documentation. > > I use shorewall, and I have found it powerful, and fairly easy to use. > > http://www.shorewall.net/ Shorewall is an excellant iptables based firewall. You can create zones and policy's to govern them. If you are also looking for easy GUI to configure it, then best way to do it is using WEBMIN http://www.webmin.com Its a free web based interface for system management. It includes Module for shorewall confuguration. -ask ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Continuously ringing Zap/4-1 TDM11B All of a sudden ?[Urgent Pls]
Dear All,
All these days I was ahpppily using Asterisk with TDM11B, but from
today all of a sudden asterisk has started acting strange.
The telephone device connected to channel 1 rings continously,
following info is displayed on console
-- Starting simple switch on 'Zap/4-1'
Jan 28 15:38:58 ERROR[77024176]: callerid.c:261 callerid_feed: fsk_serie
made mylen < 0 (-8)
Jan 28 15:38:58 WARNING[77024176]: chan_zap.c:5414 ss_thread: CallerID
feed failed: Success
Jan 28 15:38:58 WARNING[77024176]: chan_zap.c:5456 ss_thread: CallerID
returned with error on channel 'Zap/4-1'
-- Executing Answer("Zap/4-1", "") in new stack
Please tell me what must have gone wrong so unnoticed ?
Its very urgent to put it back to work, as i dont have backup plans.
I can send my config files if u wish to look at it.
Eagerly awiating.
Thanks in advance.
Anand
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[Asterisk-Users] 12 CANCEL's followed by 12 INVITE's in 5 secs
Hello All,
I have a problem that is alien to me and obvious for some of you
:). I have asterisk setup with few sip clients(using linphonec).
In a proper context, I have mentioned extensions 107 as
[EMAIL PROTECTED] (x.x.x.x=asterisk server ip)
Asterisk Sever-simputer(sip ua)
I can make calls from sipua to asterisk but not reverse way.
I get the following display on asterisk terminal.
Could anyone of you please tell em why im not able to receive calls on
linphonec ?n
-
*CLI>
-- Executing Dial("SIP/clienta-30c9", "SIP/simputer|20|tr") in new
stack
-- Called simputer
Dec 28 12:00:05 WARNING[-1116775504]: chan_sip.c:665 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
== No one is available to answer at this time
Dec 28 12:00:11 WARNING[-1116775504]: chan_sip.c:665 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Non-critical Request)
Dec 28 12:00:15 WARNING[36199344]: pbx.c:1996 ast_pbx_run: Timeout, but no
rule 't' in context 'sip'
WHERE X.X.X.X=ASTERISK SEREVR
---
I do tcpdump and see 12 INVITES going from asterisk to sipua,and then 12
CANCEL in a span of 5 secs and the session terminates with being call
established.
Please tell me what could be the problem here ?
*---
for you anaysis: i have put my extensions.conf here..
[globals]
;AbsoluteTimeout(3)
[incoming]
;incoming context is tied with channel 4 FXO device
exten => s,1,Answer
exten => s,2,Background(beep)
exten => s,3,Dial(Zap/1,40,tr)
exten => s,4,Playback(vm-isunavail)
;exten => s,5,Dial(SIP/clienta,10,tr)
exten => s,5,Background(vm-enter-num-to-call)
exten => s,6,NoOp,${CALLERID}
;exten => s,8,Dial(Zap/1,20,tr)
;exten =>s,9,Hangup
include =>sip
;sip users
[sipextensions]
exten => 100,1,Dial(SIP/clienta,20,tr)
exten => 101,1,Dial(SIP/salisa,20,tr)
exten => 102,1,Dial(SIP/salisd,20,tr)
exten => 103,1,Dial(SIP/sourabha,20,tr)
exten => 104,1,Dial(SIP/laptop,20,tr)
exten => 105,1,Dial(SIP/anurag,20,tr)
exten => 106,1,Dial(SIP/askatti,20,tr)
exten => 107,1,Dial(SIP/simputer,20,tr)
exten => 108,1,Dial(SIP/geetha,20,tr)
include=> record
;working
[extensions]
exten => _3X,1,Dial,Zap/4/${EXTEN}
exten => _4X,1,Dial,Zap/4/${EXTEN}
include =>sipextensions
;working
[centrix]
ignorepat => 9
exten => 9,1,Dial,Zap/4/${EXTEN}
exten => _93XXX,2,Dial,Zap/4/${EXTEN:1}
include =>extensions
;working
[local]
ignorepat =>0
exten => 0,1,Dial,Zap/4/${EXTEN}
exten => _0NX,2,Dial(Zap/4/${EXTEN:1})
exten => _0NXXX,2,Dial(Zap/4/${EXTEN:1})
include =>centrix
;working
[longdistance]
ignorepat => 0
exten => 25,1,Dial,Zap/4/${EXTEN}
exten => _250NXNXXX,2,Dial,Zap/4/${EXTEN:2}
include =>local
;FOR SIP
[sip]
;exten => s,1,Wait(1)
exten => 1000,1,Dial(Zap/1,20,t)
exten => 1000,2,Hangup
include =>sipextensions
include=>local
--sip.conf
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = X.X.X.X; Address to bind to
context = sip ; Default for incoming calls
;context = longdistance
disallow=all
allow=ulaw
alloq=gsm
allow=alaw
allow=iLbc
maxexpirey=180
canreinvite=yes
nat=no
defaultexpirey=160
[askatti]
type=friend
secret=oneday
host=dynamic
username=askatti
[simputer]
type=friend
username=simputer
host=dynamic
Warm Regards,
Anand
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[Asterisk-Users] Call Placing timeouts
Hello All,
I have a problem that is alien to me and obvious for some of you
:). I have asterisk setup with few sip clients.
In a proper context, I have mentioned extensions 107 as
[EMAIL PROTECTED]
Asterisk Server-simputer(sip ua)
I can make calls from sipua to asterisk but not reverse way.
I get the following display on asterisk terminal
-
*CLI>
-- Executing Dial("SIP/clienta-30c9", "SIP/simputer|20|tr") in new
stack
-- Called simputer
Dec 28 12:00:05 WARNING[-1116775504]: chan_sip.c:665 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
== No one is available to answer at this time
Dec 28 12:00:11 WARNING[-1116775504]: chan_sip.c:665 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Non-critical Request)
Dec 28 12:00:15 WARNING[36199344]: pbx.c:1996 ast_pbx_run: Timeout, but no
rule 't' in context 'sip'
---
I do tcpdump and see 12 INVITES going from asterisk to sipua,and then 12
CANCEL in a span of 5 secs and the session terminates with being call
established.
Please tell me what could be the problem here ?
*---
for you anaysis: i have put my extensions.conf here..
[globals]
;AbsoluteTimeout(3)
[incoming]
;incoming context is tied with channel 4 FXO device
exten => s,1,Answer
exten => s,2,Background(beep)
exten => s,3,Dial(Zap/1,40,tr)
exten => s,4,Playback(vm-isunavail)
;exten => s,5,Dial(SIP/clienta,10,tr)
exten => s,5,Background(vm-enter-num-to-call)
exten => s,6,NoOp,${CALLERID}
;exten => s,8,Dial(Zap/1,20,tr)
;exten =>s,9,Hangup
include =>sip
;This context is used to record voicemenu and its recording properly.
; used to record prompts
[playback]
exten =>20,1,Playback(vm-sorry)
exten =>20,2,Hangup
[record]
exten => 205,1,Wait(2)
exten => 205,2,Record(/tmp/asterisk-recording1:gsm)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording1)
exten => 205,5,Wait(2)
exten => 205,6,Hangup
include =>playback
;sip users
[sipextensions]
exten => 100,1,Dial(SIP/clienta,20,tr)
exten => 101,1,Dial(SIP/salisa,20,tr)
exten => 102,1,Dial(SIP/salisd,20,tr)
exten => 103,1,Dial(SIP/sourabha,20,tr)
exten => 104,1,Dial(SIP/laptop,20,tr)
exten => 105,1,Dial(SIP/anurag,20,tr)
exten => 106,1,Dial(SIP/askatti,20,tr)
exten => 107,1,Dial(SIP/simputer,20,tr)
exten => 108,1,Dial(SIP/geetha,20,tr)
;exten => clienta,1,Dial(SIP/clienta,20,tr)
include=> record
;working
[extensions]
exten => _3X,1,Dial,Zap/4/${EXTEN}
exten => _4X,1,Dial,Zap/4/${EXTEN}
include =>sipextensions
;working
[centrix]
ignorepat => 9
exten => 9,1,Dial,Zap/4/${EXTEN}
exten => _93XXX,2,Dial,Zap/4/${EXTEN:1}
include =>extensions
;working
[local]
ignorepat =>0
exten => 0,1,Dial,Zap/4/${EXTEN}
exten => _0NX,2,Dial(Zap/4/${EXTEN:1})
exten => _0NXXX,2,Dial(Zap/4/${EXTEN:1})
include =>centrix
;working
[longdistance]
ignorepat => 0
exten => 25,1,Dial,Zap/4/${EXTEN}
exten => _250NXNXXX,2,Dial,Zap/4/${EXTEN:2}
include =>local
;FOR SIP
[sip]
;exten => s,1,Wait(1)
exten => 1000,1,Dial(Zap/1,20,t)
exten => 1000,2,Hangup
include =>sipextensions
include=>local
--sip.conf
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = X.X.X.X; Address to bind to
context = sip ; Default for incoming calls
;context = longdistance
disallow=all
allow=ulaw
alloq=gsm
allow=alaw
allow=iLbc
maxexpirey=180
canreinvite=yes
nat=no
defaultexpirey=160
[askatti]
type=friend
secret=oneday
host=dynamic
username=askatti
[simputer]
type=friend
username=simputer
host=dynamic
Warm Regards,
Anand
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[Asterisk-Users] Asterisk behind external PBX +enable IVR
Hello List,
I have 3 questions. (I have TDM 11B)
1. My connection is like this
PSTN
|
|
SOME PBX
/
/
ASTERISK PBX
| \
|\
SIP UA1 \
SIP UA2
As you see, from aserisk box, if want to make outgoing calls, i have to
dial '0' and wait for the dialtone, and then dial the destination number.
for this to achieve i have following in extension.conf
exten=> _0._,Dial,Zap/4/${EXTEN}
And this works.
But what is the better way to do this ?, like i want to dial 0 and wait
for the dialtone if dialtoneis received i will dial 8 or 10 digit
destination number. I want to have something like this
exten=> _0.NXXX,Zap/4/${EXTEN}; for 8 digit desti..no.
exten=> _0.NXX,Zap/4/${EXTEN};FOR 10 DIGIT desti..no.
So what is the proper way to write the above dialplan ?
Similarly, for some calls, i have to dial 9 and wait for dialtone from the
external PBX, and then dial the 4 digit destination number, but im not
able to write the dialplan for this.
Could someone shed light on this ?
2. How do i enable IVR on my asterisk machine ? I want to have extensions
like 1001 for SIPUA1 and 1002 for SIPUA2. So when a call comes from
outside then MY IVR should pick the phone and ask the caller to dial an
extension 1001 top talk to SIPUA1. When caller dials 1001, then the call
should be forwarded to SIPUA1.
Could someone tell how to do this methodiacaly...
3.I want to call from SIPUA1 to SIPUA2.
How the extension, or sip.conf gets modified for this ?
Thanks in advance for reading the mail till this end.
awaiting for the inputs.
Regards,
Anand
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Re: [Asterisk-Users] sip.conf extensions.conf
Am too in same situation...but inmy case even 1,2,3 are not working. Can you send you .conf files ? thanks, anand On Fri, 5 Nov 2004, Mauro Locatelli wrote: > I have an asterisk server(x100p wildcard) that function as a gateway. > I have some local soft phone (for example 3) and I want: > > 1- call from one internal softphone the other internal softphone > 2- call out on the PSTN from internal softphone > 3- call out on the sipphone.com > 4- receive call from external PSTN and choose wich internal softphone ring > 5- receive call from sipphone and choose wich internal softphone ring > > I make sip.conf and extensions.conf but only 1,2,3 point work.. > > If someone is in my situation, and work all full, can send me his sip.conf and > extensions.conf for compare? > > Very very thanks and sorry for english.. > Mauro ><>___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk as sip proxy registrar
Hello ALl,
I have Asterisk up and running. Now I want to set it up as sip
proxy registrar. I have few machines with xlite and linphone sip UA's.
How do i register these UA's in asterisk ?
My Asterisk server IP is 10.0.0.2
and my sip UA's IP addresses are 10.0.0.3,10.0.0.4,and 10.0.0.5
When my sip ua's sends reigstration request to Asterisk server, On the
Asterisk console it displays error NOTICE saying
GOT SIP RESPONSE 403 "Forbidden" back from 10.0.0.2
Please let me know if there are any documentation on setting up soft
phones like linphone/xlite with asterisk.
Thanks in advance,
Anand
My sip.conf file looks like this..
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 10.0.0.2; Address to bind to
context = sip ; Default for incoming calls
srvlookups=yes
dtmfmode=inband
allow=all
tos = lowdelay
register => [EMAIL PROTECTED]/1000
register => [EMAIL PROTECTED]/1001
register => [EMAIL PROTECTED]/1002
[sourabha]
type=friend
secret=oneday
host=dynamic
defaultip=10.0.0.3
username=1001
[clienta]
type=friend
secret=oneday
host=dynamic
defaultip=10.0.0.4
username=1000
[salisd]
type=friend
;insecure=yes
username=1002
secret=oneday
host=dynamic
defaultip=10.0.0.4
---
$extensions.conf file
[globals]
clienta=SIP/1000
sourabha=SIP/1001
salisd=SIP/1002
[incoming]
exten => s,1,Dial,Zap/1,3
exten => s,2,Playback(vm-nobodyavail)
[sip]
;calls to SIP UAs
exten => 1000,1,Dial(${CLIENTA})
exten => 1001,1,Dial(${SOURABHA})
exten => 1002,1,Dial(${SALISD})
---
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RE: [Asterisk-Users] Dialogic supported well?
> Good luck - call the guys at Digium (Malcolm or Greg) - they are very > helpful Can I have email ID and phone number of Malcolm or Greg ? Im from INDIA, unfortunately DIGIUM doesn't have any resellers here. So I think i've to call up Australia. Its good that now I know two names there. > > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Alfred Werner > Sent: Friday, March 05, 2004 5:05 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Dialogic supported well? > > > I'm new to asterisk and quite impressed by the feature list. I have a D/4PCI > already in hand. Is there any reason NOT to use this and buy a digium card > instead? > > I basically want to set up a couple line analog system to check it out and > probably use as a a Soho setup for VM, access to a postgres database, and to > play with the VOIP stuff. > > TIA, > > Alfred Werner > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ALSA Sound dies after a while
One of the Possibilities could be that there is no disk space available, or permissions. Do you get same error if you are root ? > > I am currently running an asterisk server locally on a PC, to see what > kinds of things it can do. The output device is the sound system, via > alsa. It starts perfectly and plays the first sounds but after a minute > or to the sound dies and several messages > Mar 3 21:30:29 WARNING[180236]: chan_oss.c:272 sound_thread: Failed to > write sound > Mar 3 21:30:29 WARNING[180236]: chan_oss.c:181 send_sound: Unable to > read output space > > are issued. > This looks like an alsa lock. I have tried to run the system both with > and without esd. When running lsof for /dev/dsp only asterik and it's > child processes are using it. > > Has anybody experienced this? > > thanks in advance > Juan Pablo Morales > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: sip router gateway
Dear Girish, Thank you for the comments, my responses are inline. > >My question: > > > >CAN I use asterisk software as a replacement for siprg ? CAN I use DIGIUMs > >x100p PCI card as a replacement for QUICKNETS CTI LINE JACK CARD ? > > Asterisk is a soft PBX that has a SIP channel. It doesnt perform all the > functionalities of a sip router. You can use digium's cards with asterisk, > but is it legal to use such cards here in INDIA? As far as i know BSNL > relegations dont permit to use their infracture to use with VoIP. They dont > allow to land VoIP calls to a PSTN phone/mobile. (But it is possible, > illegal though) Yes, since we are doing this as part of the educational programme, and more over we plan to take special permission from BSNL towards this. Could you please explain me how does this soft PBX works ? and I want all the functionalities of the sip. Since ASTERISK does support SIP also. I think it should be ok to use linphone,gnuosip,partysip. because gnuosip provides a set of API's which these software will use. Asterisk may not use that since it already has sip stack. But i think we should take care of ports > > >CAN ASTERISK work as SIP ROUTER GATEWAY ? > > Asterisk is a PBX, If you just want a SIP router, i think it is better to > use SIP Express Router or anything like that. but sip express router, does it supoort linphone,gnuosip,partysip? and what linejack card should i be using for this ? > > >CAN I USE ASTERISK, DIGIUMs card along with LINPHONE, PARTYSIP and > >GNUoSIP. ? > > Partysip and Linphone uses oSIP. right? Asterisk does not use any external Yes > protocol stacks. It has its own implementation. I am not sure, but are you > planning to use * and Partysip in a same machine? What about SIP ports? I > could be wrong. Yes, want to use it on the same machine,as you said, i really dont know whether the asterisk sip stack and gnuosip will collide..definetly. > Regards, Girish > > _ > INDIA TODAY @ Rs. 5 for 5 years ! > http://www.indiatoday.com/itoday/intlsubscription/itsubs/it_offer.html > Subcribe Now ... > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip router gateway
Hello All, This is my first post to this list, I hope i'll get my problem solved here. I was using following set of software for my network within the LAN. linphone[as SIP UA],partysip[SIP PROXY SERVER],GNUosip[SIP STACK]. with ALSA SOUND DRIVERS. Then I wanted to make IP-PSTN and PSTN-IP calls, so that i thought of using siprg[a sip router gateway] using QUICKNET CTI LINE JACK CARD. But unfortunately the card is available for ISA slots only. Where as I needed PCI card. Then after reading LINUX JOURNAL [ FEB edition 2004], i came across a nice article by BRETT SCHWARZ on ASTERISK. My question: CAN I use asterisk software as a replacement for siprg ? CAN I use DIGIUMs x100p PCI card as a replacement for QUICKNETS CTI LINE JACK CARD ? CAN ASTERISK work as SIP ROUTER GATEWAY ? CAN I USE ASTERISK, DIGIUMs card along with LINPHONE, PARTYSIP and GNUoSIP. ? Kindly put light on this. Regards, Anand ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
