[asterisk-users] RTPAUDIOQOS
hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval ___ -- 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] RTPAUDIOQOS
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: 2009 m. rugsėjo 22 d. 09:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitt er=0.00;txcount=83;rlp=0;rtt=14818.715000 if any one know plese help me to or give any documentation link regards Dhaval ___ -- 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] Call deflection on Asterisk 1.6.1.6
I'm using a Asterisk 1.6.1.6 with dahdi. We need to redirect phone calls to a certain number when there is nobody. So I read about call reflection but the call reflection applications on bristuff are not for 1.6.1.6. Are there any other applications or patches that provides call reflection for Asterisk 1.6.1.6?? Greetz TM ___ -- 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] Problem with dialplan - gotoif ?
Hi
This is the output from show dialplan dial-sipmnf-sippt-pstn
[ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ]
's' =1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config]
2. Dial(SIP/${ar...@${sipmnf},${ARG2},${OUTBDIAL})
[pbx_config]
3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]
4. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pt:ok) [pbx_config]
[pt] 5. Dial(SIP/${ar...@${sippt},${ARG2},${OUTBDIAL})
[pbx_config]
6. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]
7. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pstn:ok)
[pbx_config]
[pstn] 8. Dial(${PSTN}/w${ARG1},${ARG2},${OUTBDIAL}) [pbx_config]
[ok] 9. Goto(dialer-exit,s,1(${ARG3}) [pbx_config]
-- Executing [0296332...@in-uniden:1] Verbose(DAHDI/1-1, 1,Dialing
0296332828 normal) in new stack
Dialing 0296332828 normal
-- Executing [0296332...@in-uniden:2] Log(DAHDI/1-1, Notice,Dialing
0296332828 normal) in new stack
[Sep 22 18:20:42] NOTICE[18347]: Ext. 0296332828:2 @ in-uniden: Dialing
0296332828 normal
-- Executing [0296332...@in-uniden:3] Gosub(DAHDI/1-1,
dial-sipmnf-sippt-pstn,s,1(0296332828,70)) in new stack
-- Executing [...@dial-sipmnf-sippt-pstn:1] Verbose(DAHDI/1-1, 1,Dialing
0296332828 on mnf pt pstn) in new stack
Dialing 0296332828 on mnf pt pstn
-- Executing [...@dial-sipmnf-sippt-pstn:2] Dial(DAHDI/1-1,
SIP/0296332...@mynetfone-09105023,70,WKT) in new stack
== Using SIP RTP CoS mark 5
-- Called 0296332...@mynetfone-09105023
-- SIP/MyNetFone-09105023-097809e0 is making progress passing it to
DAHDI/1-1
-- Got SIP response 486 Busy Here back from 125.213.160.81
-- SIP/MyNetFone-09105023-097809e0 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [...@dial-sipmnf-sippt-pstn:3] Set(DAHDI/1-1,
GLOBAL(FOUNDME)=BUSY) in new stack
== Setting global variable 'FOUNDME' to 'BUSY'
-- Executing [...@dial-sipmnf-sippt-pstn:4] GotoIf(DAHDI/1-1, [BUSY =
CHANUNAVAIL]?pt:ok) in new stack
-- Goto (dial-sipmnf-sippt-pstn,s,5)
-- Executing [...@dial-sipmnf-sippt-pstn:5] Dial(DAHDI/1-1,
SIP/0296332...@pennytel-8889186044,70,WKT) in new stack
i believe i have captured the relevant logging from the console. my problem is
with Gotoif statement
-- Executing [...@dial-sipmnf-sippt-pstn:4] GotoIf(DAHDI/1-1, [BUSY =
CHANUNAVAIL]?pt:ok) in new stack
-- Goto (dial-sipmnf-sippt-pstn,s,5)
from my understanding BUSY != CHANUNAVAIL, therefor it should have jumped to ok
which is s,9.
What have I missed !
thanks
Alex
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] Call deflection on Asterisk 1.6.1.6
We also need 'call deflection' and 'call redirection' using ISDN PRI (ETSI) and currently using Asterisk 1.6.1. If any one can point us in the right direction, or what is on the horizon would be a good start. Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom maris Sent: Tuesday, 22 September 2009 8:08 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call deflection on Asterisk 1.6.1.6 I'm using a Asterisk 1.6.1.6 with dahdi. We need to redirect phone calls to a certain number when there is nobody. So I read about call reflection but the call reflection applications on bristuff are not for 1.6.1.6. Are there any other applications or patches that provides call reflection for Asterisk 1.6.1.6?? Greetz TM ___ -- 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
[asterisk-users] SIPP question
Hello I would like to play file with sipp command. I want to take value of RTPAUDIOQOS for every user.. I will make it hard testing with 500 users. But when all user leave from this conference I am unable to receive proper value for highlighted in below line.. ssrc=877077954;* themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00 * However, when i made single call using SIP phone then i will receive all value from RTPAUDIOQOs. Any Idea.. how can I play or transfer/receive Audio packets while testing with SIPP command [using below command] I need specially value of receive streams . *./sipp -sn uac -d 1080 -s 8601 127.0.0.1 -l 50 -r 1 -rp 5000* ___ -- 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] problem in upgrading to 1.6.1.0
Hi D, Thanks for the suggestion. I put type = user in [general] of users.conf and that seems to have fixed it. Best regards, Örn On Mon, Sep 21, 2009 at 11:23 AM, D Tucny d...@tucny.com wrote: In the 1.6.1.* branch the line type=peer seems to be required on each user... d 2009/9/19 Örn Arnarson o...@arnarson.net Sorry I wasn't more specific. The error message is just the standard 'Can't find that extension'. The problem is, however, that asterisk parses users.conf (and doesn't complain), but none of the users specified therein are loaded into the dialplan or even shown as peers (using sip show peers/users). A downgrade from 1.6.1.6 to 1.6.0.9 promptly fixed it, as with Oguzhan. Regards, Örn 2009/9/18 Benny Amorsen benny+use...@amorsen.dk: Örn Arnarson o...@arnarson.net writes: I'm seeing the same behavior in 1.6.1.6. Any info on this? It would be helpful if you copied the exact error message involving the username field. /Benny ___ -- 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] Problem with dialplan - gotoif ?
Alex Samad wrote:
4. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pt:ok)
[pbx_config]
i believe i have captured the relevant logging from the console. my problem
is with Gotoif statement
-- Executing [...@dial-sipmnf-sippt-pstn:4] GotoIf(DAHDI/1-1, [BUSY =
CHANUNAVAIL]?pt:ok) in new stack
-- Goto (dial-sipmnf-sippt-pstn,s,5)
from my understanding BUSY != CHANUNAVAIL, therefor it should have jumped to
ok which is s,9.
What have I missed !
You have missed the leading $ in front of the $[...] portion.
Add the $, and things should start working as you expect.
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
___
-- 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 on a Beagleboard?
Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. ___ -- 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] RTPAUDIOQOS
Have you looked at voip-info.org? I'm going to guess at what these mean and I'll bet a coffee that someone will correct what I say wrong: Ssrc = local ip address Themsrc = remote (provider) ip address Rxjitter = QOS variance for received jitter buffers Rxcount = packets received Txjitter = transmitted variance Txcount = packets transmitted Rlp = ?? Rtt = ?? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Tuesday, September 22, 2009 1:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitt er=0.00;txcount=83;rlp=0;rtt=14818.715000 if any one know plese help me to or give any documentation link regards Dhaval ___ -- 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 on a Beagleboard?
Vincent wrote: Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. ___ -- 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 128m of ram 256 m flash for the 'hard drive' is not much in either catagory. And ethernet is a USB addon, not on the board. ___ -- 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 on a Beagleboard?
I was going to dismiss this, but it does offer an interesting possibility; Since it can boot Debian ARM from an SD card, you could have Asterisk-in-a-can where you would have the Debian build and Asterisk on the SD card and could hook up to a USB hub (for Ethernet connectivity) and process up to 14GB of call-data before having to offload to permanent/traditional media. If you really went nuts, you could possibly even power and use a DAHDI device off of some USB-powered peripheral. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Tuesday, September 22, 2009 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on a Beagleboard? Vincent wrote: Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. ___ -- 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 128m of ram 256 m flash for the 'hard drive' is not much in either catagory. And ethernet is a USB addon, not on the board. ___ -- 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 on a Beagleboard?
Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. 128m of ram 256 m flash for the 'hard drive' is not much in either catagory. And ethernet is a USB addon, not on the board. It appears to support a SD/MMC card, meaning that it can support gigs of low power storage space. Or a USB HDD for higher power storage space. 128m of RAM isn't a lot, but some people are apparently running Asterisk on 32MB Linksys WRT54GS's (OpenWRT). If you were careful and cautious, it'd probably work. The Ethernet as an add-on kinda stinks and is probably the largest negative. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] DTMF end '1' has duration 57 but want minimum 80 , emulating on ?IAX2/a16-q1-9657
On 09/21/2009 12:46 PM, Tilghman Lesher wrote: On Sunday 20 September 2009 22:32:41 Tzafrir Cohen wrote: Isn't 40ms the minimal time for a valid dtmf digit? $ grep -C1 'define AST_MIN_DTMF_DURATION' main/channel.c /*! Minimum allowed digit length - 80ms */ #define AST_MIN_DTMF_DURATION 80 This post, from the archives, is instructive and germane to the code in question: http://lists.digium.com/pipermail/asterisk-dev/2007-April/027271.html If a DTMF decoder doesn't register a digit in under 40ms its broken. 80ms is the minimum total digit cycle time many people specify, including the silence between digits. This time varies between specs, though. Some things specify the minimum cycle as 100ms. Steve ___ -- 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] setting up a IP based voip carrier account
Hellos,
My voip carrier has assigned me a IP based account...where they only give me
the IP to call through. I have setup the dial plan
exten = _7XXX.,1,Answer()
exten = _7XXX.,2,vmauthenticate(${CALLERID(number)})
exten = _7XXX.,3,Dial(SIP/${EXTEN:1...@y.y.y.y)
exten = _7XXX.,4,Hungup()
Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error
SIP/Y.Y.Y.Y-35dc is circuit-busy
Are there any settings I am leaving out?
Thanks
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM
can help you achieve better customer satisfaction and sales
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 on a Beagleboard?
Probably worth discussing this over on the AstLinux list as they are all about embedded Asterisk running on machines like this. On 09/22/2009 09:48 AM, Danny Nicholas wrote: I was going to dismiss this, but it does offer an interesting possibility; Since it can boot Debian ARM from an SD card, you could have Asterisk-in-a-can where you would have the Debian build and Asterisk on the SD card and could hook up to a USB hub (for Ethernet connectivity) and process up to 14GB of call-data before having to offload to permanent/traditional media. If you really went nuts, you could possibly even power and use a DAHDI device off of some USB-powered peripheral. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Tuesday, September 22, 2009 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on a Beagleboard? Vincent wrote: Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. ___ -- 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 128m of ram 256 m flash for the 'hard drive' is not much in either catagory. And ethernet is a USB addon, not on the board. ___ -- 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
[asterisk-users] AstManProxy - No pid file created when run
Hi, Looking at /etc/init.d/astmanproxy file, it seems astmanproxy should create a /var/run/astmanproxy.pid file when run. I can't find any such astmanproxy.pid (not in /var/run directory, nor anywhere else). Is it normal ? (I'm using Proxy/1.22pre081119) Regards ___ -- 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] setting up a IP based voip carrier account
James Mutuku wrote:
Hellos,
My voip carrier has assigned me a IP based account...where they only
give me the IP to call through. I have setup the dial plan
exten = _7XXX.,1,Answer()
exten = _7XXX.,2,vmauthenticate(${CALLERID(number)})
exten = _7XXX.,3,Dial(SIP/${EXTEN:1...@y.y.y.y)
exten = _7XXX.,4,Hungup()
Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error
SIP/Y.Y.Y.Y-35dc is circuit-busy
Are there any settings I am leaving out?
No; the circuit-busy error is due to negative SIP feedback that
peer is sending you; perhaps a 404 Not Found or a 503 Service
Unavailable or something of the sort. To find out what the essence of
the problem is, you may need to turn up verbosity (core set verbose
60) on the CLI. If that doesn't divine it, do a packet capture:
tcpdump -i ethX -A -s 0 -n udp port 5060 and host Y.Y.Y.Y
Some consider it good practice to add a SIP peer in sip.conf for that
endpoint regardless. You don't have to provide any authentication
information (username, secret) in the peer definition. This will also
be necessary - not optional - if you plan to accept incoming calls
from the same carrier, so that you can set it not to challenge the
carrier for authentication credentials:
insecure=port,invite
and to route the calls into a particular dial plan context:
context=incoming-calls
-- Alex
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
-- 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 on a Beagleboard?
On Tue, Sep 22, 2009 at 02:56:55PM +0200, Vincent wrote: Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. That board lacks a network adapter in the default configuration. Make sure you add one. A different Arm: SheevaPlug, or OpenRD. http://www.openplug.org/ I managed to build trunk on my SheevaPlug . Only thing that is left to fix not is a minor fix to libgsm, but then again, you'd want to use libgsm from Debian anyway :-) (the fix is to tell the configure script not to attempt any special platform-specific optimizations in the case of armv5). The latest releases of a number of days ago should include a number of minor fixes that should help build on Arm. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] SIPP question
You need to compile sipp with pcap support. Here is an example scenario: http://sipp.sourceforge.net/doc3.0/reference.html#UAC+with+media On Sep 22, 2009, at 5:13 AM, DHAVAL INDRODIYA wrote: Hello I would like to play file with sipp command. I want to take value of RTPAUDIOQOS for every user.. I will make it hard testing with 500 users. But when all user leave from this conference I am unable to receive proper value for highlighted in below line.. ssrc = 877077954 ;themssrc = 0 ;lp = 0 ;rxjitter =0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00 However, when i made single call using SIP phone then i will receive all value from RTPAUDIOQOs. Any Idea.. how can I play or transfer/receive Audio packets while testing with SIPP command [using below command] I need specially value of receive streams . ./sipp -sn uac -d 1080 -s 8601 127.0.0.1 -l 50 -r 1 -rp 5000 ___ -- 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
[asterisk-users] How to associate a custom script to core stop now
Hello, I would to create a custom file whenever a core stop now command is typed through Asterisk console. (If this file is present, it would mean admin has choosen to stop Asterisk and it shouldn't be tried to restart Asterisk automatically). Obviously, when such core stop now command is typed, /etc/init.d/asterisk stop is not launched so I can't use this single /etc/init.d/asterisk as a way to tell what should or shouldn't be done when Asterisk is starting or stopping. I can't see either in man asterisk any option with which I can specify an action to launch when asterisk stops. Any idea or suggestion ? Regards ___ -- 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 to email transcribed
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 C F wrote: I have seen lots of companies offering this as a service and have used phonetag.com in the past. They work very nicely, however I have a customer that is not interested in paying $30-$40 a month but would rather buy the software. I have googled and googled all I can come up with are companies that do it as hosted. Does anyone on the list know of software that can transcribe an email/voicemail sent to it and then forward it to the end user? TIA We've talked about this on the VoIP users conference and the feeling is that most likely there is a human on the back end who is doing the transcription, if it is to be at all accurate. Anyone who has worked with the various speech-to-text software, such as ViaVoice or Dragon, knows that training is the key to really accurate transcription. That, and a good quality audio signal. The variations in audio quality you get in voicemail is probably too great to do this all with software only. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKuO3KCFu3bIiwtTARAqWbAJwOhC3REHwQphVpXrG+XfGKwq3ccwCfRoO4 nOv6mEfTV4rQst89YU7/Wpg= =HpnP -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
Re: [asterisk-users] SIPP question
Also, you can pass -sn uac_pcap to use the default pcap_play scenario which will play 8 seconds of audio and a dtmf digit. You can also run sipp -sd uac_pcap uac_pcap.xml to save the scenario and edit it. You can then play it back using sipp -sf uac_pcap.xml ... ___ -- 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] RTPAUDIOQOS
Mindaugas Kezys schrieb: Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified In the given example: *ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000* How do I interprete the jitter value ? Is the value 0.020917 good ? Bad ? Is there a unit behind this value ? Thanks Regards Hans *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA *Sent:* 2009 m. rugsėjo 22 d. 09:28 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell *ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000* if any one know plese help me to or give any documentation link regards Dhaval ___ -- 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] Voicemail to email transcribed
Whats interesting is that most message are only around 90% accurate, if it would be a human it would be way more accurate upwards to around 97% in my opinion. There are only so many mistakes a human can make, in the case of phonetag I don't think it's humans since there are too many obvious mistakes. On Tue, Sep 22, 2009 at 11:31 AM, Barry L. Kline blkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 C F wrote: I have seen lots of companies offering this as a service and have used phonetag.com in the past. They work very nicely, however I have a customer that is not interested in paying $30-$40 a month but would rather buy the software. I have googled and googled all I can come up with are companies that do it as hosted. Does anyone on the list know of software that can transcribe an email/voicemail sent to it and then forward it to the end user? TIA We've talked about this on the VoIP users conference and the feeling is that most likely there is a human on the back end who is doing the transcription, if it is to be at all accurate. Anyone who has worked with the various speech-to-text software, such as ViaVoice or Dragon, knows that training is the key to really accurate transcription. That, and a good quality audio signal. The variations in audio quality you get in voicemail is probably too great to do this all with software only. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKuO3KCFu3bIiwtTARAqWbAJwOhC3REHwQphVpXrG+XfGKwq3ccwCfRoO4 nOv6mEfTV4rQst89YU7/Wpg= =HpnP -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 ___ -- 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 on a Beagleboard?
On Tue, 22 Sep 2009, Lyle Giese wrote: Vincent wrote: Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. 128m of ram 256 m flash for the 'hard drive' is not much in either catagory. And ethernet is a USB addon, not on the board. That's plenty. My systems run happily in just that. (Geode processor) However no Ethernet is going to be an issue! They actually use just 48MB of RAM, but it's nice to have a bit spare... However, I can't see it being hard to get it to work - I understand Asterisk was ported to the Nokia 770's way back - that's also ARM based, using it's Wi-Fi interface. Can't wait to get my Nokia E900 - Arm based Linux tablet with a mobile phone bolted on :) Gordon ___ -- 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] not hearing audio on console/dsp
I am using asterisk 1.4.26.2 and dahdi 2.2.0 in a console/dsp situation. enabling sip debug and rtp debug shows the call coming in, shows Dial(Console/dsp) and rtp traffic is displaying and all that. I just dont hear audio. I can play audio with aplay somefile.wav no problem. so sound is functional. I have used this same configuration before and it worked just fine. What can I look for here to tell me why I dont get sound? rtp shows it coming across, sip connection is there. Thanks for your thoughts. Jerry ___ -- 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] Atcom AG188N as FXO?
On 22/09/09 3:54 AM, Vincent wrote: Hello According to this article, this nice little unit can only use the PSTN port for outgoing calls (ie. as a backup in case the connection to the VoIP provider stops working), but not incoming calls: http://tinyurl.com/mwjmo8 Can someone confirm that Atcom made this strange decision, and that there's no work-around? Might be because of the update, but it reads: the PSTN port is a “passthru” port, allowing you to send some calls to the PSTN and others to your VoIP provider based on the pattern dialed, and letting you receive calls from both the PSTN and your VoIP provider without the need use two separate telephones. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] New Xorcom FXS USB Bank is not loading firmware
Hi, I just got a Xorcom Astribank with 8 FXS but it does not work. So I tried resetting and loading the firmware. But loading just times out. /usr/share/zaptel/xpp_fxloader usb - FIRMWARE LOADING: (usb) [1 devices] 'xpp_fxloader'[11561]: USB Firmware /usr/share/zaptel/USB_FW.hex into /dev/bus/usb/002/003 . I am using asterisk 1.4 and zaptel-1.4.12.1. I deployed quite alot of those devices but I do not see what I am doing wrong with this one. The only difference is that it has 2 USB ports for failover I guess. Best regards, Loïc. -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 ldide...@mixvoip.com http://www.mixvoip.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] RTPAUDIOQOS
On Tuesday 22 September 2009 10:42:44 Johann Steinwendtner wrote: Mindaugas Kezys schrieb: Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified In the given example: *ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txji tter=0.00;txcount=83;rlp=0;rtt=14818.715000* How do I interprete the jitter value ? Is the value 0.020917 good ? Bad ? Is there a unit behind this value ? It's a ratio of out-of-order (jittered) to in-order packets, calculated progressively. Due to the progressive calculation, it's not exactly 3/147, in this case, but it's close enough to know that 3 packets were received out-of-order. The closer the value is to 0, the better. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- 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] New Xorcom FXS USB Bank is not loading firmware
This is the new Astribank2 unit. It will only work with DAHDI 2.2.0 or higher. Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP - Loic Didelot ldide...@mixvoip.com escreveu: Hi, I just got a Xorcom Astribank with 8 FXS but it does not work. So I tried resetting and loading the firmware. But loading just times out. /usr/share/zaptel/xpp_fxloader usb - FIRMWARE LOADING: (usb) [1 devices] 'xpp_fxloader'[11561]: USB Firmware /usr/share/zaptel/USB_FW.hex into /dev/bus/usb/002/003 . I am using asterisk 1.4 and zaptel-1.4.12.1. I deployed quite alot of those devices but I do not see what I am doing wrong with this one. The only difference is that it has 2 USB ports for failover I guess. Best regards, Loïc. -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 ldide...@mixvoip.com http://www.mixvoip.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 ___ -- 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] not hearing audio on console/dsp
Jerry Geis wrote: I am using asterisk 1.4.26.2 and dahdi 2.2.0 in a console/dsp situation. enabling sip debug and rtp debug shows the call coming in, shows Dial(Console/dsp) and rtp traffic is displaying and all that. I just dont hear audio. I can play audio with aplay somefile.wav no problem. so sound is functional. I have used this same configuration before and it worked just fine. What can I look for here to tell me why I dont get sound? rtp shows it coming across, sip connection is there. Thanks for your thoughts. Jerry Even more strange is I setup a call file on the local machine, Channel: Console/dsp Context: dialout Extension: 99 Priority: 1 Application: Playback Data: demo-congrats And I hear the audio just fine for demo-congrats. What can be stopping the audio from the server to the machine I am using when SIP shows the call coming in, rtp shows the data coming across - but no audio is heard. mixer volumes are up. speakers are on. data incoming. Any other ideas? Thanks, Jerry ___ -- 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 Cisco 5300/E1/DSP
Hi I search to know if a company or user use that a Cisco AS5300 with DSP/Voice with Asterisk ? I want use the AS5300 only for the E1/PSTN link in/out thanks Jpc ___ -- 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] New Xorcom FXS USB Bank is not loading firmware
On Tue, Sep 22, 2009 at 02:37:39PM -0300, Vinícius Fontes wrote: This is the new Astribank2 unit. It will only work with DAHDI 2.2.0 or higher. Or with latest Zaptel: Svn snapshot, or the latest Zaptel tarball from http://updates.xorcom.com/astribank/ (which was made using http://svn.asterisk.org/svn/zaptel/branches/1.4/build_tools/zaptel_svn_tarball ) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Problem with dialplan - gotoif ?
On Tue, Sep 22, 2009 at 07:57:56AM -0400, Leif Madsen wrote:
Alex Samad wrote:
4. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pt:ok)
[pbx_config]
i believe i have captured the relevant logging from the console. my problem
is with Gotoif statement
-- Executing [...@dial-sipmnf-sippt-pstn:4] GotoIf(DAHDI/1-1, [BUSY
= CHANUNAVAIL]?pt:ok) in new stack
-- Goto (dial-sipmnf-sippt-pstn,s,5)
from my understanding BUSY != CHANUNAVAIL, therefor it should have jumped
to ok which is s,9.
What have I missed !
You have missed the leading $ in front of the $[...] portion.
Add the $, and things should start working as you expect.
thanks the bloody obvious :)
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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] Sangoma A200 and battery removal detection ??!!!
On Sat, Sep 19, 2009 at 3:33 AM, M Shokuie sena...@gmail.com wrote: Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Regards. -- M. Shokuie Nia. Hello Shokuie, Tzafrir clarified to me that what you were asking for is alarm notification (red alarm / alarm cleared) on battery removal and not hook notifications. Until today, this wasn't implemented in Sangoma Wanpipe drivers, but it was simple enough that I just did the quick fix and this little feature should be available in the next wanpipe release within this week. -- 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] Asterisk on a Beagleboard?
Yes. Using Ubuntu, Asterisk with Dahdi. USB to Ethernet HUB and a Redfone fonebridge T1/E1 gateway connected to it. It can process a T1/E1 worth of calls no problem. - Original Message - From: Vincent vincent.delpo...@bigfoot.com To: asterisk-users@lists.digium.com Sent: Tuesday, September 22, 2009 8:56:55 AM GMT -05:00 US/Canada Eastern Subject: [asterisk-users] Asterisk on a Beagleboard? Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. ___ -- 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
[asterisk-users] 1.6.0.5: I need a really simple analog SendFax dialplan
Using Digium fax I've tried a simple dialplan: '8447' = 1. Answer() [pbx_config] 2. Set(CALLERID(num)=xxxyyy) [pbx_config] 3. Dial(DAHDI/g0/1bbbccc,,G(send))[pbx_config] [send]4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config] 5. HangUp() But I doesn't work. It executes hangup: DAHDI/g0/1bbbccc,,G(send)) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/1bbbccc -- DAHDI/1-1 is proceeding passing it to SIP/173-b55f7448 -- DAHDI/1-1 is ringing -- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448 -- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448 -- DAHDI/1-1 answered SIP/173-b55f7448 -- Executing [8...@outbound-fax:4] SendFAX(SIP/173-b55f7448, /var/spool/asterisk/fax/20090922_1301.tif) in new stack -- Channel 'SIP/173-b55f7448' sending fax '/var/spool/asterisk/fax/20090922_1301.tif' -- Channel 'SIP/173-b55f7448' fax session '16' started -- Executing [8...@outbound-fax:5] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (outbound-fax, 8447, 5) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -- Channel 'SIP/173-b55f7448' fax session '16', [ 000.003512 ], STAT_EVT_STRT_TX st: IDLE rt: IDLENSTX So why does it hangup before completing the fax? Does anyone have a SendFax dialplan that works for an analog channel? Thanks for any help. sean ___ -- 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 on a Beagleboard?
I do not know if fonebridge would work here since it sends/receives the ~2 Mbps (for each circuit/port) of data over ethernet ... constantly. That could choke the USB ... Martin On Tue, Sep 22, 2009 at 5:54 PM, astgro...@comcast.net wrote: Yes. Using Ubuntu, Asterisk with Dahdi. USB to Ethernet HUB and a Redfone fonebridge T1/E1 gateway connected to it. It can process a T1/E1 worth of calls no problem. - Original Message - From: Vincent vincent.delpo...@bigfoot.com To: asterisk-users@lists.digium.com Sent: Tuesday, September 22, 2009 8:56:55 AM GMT -05:00 US/Canada Eastern Subject: [asterisk-users] Asterisk on a Beagleboard? Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. ___ -- 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] 1.6.0.5: I need a really simple analog SendFax dialplan
from RTFM G(context^exten^pri) - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority+1. Optionally, an extension, or extension and context may be specified. Otherwise, the current extension is used. You cannot use any additional action post answer options in conjunction with this option. your priority+1 is Hangup ... is that it ? Martin On Tue, Sep 22, 2009 at 7:32 PM, sean darcy seandar...@gmail.com wrote: Using Digium fax I've tried a simple dialplan: '8447' = 1. Answer() [pbx_config] 2. Set(CALLERID(num)=xxxyyy) [pbx_config] 3. Dial(DAHDI/g0/1bbbccc,,G(send))[pbx_config] [send]4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config] 5. HangUp() But I doesn't work. It executes hangup: DAHDI/g0/1bbbccc,,G(send)) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/1bbbccc -- DAHDI/1-1 is proceeding passing it to SIP/173-b55f7448 -- DAHDI/1-1 is ringing -- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448 -- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448 -- DAHDI/1-1 answered SIP/173-b55f7448 -- Executing [8...@outbound-fax:4] SendFAX(SIP/173-b55f7448, /var/spool/asterisk/fax/20090922_1301.tif) in new stack -- Channel 'SIP/173-b55f7448' sending fax '/var/spool/asterisk/fax/20090922_1301.tif' -- Channel 'SIP/173-b55f7448' fax session '16' started -- Executing [8...@outbound-fax:5] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (outbound-fax, 8447, 5) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -- Channel 'SIP/173-b55f7448' fax session '16', [ 000.003512 ], STAT_EVT_STRT_TX st: IDLE rt: IDLENSTX So why does it hangup before completing the fax? Does anyone have a SendFax dialplan that works for an analog channel? Thanks for any help. sean ___ -- 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
[asterisk-users] (no subject)
___ -- 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] Simple dialplan issue
I have an issue where a particular dialplan works but another doesn't. I'm
not sure why. To me they look identical and it has me stumped.
This works:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 3, Ringing
exten = _X., 4, Dial(SIP/9...@a-test,20,ro)
exten = _X., 5, GotoIf($[${DIALSTATUS} = ANSWER]?9)
exten = _X., 6, GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = _X., 7, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 8, GotoIf($[${DIALSTATUS} = ANSWER]?9)
exten = _X., 9, Hangup
This does NOT work:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 20, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 30, Ringing
exten = _X., 40, Dial(SIP/9...@a-test,20,ro)
exten = _X., 50, GotoIf($[${DIALSTATUS} = ANSWER]?90)
exten = _X., 60, GotoIf($[${DIALSTATUS} = NOANSWER]?70)
exten = _X., 70, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 80, GotoIf($[${DIALSTATUS} = ANSWER]?90)
exten = _X., 90, Hangup
This does NOT work either:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 3, Ringing
exten = _X., 4, Dial(SIP/9...@a-test,20,ro)
exten = _X., 5, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 6, GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = _X., 7, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 8, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 200, Hangup
___
-- 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] International Numbering plan ?
Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc ___ -- 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] Simple dialplan issue
How are these identical?
On Tue, Sep 22, 2009 at 11:32 PM, Julian Yap julianok...@gmail.com wrote:
I have an issue where a particular dialplan works but another doesn't. I'm
not sure why. To me they look identical and it has me stumped.
This works:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 3, Ringing
exten = _X., 4, Dial(SIP/9...@a-test,20,ro)
exten = _X., 5, GotoIf($[${DIALSTATUS} = ANSWER]?9)
exten = _X., 6, GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = _X., 7, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 8, GotoIf($[${DIALSTATUS} = ANSWER]?9)
exten = _X., 9, Hangup
This does NOT work:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 20, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 30, Ringing
exten = _X., 40, Dial(SIP/9...@a-test,20,ro)
exten = _X., 50, GotoIf($[${DIALSTATUS} = ANSWER]?90)
exten = _X., 60, GotoIf($[${DIALSTATUS} = NOANSWER]?70)
exten = _X., 70, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 80, GotoIf($[${DIALSTATUS} = ANSWER]?90)
exten = _X., 90, Hangup
This does NOT work either:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 3, Ringing
exten = _X., 4, Dial(SIP/9...@a-test,20,ro)
exten = _X., 5, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 6, GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = _X., 7, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 8, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 200, Hangup
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International Numbering plan ?
On 23/09/09 4:19 PM, Phibee Network Operation Center wrote: Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? I'm not sure you understand the scale of what you're asking, but anyways. Here's a start: http://www.itu.int/oth/T0202.aspx?parent=T0202 Bear in mind that these numbers change reasonably regularly. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple dialplan issue
On 23/09/09 3:32 PM, Julian Yap wrote:
I have an issue where a particular dialplan works but another doesn't.
I'm not sure why. To me they look identical and it has me stumped.
This works:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
1, 2, yep.
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 20, Monitor(wav,/tmp/test-${UNIQUEID},mb)
Normally 2 comes after 1 rather than 20 - looks like you're missing 2
through 19 here :)
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
___
-- 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] International Numbering plan ?
On Wed, 23 Sep 2009 16:19:26 Phibee Network Operation Center wrote: Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc Country numbering plan can be easily found. Anything finer then that and you will need to pay. ___ -- 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] International Numbering plan ?
The URL is a good start but for some large countries which I have worked for, the list misses some important information like inter-city, inter-state, inter-city mobile and local mobile and IDD. To me, nothing can replace local intelligence. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Wednesday, 23 September 2009 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] International Numbering plan ? On 23/09/09 4:19 PM, Phibee Network Operation Center wrote: Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? I'm not sure you understand the scale of what you're asking, but anyways. Here's a start: http://www.itu.int/oth/T0202.aspx?parent=T0202 Bear in mind that these numbers change reasonably regularly. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Simple dialplan issue
On Tue, Sep 22, 2009 at 6:34 PM, Matt Riddell li...@venturevoip.com wrote:
On 23/09/09 3:32 PM, Julian Yap wrote:
I have an issue where a particular dialplan works but another doesn't.
I'm not sure why. To me they look identical and it has me stumped.
This works:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
1, 2, yep.
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 20, Monitor(wav,/tmp/test-${UNIQUEID},mb)
Normally 2 comes after 1 rather than 20 - looks like you're missing 2
through 19 here :)
Hmm, I guess I was under the understanding that it would work like that.
Why wouldn't this work?:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 3, Ringing
exten = _X., 4, Dial(SIP/9...@a-test,20,ro)
exten = _X., 5, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 6, GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = _X., 7, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 8, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 200, Hangup
___
-- 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] Bringing people into a conference
G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO from voip-info is trying to do, but it doesn't work quite properly for me. Here's what happens: 1. Internal person A calls person B 2. Person A presses *0, he is given a dial tone and person B is taken to a conference room 3. Person A calls person C and they can talk, and then person A presses **. 4. Person C is brought to the conference room, but person A is disconnected. In the last step, A should be taken to the conference room as well. Or at least, that's what I think should happen. Here's the relevant logs, where 230 is person A, 231 is person B, 207 is person C, and 282 is the conference room: http://dpaste.com/hold/97072/. Thanks, Harley Would you like total visibility and control over use of Web 2.0 applications such as media streaming, gaming and instant messaging at your company? If so, see: http://netboxblue.com/products/advancedidsips Scanned by the Netbox from Netbox Blue (http://netboxblue.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] Simple dialplan issue
On 23/09/09 4:59 PM, Julian Yap wrote:
Why wouldn't this work?:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${
UNIQUEID},mb)
exten = _X., 3, Ringing
exten = _X., 4, Dial(SIP/9...@a-test,20,ro)
exten = _X., 5, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 6, GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = _X., 7, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 8, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 200, Hangup
1. works
2. should work
3. works
4. r option is horrible, but works
5. works
6. weird but works - it's going to go to 7 anyway, so effectively this
line does nothing
7. again, r is horrible but works
8. Why? If it get's to 9 (which doesn't exist) it's going to hang up
anyway.
200. Pointless (see 8)
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
___
-- 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] International Numbering plan ?
On 23/09/09 4:39 PM, Michael wrote: On Wed, 23 Sep 2009 16:19:26 Phibee Network Operation Center wrote: Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc Country numbering plan can be easily found. Anything finer then that and you will need to pay. That link I provided is correct at least for New Zealand cities etc -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Bringing people into a conference
On 23/09/09 5:07 PM, Harley Holcombe wrote: G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO from voip-info is trying to do, but it doesn't work quite properly for me. Here's what happens: 1. Internal person A calls person B 2. Person A presses *0, he is given a dial tone and person B is taken to a conference room 3. Person A calls person C and they can talk, and then person A presses **. 4. Person C is brought to the conference room, but person A is disconnected. Why doesn't A just call the number they've been transferring people to? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Simple dialplan issue
Thanks all, I worked this out with your help. ___ -- 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] Bringing people into a conference
They could, but the less steps salespeople have to go through the better. It's also a bit jarring to be disconnected as soon as you transfer someone into a meeting room you're already meant to be in. - Harley From: Matt Riddell li...@venturevoip.com To: asterisk-users@lists.digium.com Date: 23/09/2009 03:37 PM Subject: Re: [asterisk-users] Bringing people into a conference Sent by: asterisk-users-boun...@lists.digium.com On 23/09/09 5:07 PM, Harley Holcombe wrote: G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO from voip-info is trying to do, but it doesn't work quite properly for me. Here's what happens: 1. Internal person A calls person B 2. Person A presses *0, he is given a dial tone and person B is taken to a conference room 3. Person A calls person C and they can talk, and then person A presses **. 4. Person C is brought to the conference room, but person A is disconnected. Why doesn't A just call the number they've been transferring people to? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 Would you like total visibility and control over use of Web 2.0 applications such as media streaming, gaming and instant messaging at your company? If so, see: http://netboxblue.com/products/advancedidsips Scanned by the Netbox from Netbox Blue (http://netboxblue.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
