Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?
2009/8/4 Faheem faheem_...@yahoo.com
how to implement CLONED LINE Feature in asterisk
Hey, I want to implement Clone Line feature in asterisk. I am using
SPA-2100.
The feature should work in this way.
There are two ports in the SPA-2100 both are registered with asterisk with
same username/password, and have the same (phone number)
*No one on the phone *
*One phone in use *
*Both phones in use *
*Incoming Calls *
Both phones ring
Phone in use receives call waiting notification, unused phone rings
Both phones receive call waiting notification
*Outgoing Calls *
Both phones can call out
The unused phone can call out
Neither phone can call out
* Inbound:
- Both ports will ring. Whichever port is picked up first, will field
the call.
- Any additional calls that come in would give call waiting
notification to the first line, and ring the second line.
- Once the second line is being utilized, all incoming calls will be
notifications in the form of call waiting beeps.
* Outbound:
- You will have the ability to dial out from port one.
- You will be able to dial a different party on port two.
*** Note ***
- If you have an active call on port one, and pick up port two,
you will NOT have the same call that is currently active on port one. The
Cloned Line will share the same voice mail and will have the same telephone
number as the original telephone line.
- The Cloned Line is NOT a second telephone number. The telephone
number that is assigned to the second phone port on the device is the same
telephone number as the number assigned to phone port one.
In sip.conf
[line1]
username=line1
secret=line1password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default
[line2]
username=line2
secret=line2password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default
In extensions.conf
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Macro(stdexten,${EXTEN},SIP/line1SIP/line2)
exten = 1234,n,Hangup()
or for trunk
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Gosub(stdexten(${EXTEN},SIP/line1SIP/line2))
exten = 1234,n,Hangup()
then stdexten would be default as comes in the sample configs...
That should be everything you want if you configure the SPA-2100 to register
line 1 with username line1 and line 2 with username line2...
d
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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] PMR446 interface
On Tuesday 04 August 2009 01:29:52 am Pascal Maugeri wrote: Hi Is there anybody here who has tried to interface Asterisk with PMR446 system (http://en.wikipedia.org/wiki/PMR446) using the native EM interface ? One way is the RoA (Radio over Asterisk) project http://code.google.com/p/radio-over-asterisk/ and also app_rpt http://app-rpt.qrvc.com/ We would like to use Amtelco product H.100 (http://xds.amtelco.com/h100.htm ). Regards, Pascal -- Thanks, Michael Maxwell eMail: metalm...@gmail.com Phone: +61 (03) 8680 4946 Web: mikey.webhop.org Powered By: PCBSD.org | FreeBSD.org | OpenSource.org PRAIL.org - Australia-Wide radio communications for free Sponsor: Hightek Hosting - A New Wave in IT and Hosting Technology Hosting, IT services, sales and onsite support! 1300 85 34 30 - www.hightekhosting.com.au ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Message Waiting Indicator on DAHDI line
2009/8/5 Mike asterisk-us...@norgie.net On Tue, Aug 04, 2009 at 03:35:22PM -0500, Doug Bailey wrote: This code is designed to handle Message Waiting Indication (MWI) incoming on FXO line. This data could very well be embedded in your CID spill as part of an MDMF message that also contains the caller id information. (See main/callerid.c in the callerid_feed function.) If your incoming line has a mailbox associated with it, the MWI information will be pushed to that mailbox. You may want to look at how your mailboxes are defined and the channels to which they are associated. Doug Doug, Thanks, that pointed me in the right direction. I found the problem in chan_dadhi.conf, I just don't totally understand it. signalling=fxo_ks echocancel=yes pulsedial=yes ;group=1 channel=1 ;mailbox=3203 signalling=fxs_ks ;group=1 usecallerid=no faxdetect=none signalling=fxs_ks rxgain=4 txgain=4 ;callerid=1234 channel=3 signalling=fxs_ks ;callerid=1234 echocancel=yes ;group=2 channel=4 Forgive the messy config, it's been a bit butchered in my efforts to get it working. The MWI problem was solved when I commented out the mailbox=3203 for chan 1. This is a phone that is dailed, along with the SIP phone, when the FXO lines ring. The SIP phone is tied against the same mailbox in the sip.conf. What I don't understand is, I thought the mailbox= in chan_dahdi.conf was there to do a stutter dialtone on the FXS port when there was a message in that mailbox? Why should commenting it out help? I assume that what happens is that the FXO line rings, so Asterisk rings the FXS phone as per the extensions.conf, this creates a MWI event which goes to the voicemail system, which then passes a MWI event to the SIP phone (as per sip.conf)? Or I could just be talk rubbish! The problem is that your mailbox line was below channel=1, as such, it applied to the next channel, channel=3 not channel=1... d ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] original reformat extension
Question:
Naturally there are times when need to I reformat an extension in a context as
such:
;Reformat add CC1
exten = _NXXNXX,1,Goto(1${EXTEN},1)
-or-
;Reformat 011 with with +CC
exten = _011X. ,1,Goto(+${EXTEN:3},1)
It's a helpful trick, BUT there are times when I want to send the call to
another context in its original un-reformatted state. Naturally the ${EXTEN}
variable has been changed. It occurred to me to use CALLERID(DNID) as such:
exten = _1NXXNXX,n(fail),Goto(other-context,${CALLERID(DNID)},1)
...but I am wondering if I can be 100% confident that CALLERID(DNID) will
always be equal to the 'original' ${EXTEN} variable. I observe that they are
the same, but it occurs to me that it may be up to the convention of my ITSP(s)
or PRI provider to populate that value equal to the original ${EXTEN}, and
therefore it may vary over time and between providers.
If the answer is NO, CALLERID(DNID) is not reliable and can vary between
providers and implementations, I then ask whether there is a RELIABLE place
from which to fetch that original EXTEN value without the added dialplan
overhead of saving it myself
Thanks!
-Karl
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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] original reformat extension
You could just store the initial value of ${EXTEN} in another channel
variable before doing any postprocessing on it. You can then refer to
the original received DNIS whenever you like elsewhere.
Karl Fife wrote:
Question:
Naturally there are times when need to I reformat an extension in a
context as such:
;Reformat add CC1
exten = _NXXNXX,1,Goto(1${EXTEN},1)
-or-
;Reformat 011 with with +CC
exten = _011X. ,1,Goto(+${EXTEN:3},1)
It's a helpful trick, BUT there are times when I want to send the call
to another context in its original un-reformatted state. Naturally the
${EXTEN} variable has been changed. It occurred to me to use
CALLERID(DNID) as such:
exten = _1NXXNXX,n(fail),Goto(other-context,${CALLERID(DNID)},1)
...but I am wondering if I can be 100% confident that CALLERID(DNID)
will always be equal to the 'original' ${EXTEN} variable. I observe
that they are the same, but it occurs to me that it may be up to the
convention of my ITSP(s) or PRI provider to populate that value equal to
the original ${EXTEN}, and therefore it may vary over time and between
providers.
If the answer is NO, CALLERID(DNID) is not reliable and can vary
between providers and implementations, I then ask whether there is a
RELIABLE place from which to fetch that original EXTEN value without the
added dialplan overhead of saving it myself
Thanks!
-Karl
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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] Server linux requirements
On 04/08/09 23:57, Miguel Molina wrote: Edwin Quijada escribió: It depends about your traffic. But myabe and I guess Core 2Duo 4Gb ram , Sata 160Gb +_ It's pretty well known that asterisk is CPU intensive, not RAM intensive. It think 4GB is much more than enough. BTW, if your asterisk is consuming more than 150-200MB of RAM and growing... suspect of memory leaks. I once read on this list than seeing 500MB of memory usage for an asterisk instance is *huge*. Huh? CPU load is generally down to things like transcoding... If you are switching just G.711 (a/ulaw) then CPU load is negligible in my experience. We've seen Asterisk systems for tens of users that run on tiny embedded processors or the low power VIA C7 type CPUs. I think it is more important to determine what level of reliability you need and engineer your server(s) around that. If you want Five 9s you'll need to buy the right hardware and probably duplicate it. Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] how to implement CLONED LINE Feature in asterisk?
By placing OPENSIP in front of Asterisk, we can register multiple
accounts, and we can successfully make call for Outgoing only. But in
case of incoming it fails.
If two users are registered with asterisk or OpenSIP then the user that
is registered latest is considered to be valid, and he is able to make
calls, other user with earlier registration can not make call.
My point here is in chain_sip.c what are variables or structure that
need to maintain so that we can consider all registered users as active
users.
Thanks!
Faheem
--- On Wed, 8/5/09, D Tucny d...@tucny.com wrote:
From: D Tucny d...@tucny.com
Subject: Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wednesday, August 5, 2009, 11:06 AM
2009/8/4 Faheem faheem_...@yahoo.com
how to implement CLONED LINE Feature in asterisk
Hey, I want to implement Clone Line feature in asterisk. I am using SPA-2100.
The feature should work in this way.
There are two ports in the SPA-2100 both are registered with asterisk with same
username/password, and have the same (phone number)
No
one on the phone
One
phone in use
Both
phones in use
Incoming
Calls
Both
phones ring
Phone
in use receives call waiting notification,
unused phone rings
Both
phones receive call waiting notification
Outgoing
Calls
Both
phones can call out
The
unused phone can call out
Neither
phone can call out
* Inbound:
- Both ports will ring. Whichever port is picked up first, will field the
call.
- Any additional calls that come in would give call waiting notification
to the first line, and ring the second line.
- Once the second line is being utilized, all incoming calls will be
notifications in the form of call waiting beeps.
* Outbound:
- You will have the ability to dial out from port one.
- You will be able to dial a different party on port two.
*** Note ***
- If you have an active call on port one, and pick up port two, you
will NOT have the same call that is currently active on port one. The Cloned
Line will share the same voice mail and will have the same telephone number as
the original
telephone line.
- The Cloned Line is NOT a second telephone number. The telephone number
that is assigned to the second phone port on the device is the same telephone
number as the number assigned to phone port one.
In sip.conf
[line1]
username=line1
secret=line1password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default
[line2]
username=line2
secret=line2password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default
In extensions.conf
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Macro(stdexten,${EXTEN},SIP/line1SIP/line2)
exten = 1234,n,Hangup()
or for trunk
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Gosub(stdexten(${EXTEN},SIP/line1SIP/line2))
exten = 1234,n,Hangup()
Re: [asterisk-users] original reformat extension
Karl Fife a écrit :
[...] there are times when I want to send the call to another context in its
original un-reformatted state. Naturally the ${EXTEN} variable has been
changed. It occurred to me to use CALLERID(DNID) as such:
exten = _1NXXNXX,n(fail),Goto(other-context,${CALLERID(DNID)},1)
Before goto
exten = _NXXNXX,1,Set(__DIALEDNUMBER=${EXTEN})
exten = _NXXNXX,n,Goto(1${EXTEN},1)
and then you always have the original unformated state in ${DIALEDNUMBER}
--
Daniel
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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] Calling issue for non-extension numbers
Kayton Sapale a écrit : Hi all, HI alone :-) Thanks to the previous replies that helped me with this before, but I got side-tracked in the middle of trying to figure this out, so apologies for posting the same issue. I use a Nokia e71, with an asterisk server and am having an issue dialing certain numbers. When I attempt to dial a local number, like xxx-xxx-, I cannot connect. What shows in the asterisk debug is the following: Found peer '104' However, if I try to dial an extension that is configured on the asterisk server, the call goes through fine. When I use another device to connect the server (another nokia actually) and dial a local number like xxx-xxx-, I see this in the debug dialog: Found peer '103' [...] Looking for 6789940793 in DLPN_Free_Outbound (domain sip.speartek.com) list_route: hop: sip:1...@192.168.111.183 It appears that my device cannot connect to the server when dialing certain numbers. Does anyone have any idea about this? From what you show us above there is nothing wrong. You should better debug your dialplan, specially if DLPN_Free_Outbound context allow numbers like 6789940793. Regards -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Gizmo Dial Out No CALLED PARTY AUDIO??
Rob a écrit : Hi all, Hi I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a while and it works fine I just added CALL OUT ... I have no problem with call setup ... the called party hears me ... but I can't hear them again if the call comes INTO the server both sides work fine. Looks like a nat issue: do you have nat=yes and canreinvite=no in your sip.conf for Gizmo5? -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] CDR Problem - No CDRs when call is not bridged
Miguel Molina schrieb: Klaus Darilion escribió: Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten = 997,1,Answer() exten = 997,2,Playback(tt-weasels) exten = 997,3,Hangup() exten = 999,1,Playback(tt-weasels|noanswer) exten = 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not for 999. How can I change this behavior? Thanks Klaus Try unanswered = yes on cdr.conf I tried, but it did not worked. Also, reading the documentation, it seems as this parameter set the behavior for outgoing channels (e.g. failed Dial()) . regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk Ipv6 with libss7 version 1.0.2
Hi All, I want to make an SS7 connection using latest available Asterisk IPv6 build. when I try to compile Asteriskv6-20080107 build with lib-ss7 1.0.2 version I get an error saying SS7_TRANSPORT_ZAP is not found. If I search for SS7_TRANSPORT_ZAP in libss7.h file, I donot find it. Looks like a version mismatch. Can anyone please suggest what needs to be done. Any suggestion on this would be of great help !! Thanks kavitha Matthew Fredrickson Digium, Inc. ___ -- 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 _ Log on to MSN India for a lowdown on what’s hot in the world today http://in.msn.com___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] CDR Problem - No CDRs when call is not bridged
FYI: I checked the sources and Asterisk does write CDRs only if the call in answered locally or forwarded to an outgoing channel. Thus, as workaround I wrapped the extensions behind Dial(Local/...) regards klaus Klaus Darilion schrieb: Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten = 997,1,Answer() exten = 997,2,Playback(tt-weasels) exten = 997,3,Hangup() exten = 999,1,Playback(tt-weasels|noanswer) exten = 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not for 999. How can I change this behavior? Thanks Klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk Vyatta routers solving NAT problems
Hi, Vyatta Asterisk works fine here. We are using traffic shaping DynDNS and NAT. Bye, Patrick On Tue, Aug 4, 2009 at 2:27 PM, Barry L. Klineblkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tarek Sawah wrote: First of all it acts like a firewall and a router.. compared to Cisco routers it has good ACL and firewall policies that can be used and written very well.. second it's easy to setup third my question is has anyone tested it ? and what are their openion regarding this? Tarek -- not only tested it but have it deployed in three different businesses. I have had no trouble whatsoever with Asterisk Vyatta. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD4DBQFKeCk5CFu3bIiwtTARAg7WAJjpXf/fvgcqh1AaZk5TAe0kalk/AJ4h07BK KKeo1MpuvfN9PmolAepbzg== =OaFh -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk sending an sms
Hi list, via the network of my VoIP-provider there is a possibility to send an sms. Now, I know that Asterisk can interact with an external sms-module/gsm-gateway, but how does one let Asterisk send an sms via the IP-network ? Like the Dial()-application, is there a similar way of letting Asterisk send an sms through the IP-connection with the VoIP-provider ?? Greetingz, Jonas Kellens. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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.conf parameter and sip msg between server - client
Hello I have few questions : - what's the difference between a subscribe request et a register request ? - in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please someone could explain how doest it work because I think i'm a little bit confuse. - if I configure a sip terminal in sip.conf like this [john] type=friend username=JOHN secret=mypassword host=dynamic context=default these affirmations are right or wrong : a) 'john' and 'mypassword' are variables which are used when I want to connect my softphone or phone to Asterisk server (register request) AND when I initiate a call (invite request)? b) 'dynamic' mean that [john] will be automatically registred to Asterisk server and 'qualify=yes' parameter may not be necessary ? c) in your softphone setting (here i use xlite), parameter 'username' must be the same as parameter 'username' in sip.conf ? d) in you softphone setting (here i use xlite), parameter 'Authorization username' must be the same as parameter [john] in sip.conf ? e) instead of using 'dynamic' for parameter 'host', if I use @IP or a hostname or FQDN, parameter qualify must set to' yes' or to '2000' in order to Asterisk server can know when [john] is reachable ? regards H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Gizmo Dial Out No CALLED PARTY AUDIO??
Yes ... as a matter of fact here is the sip.conf ... obviously private info removed [general] register = 1747xxx:x...@proxy01.sipphone.com1747xxx%3ax...@proxy01.sipphone.com port = 5060 bindaddr = 192.168.22.5 context = incoming svrlookup=yes ;dtmfmode=inband allow=all externip=76.98.xxx.xxx localnet=192.168.22.0/255.255.255.0 [proxy01.sipphone.com] nat=yes ;type=peer type=friend context=incoming disallow=all allow=ulaw allow=alaw allow=ilbc dtmfmode=rfc2833 host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com ;insecure=very deprecated; use insecure=port,invite instead insecure=port,invite qualify=yes secret= authuser=1747XXX fromuser=1747XXX username=1747XXX canreinvite=no On Wed, Aug 5, 2009 at 6:07 AM, Administrator TOOTAI ad...@tootai.netwrote: Rob a écrit : Hi all, Hi I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a while and it works fine I just added CALL OUT ... I have no problem with call setup ... the called party hears me ... but I can't hear them again if the call comes INTO the server both sides work fine. Looks like a nat issue: do you have nat=yes and canreinvite=no in your sip.conf for Gizmo5? -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Skype for Asterisk: Public Beta available
On Thu, Jul 30, 2009 at 8:50 PM, John Todd jt...@digium.com wrote: I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one. Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software. It crashes my box after the incoming call is answered :( http://betareports.digium.com/mantis/view.php?id=21 -- Shimi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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.conf parameter and sip msg between server - client
Hello, well let me explain one part of your question, the host parameter. if you want to restrict the access to one ip you can say it here. host=192.168.2.13 means, that you can only use this account from 192.168.0.13, eg. for security reasons. i recommend so set it to dynamic at the moment and if it works you can change if you wish. aot of information about the sip.conf you can find here: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf On Wed, Aug 5, 2009 at 2:32 PM, harry Rrhm.noa...@gmail.com wrote: Hello I have few questions : - what's the difference between a subscribe request et a register request ? - in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please someone could explain how doest it work because I think i'm a little bit confuse. - if I configure a sip terminal in sip.conf like this [john] type=friend username=JOHN secret=mypassword host=dynamic context=default these affirmations are right or wrong : a) 'john' and 'mypassword' are variables which are used when I want to connect my softphone or phone to Asterisk server (register request) AND when I initiate a call (invite request)? b) 'dynamic' mean that [john] will be automatically registred to Asterisk server and 'qualify=yes' parameter may not be necessary ? c) in your softphone setting (here i use xlite), parameter 'username' must be the same as parameter 'username' in sip.conf ? d) in you softphone setting (here i use xlite), parameter 'Authorization username' must be the same as parameter [john] in sip.conf ? e) instead of using 'dynamic' for parameter 'host', if I use @IP or a hostname or FQDN, parameter qualify must set to' yes' or to '2000' in order to Asterisk server can know when [john] is reachable ? regards H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] [asterisk]q: asterisk 1.6.1 install
hi just donwloaded the 1.6.1 branch and made configure install. so far so good. after staerting asterisk with: asterisk -cr Could not load features.conf == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls == Manager registered action Park == Manager registered action Bridge == Manager registered action DBGet == Manager registered action DBPut == Manager registered action DBDel == Manager registered action DBDelTree Asterisk Dynamic Loader Starting: No 'modules.conf' found, no modules will be loaded. Asterisk Ready. CLI but for example the command sip show peers is not present anymoream i missing something here? 2) add-ons whats the current deal with them, eg cdr-mysql? thx in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Gizmo Dial Out No CALLED PARTY AUDIO??
I do not use Gizmo for inbound, only out. I have a register line that
looks like yours. In addition I have this:
[general]
context=nonesaid
allowguest=no
allowoverlap=yes
allowtransfer=yes
realm=my system's host name
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
maxexpiry=3600
minexpiry=60
defaultexpiry=1200
qualifyfreq=60
notifymimetype=text/plain
disallow=all
allow=gsm
allow=ulaw
allow=alaw
mohinterpret=default
mohsuggest=default
language=en
videosupport=yes
callevents=yes
alwaysauthreject=yes
externip=mypublicIP
localnet=192.168.0.0/255.255.255.0
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=60
allowsubscribe=yes
callcounter=yes
counteronpeer=yes
registertimeout=20
registerattempts=10
nat=yes
canreinvite=nonat
[gizmo5]
type=peer
host=198.65.166.131
fromdomain=proxy01.sipphone.com
canreinvite=no
dtmfmode=rfc2833
insecure=port,invite
qualify=yes
fromuser=myusername
authuser=myusername
defaultuser=myusername
secret=mypass
context=sip-out
disallow=all
allow=ulaw
allow=alaw
Then in extensions.conf I have this:
exten = _9XX.,1,SIPAddHeader(No-Answer: true)
exten = _9XX.,n,Dial(SIP/gizmo5/${EXTEN:1},20)
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 5, 2009, at 6:02 AM, Rob wrote:
Yes ... as a matter of fact here is the sip.conf ... obviously
private info removed
[general]
register = 1747xxx:x...@proxy01.sipphone.com
port = 5060
bindaddr = 192.168.22.5
context = incoming
svrlookup=yes
;dtmfmode=inband
allow=all
externip=76.98.xxx.xxx
localnet=192.168.22.0/255.255.255.0
[proxy01.sipphone.com]
nat=yes
;type=peer
type=friend
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
dtmfmode=rfc2833
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
;insecure=very deprecated; use insecure=port,invite instead
insecure=port,invite
qualify=yes
secret=
authuser=1747XXX
fromuser=1747XXX
username=1747XXX
canreinvite=no
On Wed, Aug 5, 2009 at 6:07 AM, Administrator TOOTAI
ad...@tootai.net wrote:
Rob a écrit :
Hi all,
Hi
I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN
for a
while and it works fine I just added CALL OUT ... I have no
problem
with call setup ... the called party hears me ... but I can't hear
them
again if the call comes INTO the server both sides work fine.
Looks like a nat issue: do you have nat=yes and canreinvite=no in your
sip.conf for Gizmo5?
--
Daniel
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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] Message Waiting Indicator on DAHDI line
On Wed, Aug 05, 2009 at 02:13:01PM +0800, D Tucny wrote: The problem is that your mailbox line was below channel=1, as such, it applied to the next channel, channel=3 not channel=1... d Nice one. Thanks for spotting that. Mike. signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] [asterisk]q: asterisk 1.6.1 install
tom schrieb: hi just donwloaded the 1.6.1 branch and made configure install. so far so good. after staerting asterisk with: asterisk -cr Could not load features.conf == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls == Manager registered action Park == Manager registered action Bridge == Manager registered action DBGet == Manager registered action DBPut == Manager registered action DBDel == Manager registered action DBDelTree Asterisk Dynamic Loader Starting: No 'modules.conf' found, no modules will be loaded. Asterisk Ready. CLI Did you make samples? Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] [asterisk]q: asterisk 1.6.1 install
;-) thx On Wed, Aug 5, 2009 at 10:56 AM, Christian Victor christ...@victormedia.dewrote: tom schrieb: hi just donwloaded the 1.6.1 branch and made configure install. so far so good. after staerting asterisk with: asterisk -cr Could not load features.conf == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls == Manager registered action Park == Manager registered action Bridge == Manager registered action DBGet == Manager registered action DBPut == Manager registered action DBDel == Manager registered action DBDelTree Asterisk Dynamic Loader Starting: No 'modules.conf' found, no modules will be loaded. Asterisk Ready. CLI Did you make samples? Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Calling issue for non-extension numbers
Thanks Daniel. It looks like I didn't paste everything into the email, but not sure if this will make a difference: What I saw in debug with the device that does not work: Found peer '104' What I saw in debug with a device that does work: Found peer '103' Found RTP audio format 96 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 98 Found RTP audio format 13 Peer audio RTP is at port 192.168.111.183:49152 Found unknown media description format AMR for ID 96 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format iLBC for ID 97 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 98 Found audio description format CN for ID 13 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.111.183:49152 Looking for 6789940793 in DLPN_Free_Outbound (domain sip.speartek.com) list_route: hop: sip:1...@192.168.111.183 It just seems that the device that does not work cannot get past a certain point (or asterisk does not allow it past a certain point). -Kayton HI alone :-) Thanks to the previous replies that helped me with this before, but I got side-tracked in the middle of trying to figure this out, so apologies for posting the same issue. I use a Nokia e71, with an asterisk server and am having an issue dialing certain numbers. When I attempt to dial a local number, like xxx-xxx-, I cannot connect. What shows in the asterisk debug is the following: Found peer '104' However, if I try to dial an extension that is configured on the asterisk server, the call goes through fine. When I use another device to connect the server (another nokia actually) and dial a local number like xxx-xxx-, I see this in the debug dialog: Found peer '103' [...] Looking for 6789940793 in DLPN_Free_Outbound (domain sip.speartek.com) list_route: hop: sip:1...@192.168.111.183 It appears that my device cannot connect to the server when dialing certain numbers. Does anyone have any idea about this? From what you show us above there is nothing wrong. You should better debug your dialplan, specially if DLPN_Free_Outbound context allow numbers like 6789940793. Regards -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Calling issue for non-extension numbers
When I try that number I get a message on the device: Connection time-out I get the same message for other local numbers also. Message: 13 Date: Tue, 4 Aug 2009 16:22:11 -0500 From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] Calling issue for non-extension numbers To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: 68dc215933a34b169ff8c21f01596...@db0002 Content-Type: text/plain; charset=us-ascii It is probably a dialplan or timeout issue. What happens if you do 80055511212# ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Tuesday, August 04, 2009 4:13 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Calling issue for non-extension numbers Hi all, Thanks to the previous replies that helped me with this before, but I got side-tracked in the middle of trying to figure this out, so apologies for posting the same issue. I use a Nokia e71, with an asterisk server and am having an issue dialing certain numbers. When I attempt to dial a local number, like xxx-xxx-, I cannot connect. What shows in the asterisk debug is the following: Found peer '104' However, if I try to dial an extension that is configured on the asterisk server, the call goes through fine. When I use another device to connect the server (another nokia actually) and dial a local number like xxx-xxx-, I see this in the debug dialog: Found peer '103' Found RTP audio format 96 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 98 Found RTP audio format 13 Peer audio RTP is at port 192.168.111.183:49152 Found unknown media description format AMR for ID 96 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format iLBC for ID 97 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 98 Found audio description format CN for ID 13 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.111.183:49152 Looking for 6789940793 in DLPN_Free_Outbound (domain sip.speartek.com) list_route: hop: mailto:sip:1...@192.168.111.183 sip:1...@192.168.111.183 It appears that my device cannot connect to the server when dialing certain numbers. Does anyone have any idea about this? Thanks, Kayton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Calling issue for non-extension numbers
Kayton Sapale a écrit : Thanks Daniel. It looks like I didn't paste everything into the email, but not sure if this will make a difference: No need to send agian the same datas, I cutted non relevant part in my answer. From your other mail I'm sure that your problem is dialplan related. Could you increase verbosity to 3 or 4, pass a call and check what you have in console. Also review your dialplan to check why calls to 80055511212# finish in time out. Or simply modify your dialplan with something like exten = _8005.,1,dial(SIP/104) which should make ring your e71 at ext 104 when dialing any number of 5 digits starting with 8005 from your extension 103. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Several mailboxes on SIP peer
I have in my sip.conf the following [jon.moore] type=friend mailbox=8100,8150 In voicemail.conf, both mailboxes are defined. On my Aastra 480i phone, I only see the first mailbox listed. I've verified this, by changing mailbox= to reverse the order, and I then see 8150 when I go to Services Voicemail on the phone. I also only get MWI events for whichever mailbox is listed first. I would have expected to see the status of both mailboxes. Am I doing something wrong, or is this behavior how things are supposed to be? Thanks, jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Several mailboxes on SIP peer
On Wed, 2009-08-05 at 13:12 -0500, Jon Moore wrote: I have in my sip.conf the following [jon.moore] type=friend mailbox=8100,8150 In voicemail.conf, both mailboxes are defined. Have you tried 81008150 (using an ampersand instead of a comma)? -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best ISDN BRI solutions?
Hi all, For a while now I've been using Asterisk together with HFC-PCI cards (Cologne chipset) for Euro-ISDN BRI support. However, I do not consider this to be the most reliable solution and believe that the most stubborn problems have always been software related. If my clients are willing to spend a bit more money on different hardware, what do you think the best solution would be? I might even be willing to try out a more expensive PRI card if I knew it also supported BRI: just as long as I would no longer have to worry about the software support for it -- for both Asterisk 1.4 and 1.6. Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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.conf parameter and sip msg between server - client
On Wed, 2009-08-05 at 14:32 +0200, harry R wrote: - what's the difference between a subscribe request et a register request ? A subscription in the SIP protocol is saying Hey, I'd like to be notified when something happens. This is most often used when a phone wants to subscribe to the state of another extension, or to the status of a voicemail box. A registration is where one SIP device tells another Hey, I'm over here. If you get any calls for me, send them to me at this IP address and port. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk with gizmo5 and google voice only takes one call at a time.
my problem is this. I have google forward the call to gizmo5. I have this line in my sip file : register = user:passw...@proxy01.sipphone.com I believe this lines connects asterisk with gizmo5 so when it gets a call from Google, asterisk will answer it? At the end of my sip file i have this [Calls-From-Gizmo-Network] type=user context=demo disallow=all allow=ulaw allow=ilbc allow=gsm dtmfmode=rfc2833 host=proxy01.sipphone.com insecure=very username=user secret=password canreinvite=no In my extentions i have this: [fromgizmo] exten = s,1,Wait(5) exten = s,n,Answer exten = s,n,Wait(2) exten = s,n,Playback(welcome) exten = s,n,Playback(test) exten = s,n,Playback(test2) exten = s,n,Hangup The odd thing is i would have thought the context=demo line from sip.conf would play the demo in extensions? Instead it plays default which i put a line in to direct to fromgizmo... [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; include = fromgizmo why not play demo? Anyways, The first caller goes through just fine but the 2nd caller just gets a ringing. the output looks like this. -- Executing [...@default:1] Wait(SIP/198.65.166.147-084fe8b0, 5) in new s tack -- Executing [...@default:2] Answer(SIP/198.65.166.147-084fe8b0, ) in new stack -- Executing [...@default:3] Wait(SIP/198.65.166.147-084fe8b0, 2) in new s tack -- Executing [...@default:4] Playback(SIP/198.65.166.147-084fe8b0, welcome ) in new stack -- SIP/198.65.166.147-084fe8b0 Playing 'welcome' (language 'en') -- Executing [...@default:5] Playback(SIP/198.65.166.147-084fe8b0, test) i n new stack -- SIP/198.65.166.147-084fe8b0 Playing 'test' (language 'en') -- Executing [...@default:1] Wait(SIP/198.65.166.147-084fc2e0, 5) in new s tack -- Executing [...@default:2] Answer(SIP/198.65.166.147-084fc2e0, ) in new stack -- Executing [...@default:3] Wait(SIP/198.65.166.147-084fc2e0, 2) in new s tack -- Executing [...@default:4] Playback(SIP/198.65.166.147-084fc2e0, welcome ) in new stack -- SIP/198.65.166.147-084fc2e0 Playing 'welcome' (language 'en') -- Executing [...@default:5] Playback(SIP/198.65.166.147-084fc2e0, test) i n new stack -- SIP/198.65.166.147-084fc2e0 Playing 'test' (language 'en') -- Executing [...@default:6] Playback(SIP/198.65.166.147-084fe8b0, test2) in new stack -- SIP/198.65.166.147-084fe8b0 Playing 'test2' (language 'en') == Spawn extension (default, s, 5) exited non-zero on 'SIP/198.65.166.147-084fc2e0' -- Executing [...@default:7] Hangup(SIP/198.65.166.147-084fe8b0, ) in new stack == Spawn extension (default, s, 7) exited non-zero on 'SIP/198.65.166.147-084fe8b0' The odd thing is that to asterisk it looks like both calls are taken right? But whoever is the 2nd caller goes not get the call (it just rings and then goes to google voice mail). One more thing to note is that if i make one call online (from sip softphone) and the other from a land line or cell it works! Its only when i try to two phones (cell and/or land line) that it does not. How can i get two phones connected? Thanks! _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Several mailboxes on SIP peer
On Wed, Aug 5, 2009 at 1:47 PM, Jared Smithjsm...@digium.com wrote: Have you tried 81008150 (using an ampersand instead of a comma)? Just changed it. Reloaded asterisk and restarted the phone. Same behavior as before. Well, only a single mailbox shows up anyways. -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Several mailboxes on SIP peer
Jonathan Moore wrote: Just changed it. Reloaded asterisk and restarted the phone. Same behavior as before. Well, only a single mailbox shows up anyways. Add the @context on each of the mailboxes: mailbox=8...@yourcontext,8...@yourcontext Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Several mailboxes on SIP peer
On Wed, Aug 5, 2009 at 2:55 PM, Doug Lytlesupp...@drdos.info wrote: Jonathan Moore wrote: Just changed it. Reloaded asterisk and restarted the phone. Same behavior as before. Well, only a single mailbox shows up anyways. Add the @context on each of the mailboxes: mailbox=8...@yourcontext,8...@yourcontext Shall give that a try. After I make this change, how's the best way to check to see if the phone is checking both mailboxes? Would those type messages appear in SIP debugging? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Several mailboxes on SIP peer
Just to update on my troubles. I noticed the MWI light wasn't coming on when I received a new message, so I removed the mailbox= from sip.conf and added just a single mailbox. Now, my sip.conf looks as follows.. [jon.moore] type=friend mailbox=8...@default And I do get the message indicator. (as to my last email, sip set debug does what I think I need). When I add two mailboxes, I never see the message in debug about waiting messages. So, it seems I was mistaken earlier, and I'm not get MWI events when two mailboxes are configured as I thought I was. -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best ISDN BRI solutions?
I might even be willing to try out a more expensive PRI card if I knew it also supported BRI: just as long as I would no longer have to worry about the software support for it -- for both Asterisk 1.4 and 1.6. Thanks, Jaap You can use Sangoma Media Gateway along with Asterisk ( http://wiki.sangoma.com/sangoma-wanpipe-smg-asterisk-bri-installation) That is known to work pretty well for lots of people. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] how to implement CLONED LINE Feature in asterisk?
I'd suggest using different user names and getting asterisk to handle the
cleverness... And, well, doing it this way is pretty simple, straight
forward, basic asterisk functionality...
Trying to get two different instances registered as the same user, is, as
you've found out, not going to be trivial to implement... It's just not how
it works...
If you implemented what I suggested below, using two different usernames for
the two ports on the SPA, it would just work...
d
2009/8/5 Faheem faheem_...@yahoo.com
By placing OPENSIP in front of Asterisk, we can register multiple accounts,
and we can successfully make call for Outgoing only. But in case of incoming
it fails.
If two users are registered with asterisk or OpenSIP then the user that is
registered latest is considered to be valid, and he is able to make calls,
other user with earlier registration can not make call.
My point here is in chain_sip.c what are variables or structure that need
to maintain so that we can consider all registered users as active users.
Thanks!
Faheem
--- On *Wed, 8/5/09, D Tucny d...@tucny.com* wrote:
From: D Tucny d...@tucny.com
Subject: Re: [asterisk-users] how to implement CLONED LINE Feature in
asterisk?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wednesday, August 5, 2009, 11:06 AM
2009/8/4 Faheem
faheem_...@yahoo.comhttp://mc/compose?to=faheem_...@yahoo.com
how to implement CLONED LINE Feature in asterisk
Hey, I want to implement Clone Line feature in asterisk. I am using
SPA-2100.
The feature should work in this way.
There are two ports in the SPA-2100 both are registered with asterisk with
same username/password, and have the same (phone number)
*No one on the phone *
*One phone in use *
*Both phones in use *
*Incoming Calls *
Both phones ring
Phone in use receives call waiting notification, unused phone rings
Both phones receive call waiting notification
*Outgoing Calls *
Both phones can call out
The unused phone can call out
Neither phone can call out
* Inbound:
- Both ports will ring. Whichever port is picked up first, will
field the call.
- Any additional calls that come in would give call waiting
notification to the first line, and ring the second line.
- Once the second line is being utilized, all incoming calls will be
notifications in the form of call waiting beeps.
* Outbound:
- You will have the ability to dial out from port one.
- You will be able to dial a different party on port two.
*** Note ***
- If you have an active call on port one, and pick up port two,
you will NOT have the same call that is currently active on port one. The
Cloned Line will share the same voice mail and will have the same telephone
number as the original telephone line.
- The Cloned Line is NOT a second telephone number. The telephone
number that is assigned to the second phone port on the device is the same
telephone number as the number assigned to phone port one.
In sip.conf
[line1]
username=line1
secret=line1password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default
[line2]
username=line2
secret=line2password
type=friend
host=dynamic
context=outboundcalls
mailbox=1...@default
In extensions.conf
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Macro(stdexten,${EXTEN},SIP/line1SIP/line2)
exten = 1234,n,Hangup()
or for trunk
[default]
exten = 1234,1,NoOp(About to dial both phones)
exten = 1234,n,Gosub(stdexten(${EXTEN},SIP/line1SIP/line2))
exten = 1234,n,Hangup()
then stdexten would be default as comes in the sample configs...
That should be everything you want if you configure the SPA-2100 to
register line 1 with username line1 and line 2 with username line2...
d
-Inline Attachment Follows-
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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] CDR Problem - No CDRs when call is not bridged
Klaus Darilion wrote: FYI: I checked the sources and Asterisk does write CDRs only if the call in answered locally or forwarded to an outgoing channel. Thus, as workaround I wrapped the extensions behind Dial(Local/...) regards klaus Klaus Darilion schrieb: Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten = 997,1,Answer() exten = 997,2,Playback(tt-weasels) exten = 997,3,Hangup() exten = 999,1,Playback(tt-weasels|noanswer) exten = 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not for 999. How can I change this behavior? Thanks Klaus This is the intended behavior, you should always use answer if you will handle the call with an IVR, otherwhise you also can cause problems on the remote end, for instance, if they are calling you from a CIsco 79xx phone and the phone never gets an answered state message the soft keys never switch to allow placing the call on hold or transferring the call, or selecting join if they where trying to do a three-way call to you. Please, instead of looking for Asterisk to change it's behavior, in this case I would implore you to change yours, as it may get you into trouble. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Several mailboxes on SIP peer
On Wednesday 05 August 2009 15:13:47 Jonathan Moore wrote: Just to update on my troubles. I noticed the MWI light wasn't coming on when I received a new message, so I removed the mailbox= from sip.conf and added just a single mailbox. Now, my sip.conf looks as follows.. [jon.moore] type=friend mailbox=8...@default And I do get the message indicator. (as to my last email, sip set debug does what I think I need). When I add two mailboxes, I never see the message in debug about waiting messages. So, it seems I was mistaken earlier, and I'm not get MWI events when two mailboxes are configured as I thought I was. Are you using plaintext storage, ODBC storage, or IMAP storage for your voicemail messages? -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Several mailboxes on SIP peer
On Wed, Aug 5, 2009 at 4:05 PM, Tilghman Leshertilgh...@mail.jeffandtilghman.com wrote: Are you using plaintext storage, ODBC storage, or IMAP storage for your voicemail messages? Plain storage. My voicemail.conf is just about the same as the sample config that's installed, with the expection of the mailboxes added. Those lines are like this.. 8100 = 1234,Jonathan 8500 = 1234,Support -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Strange Case.
Greetings again List. I'm facing a strange case with one of the productive Asterisk servers.. i have 3 providers sending traffic to the call center where agents pickup the calls. calls come into the server Queue Agents Last October .. an undersea cable got disconnected placing Egypt and the countries in the region offline.. when internet came back .. the call center located in Egypt had no SIP protocol working.. and we shifted to IAX.. 26 days later SIP started to work again .. but since then calls started to disconnect out of the blue.. we get calls that may last for 45 minutes.. and end normaly .. and we get calls that ring and disconnect the moment the agent picks up been facing a problem with my client as they use the Flash Operator Panel to monitor the call flow through the server and the regualr setup Queue Local users won't work for them as the Flash operator flash offline static agents as online so the client won't know who is on and who is off.. and it's impossible to teach the agents to Login and Logoff the Queue.. so the only solution is the following.. Caller Queue FindMeFollowMe Extension Local SIP extensions this way .. my client is able to monitor the calls and things won't get complicated.. (this is the setup we have been using for 6 months before the problem with the internet occures) since the internet problem and calls are getting disconnected .. out of the blue.. nothing has changed.. and to make sure things are going well .. we moved the server to a Hosting company in California with 10 mb/s connection speed.. (Same Setup that was working well) and still calls get disconnected.. after a lot of problems with the client .. i asked them to change the ISP (my prime suspect was the internet) and finaly they managed to change the ISP .. but the problem is still there.. my server informations are the following Asterisk 1.4.22-3 Uname -a: Linux 2.6.18-92.1.18.el5 sip.conf ;;Agent Sample from Sip.conf [3000] type=friend secret=3000 qualify=yes port=5060 disallow=all allow=g729 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/3000 context=from-internal canreinvite=no call-limit=1 busy-limit=1 ;;Provider's Sample from Sip.conf [50011] type=peer qualify=yes port=5060 pickupgroup= nat=no host=XXX.YYY.ZZZ.NNN disallow=all allaw=alaw allaw=ulaw allow=g729 dial=SIP/50011 context=from-internal canreinvite=no deny=0.0.0.0/0.0.0.0 permit=XXX.YYY.ZZZ.NNN/255.255.255.255 # extensions.conf ;;the provider sends calls to Virtual DIDs (Extensions) in my system which is 8000 exten = 8000,1,GotoIfTime(07:00-16:00|sat-fri|1-31|jan-dec?ext-queues,*8000,1) exten = 8000,n,Answer exten = 8000,n,Queue(8000,t,,,10) exten = 8000,n,Dial(IAX2/6005:6...@backupserver/11) ;; sends the call to a backup server. exten = *8000,1,Answer exten = *8000,n,Dial(IAX2/6005:6...@backupserver/11) the Providers strictly send calls with codec G.729 my agents get best voice quality with G.711u I need your advice .. am i missing anything in this setup?? it used to work .. and it STILL works on another hosted server with Agents located in Morocco.. with a different version of Asterisk 1.4.20-1 and better hold time for the calls.. -- AHD Tarek Sawah _ Get your vacation photos on your phone! http://windowsliveformobile.com/en-us/photos/default.aspx?OCID=0809TL-HM___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best ISDN BRI solutions?
We use Patton BRI gateways. No problems so far. If possible, we prefer to keep telephony interfaces out of Asterisk box. Regards Jorge Jaap Winius wrote: Hi all, For a while now I've been using Asterisk together with HFC-PCI cards (Cologne chipset) for Euro-ISDN BRI support. However, I do not consider this to be the most reliable solution and believe that the most stubborn problems have always been software related. If my clients are willing to spend a bit more money on different hardware, what do you think the best solution would be? I might even be willing to try out a more expensive PRI card if I knew it also supported BRI: just as long as I would no longer have to worry about the software support for it -- for both Asterisk 1.4 and 1.6. Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best ISDN BRI solutions?
On Wed, Aug 5, 2009 at 5:28 PM, Jorge Mendoza mend...@tcc.com.pe wrote: We use Patton BRI gateways. No problems so far. If possible, we prefer to keep telephony interfaces out of Asterisk box. Regards Jorge Just for the record, Sangoma Media Gateway does exactly that, leave all your PSTN interfaces (BRI, SS7, PRI) in another box and communicates with Asterisk through the Woomera protocol. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best ISDN BRI solutions?
Quoting Jorge Mendoza mend...@tcc.com.pe: We use Patton BRI gateways. No problems so far. If possible, we prefer to keep telephony interfaces out of Asterisk box. What a great idea! I'm going to remember that. Unfortunately, I believe that would be of no use if you also wanted to use your ISDN connection for a networked fax system, such as with Hylafax and IAXmodem. Cheers, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Voicemail feature: enable or disable the ability to leave a message
At 11:24 PM on 31 Jul 2009, Emrah wrote: Doug, Thanks for the suggestion. I know there are plenty of workarounds there, I am not asking how to do it because I know how to do it too. What I am saying is that it could be an embedded feature in the Voicemail application, like the recent ability to flag a message as urgent. Maybe the work being done (been done?) through Google Summer of Code 2009[1] will net some progress in this direction. MiniVM looks like a pretty good idea... but it's got a long way to go. I seriously doubt app_voicemail itself will get any new features, regardless of how many good ideas get thrown around. Just like what happened with AgentCallbackLogin, I think instead of augmenting the incumbent system, Digium will probably replace VoiceMail and VoiceMailMain with an equivalent dialplan solution using MiniVM. It's much more flexible that way. [1] http://lists.digium.com/pipermail/asterisk-dev/2009-April/038028.html Doug Lytle wrote: Emrah wrote: Mark, I think you did not understand my message. I am accustomed to have the option to allow or disallow the recording of a message in my voicemail, even my mobile carrier provides it. E.g.: I The simplest thing to do is to allow users to set a flag, maybe using mysql or the astdb, if they want that option. And, in your dial plan, check for the existance of that flag. If it's there, then don't jump to the voice mail app, just jump to your context that would play back an audio file that the user has pre-recorded Doug -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Fwd: User Authentication in sip.conf
Please any one help for this problem. -- Forwarded message -- From: velusamy velu velu.techni...@gmail.com Date: Mon, Aug 3, 2009 at 10:22 AM Subject: User Authentication in sip.conf To: asterisk-users@lists.digium.com Dear all, I want to setup the incoming calls, that don't use authentication in sip.conf file. My configurations as follows, [2000] type=peer host=dynamic insecure=port,invite; (both) context=Testing But when I call '2000', I noticed the following message in Asterisk console, NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user Velusamy sip:7...@192.168.1.222sip%3a...@192.168.1.222 ;tag=yj66acQcycvrN What would be the problem?? Please help me to solve this problem. Best Regards, Velusamy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best ISDN BRI solutions?
Jaap Winius wrote: Quoting Jorge Mendoza mend...@tcc.com.pe: We use Patton BRI gateways. No problems so far. If possible, we prefer to keep telephony interfaces out of Asterisk box. What a great idea! I'm going to remember that. Unfortunately, I believe that would be of no use if you also wanted to use your ISDN connection for a networked fax system, such as with Hylafax and IAXmodem. Sure it is. Just get a media gateway that does T.38 - and does it relatively well. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] dialplan tips
On Fri, Jul 24, 2009 at 08:28:48AM -0500, Danny Nicholas wrote: Here's how I think your dialplan should look: exten = 101,1,Ringing exten = 101,2,Answer() exten = 101,3,Dial(SIP/quentin,10) exten = 101,n,VoiceMail(1...@default,u) exten = 101,n,Playback(vm-goodbye) exten = 101,n,Hangup() exten = 101-BUSY,1,Playback(busy) exten = 101-BUSY,n,Wait(3) exten = 101-BUSY,n,VoiceMail(1...@default,b) exten = 101-BUSY,n,Playback(vm-goodbye) exten = 101-BUSY,n,Hangup() Hi I have a question about this dialplan, why does the dial do a jump to 101-DIALSTATUS, is there a goto 101-DIALSTATUS missing ? Alex [snip] signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
