[asterisk-users] Unknown calls
Hi there everyone, I am a bit confused these days due to some problem I am having. Its not a technical problem. Asterisk is working fine. Most of the users are happy, but some handful of users are getting calls in the middle of the night even though they have enabled Anonymous Call Rejection (blocks calls with no caller id on asterisk server) and TIMED DO NOT DISTURB which also blocks calls unconditionally from 11pm to 6 am. Now the seems to have make it through to the user still. The caller id of the call is Asterisk Unknown. about six users are getting this call only at night time. Asterisk server has no record of this call in log file or cdr. I have also blocked all incoming calls coming from unknown ip addresses etc. Still last night there was a call to a customer. Plz help me figure out the solution for this problem. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown calls
On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote: Still last night there was a call to a customer. Plz help me figure out the solution for this problem. Can you be sure that the call _is_ coming through your Asterisk server, rather than being the result of random scanning for your customers' phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extend the timout on ringing for pri or sip
use the timeout option in the Dial application like so Dial(SIP/trunk,120) If you dont specify the timeout the default timeout used bya sterisk is probably more than 60 seconds. On Wed, Feb 23, 2011 at 3:17 PM, Israel Gottlieb isr...@gmail.com wrote: Hi Does anyone know how i could extend the timer for the ringing time on a pri or sip trunk ? Today the call gets a cancel request after a minute if not answerd yet is it on asterisk or is a provider side setting? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown calls
Thats what im unsure about. I think the calls maybe going to the user directly through sip uri or some other method. How can i test that. I have already tried to call those customers with direct sip uri dial but does not work. On Thu, Feb 24, 2011 at 3:19 PM, Roger Burton West ro...@firedrake.orgwrote: On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote: Still last night there was a call to a customer. Plz help me figure out the solution for this problem. Can you be sure that the call _is_ coming through your Asterisk server, rather than being the result of random scanning for your customers' phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in dialing out
try this http://www.voip-info.org/wiki/view/Asterisk+sip+qualify On Sat, Feb 19, 2011 at 5:00 AM, asterisk asterisk aster...@ck-lee.comwrote: I have a sip trunk connecting to a huawei softx3000. At the moment, I can register and dial in. However, peer status shows not reachable sip show peer as follow * Name : cmphone Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-cmphone Subscr.Cont. : device-hints Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes Outb. proxy : 202.0.179.3 DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 202.0.179.3 Addr-IP : 202.0.179.3:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 852350xx SIP Options : 100rel Codecs : 0xe (gsm|ulaw|alaw) Codec Order : (alaw:20,ulaw:20,gsm:20) Auto-Framing : No 100 on REG : No Status : UNREACHABLE Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No In sip.conf I have register = 852350x:secret@202.0.179.3 [cmphone] type = friend host = 202.0.179.3 secret = secret username = 852350x context = from-cmphone dtmfmode = rfc2833 outboundproxy = 202.0.179.3 caninvite=no insecure = port,invite nat = yes When debug is on, the error message is --- SIP read from UDP:202.0.179.3:5060 --- SIP/2.0 504 Server Time-out From: asterisk sip:aster...@sip.x.xxx;tag=as2d14b9ec To: sip:202.0.179.3;tag=6b0704d0 CSeq: 102 OPTIONS Call-ID: 17e0315c21d7dbc10e8c185740e21...@sip.x.xxx Via: SIP/2.0/UDP 14.xxx.xxx.xxx:5060;branch=z9hG4bK3646eaf2;received=14.xxx.xxx.xxx;rport=5060 Content-Length: 0 Any help is appreciate. CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [1.4.39.2] Simple AGI doesn't reply
Hello
The following, dead simple Bash script ran as AGI doesn't reply to
Asterisk:
= extensions.conf
[from_fxo]
exten = s,1,Wait(2)
exten = s,n,Set(CID=${CALLERID(num)})
exten = s,n,AGI(/var/tmp/basic.agi)
exten = s,n,Hangup()
= /var/tmp/basic.agi
#!/bin/bash
#Ripped from
#http://lists.digium.com/pipermail/asterisk-users/2003-July/008554.html
while read -e ARG [ $ARG ] ; do
done
echo NOOP Here
while read line
do
= CLI
centos*CLI
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from_fxo:1] Wait(DAHDI/1-1, 2) in new stack
-- Executing [s@from_fxo:2] Set(DAHDI/1-1, CID=123456) in new
stack
-- Executing [s@from_fxo:3] AGI(DAHDI/1-1, /var/tmp/basic.agi)
in new stack
-- Launched AGI Script /var/tmp/basic.agi
AGI Tx agi_request: /var/tmp/basic.agi
AGI Tx agi_channel: DAHDI/1-1
AGI Tx agi_language: en
AGI Tx agi_type: DAHDI
AGI Tx agi_uniqueid: 1298544498.10
AGI Tx agi_callerid: 123456
AGI Tx agi_calleridname: unknown
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: unknown
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: from_fxo
AGI Tx agi_extension: s
AGI Tx agi_priority: 3
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx
-- AGI Script /var/tmp/basic.agi completed, returning 0
-- Executing [s@from_fxo:4] Hangup(DAHDI/1-1, ) in new stack
== Spawn extension (from_fxo, s, 4) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
=
As you can see, the AGI script doesn't reply to Asterisk: Does CentOS
and/or Asterisk require some tweaking so that an AGI script can read
data from stdin?
FWIW, before using this Bash script, I could successfully run
Asterisk's agi-test.agi successfully.
Thank you.
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Re: [asterisk-users] extend the timout on ringing for pri or sip
sorry i wasnt clear enough i meen inbound On Thu, Feb 24, 2011 at 12:25 PM, Rizwan Hisham rizwanhas...@gmail.comwrote: use the timeout option in the Dial application like so Dial(SIP/trunk,120) If you dont specify the timeout the default timeout used bya sterisk is probably more than 60 seconds. On Wed, Feb 23, 2011 at 3:17 PM, Israel Gottlieb isr...@gmail.com wrote: Hi Does anyone know how i could extend the timer for the ringing time on a pri or sip trunk ? Today the call gets a cancel request after a minute if not answerd yet is it on asterisk or is a provider side setting? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Carrying context from one server to another?
The relevant part of my setup is something like: SIP phones - local server - remote server - SIP-to-PSTN provider I want _some_ of the SIP phones on the local server to be able to get access to SIP-to-PSTN, but not all of them. The local-to-remote connection is IAX2 over VPN. Do I need to set up two separate IAX2 connections, one privileged and the other not, or can I somehow tag calls from some phones on the local server so that they're noted as privileged on the remote server? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4.39.2] Simple AGI doesn't reply
On Thu, 24 Feb 2011 11:56:25 +0100, Gilles codecompl...@free.fr wrote: The following, dead simple Bash script ran as AGI doesn't reply to Asterisk: Turns out Bash doesn't allow empty loops. This version does reply as expected: == #!/bin/bash read line while [[ $line != ]] ; do read line done echo NOOP Here read line exit 0 == HTH, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [1.4] Still can't get it to call back
Hello
No matter what I try, Asterisk still fails dialing back through a
callfile built through an AGI script.
The whole thing works fine when the original call that triggers
Asterisk is from an internal extension (Xlite), but it fails when it's
from my cellphone ringing through the FXO/Zaptel port and I want to
wait a few seconds and call back through the FXO/Zaptel.
Could it that even though the Zap channel is dead when calling the
script in the h extension, for some reason, the channel is still in
use at that point, so even an AGI script won't be able to make an
outgoing call through the FXO?
Here are the relevant parts:
extensions.conf
[internal]
;Call from Xlite to simulate incoming external call
exten = ,1,Dial(Local/s@from_fxo)
[from_fxo]
;Incoming call just to trigger things
exten = s,1,Wait(2)
exten = s,n,Set(CID=${CALLERID(num)})
exten = s,n,Wait(2)
exten = s,n,Hangup
;Call script to build callfile
exten = h,1,DeadAGI(/var/tmp/callback.lua,${CID})
;Context used by callfile
[callback]
;Zaptel doesn't support call progress, so just wait 10s
exten = start,1,Wait(10)
exten = start,n,Answer()
exten = start,n,Playback(tt-monkeysintro)
exten = start,n,Hangup()
AGI script
#!/var/tmp/lua
--Must first empty stdin
while true do
local line = io.read()
if line == then break end
-- Without line below, script never ends
io.write(NOOP ,line,\n)
end
file = io.open (/var/tmp/callback.call,w)
channel = string.format(Channel: Zap/1/123456\n)
file:write(channel)
file:write(Context: callback\n)
file:write(Extension: start\n)
file:close()
os.execute(mv /var/tmp/callback.call /var/tmp/asterisk/outgoing)
CLI output
ip04*CLI
-- Starting simple switch on 'Zap/1-1'
-- Executing [s@from_fxo:1] Wait(Zap/1-1, 2) in new stack
-- Executing [s@from_fxo:2] Set(Zap/1-1, CID=123456) in new
stack
-- Executing [s@from_fxo:3] Wait(Zap/1-1, 2) in new stack
-- Executing [s@from_fxo:4] Hangup(Zap/1-1, ) in new stack
== Spawn extension (from_fxo, s, 4) exited non-zero on 'Zap/1-1'
-- Executing [h@from_fxo:1] DeadAGI(Zap/1-1,
/var/tmp/callback.lua|123456) in new stack
-- Launched AGI Script /var/tmp/test7.lua
AGI Tx agi_request: /var/tmp/test7.lua
AGI Tx agi_channel: Zap/1-1
AGI Tx agi_language: en
AGI Tx agi_type: Zap
AGI Tx agi_uniqueid: 1298539792.116
AGI Tx agi_callerid: 123456
AGI Tx agi_calleridname: unknown
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: unknown
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: from_fxo
AGI Tx agi_extension: h
AGI Tx agi_priority: 1
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx
AGI Rx NOOP agi_request: /var/tmp/test7.lua
AGI Tx 200 result=0
AGI Rx NOOP agi_channel: Zap/1-1
AGI Tx 200 result=0
AGI Rx NOOP agi_language: en
AGI Tx 200 result=0
AGI Rx NOOP agi_type: Zap
AGI Tx 200 result=0
AGI Rx NOOP agi_uniqueid: 1298539792.116
AGI Tx 200 result=0
AGI Rx NOOP agi_callerid: 123456
AGI Tx 200 result=0
AGI Rx NOOP agi_calleridname: unknown
AGI Tx 200 result=0
AGI Rx NOOP agi_callingpres: 0
AGI Tx 200 result=0
AGI Rx NOOP agi_callingani2: 0
AGI Tx 200 result=0
AGI Rx NOOP agi_callington: 0
AGI Tx 200 result=0
AGI Rx NOOP agi_callingtns: 0
AGI Tx 200 result=0
AGI Rx NOOP agi_dnid: unknown
AGI Tx 200 result=0
AGI Rx NOOP agi_rdnis: unknown
AGI Tx 200 result=0
AGI Rx NOOP agi_context: from_fxo
AGI Tx 200 result=0
AGI Rx NOOP agi_extension: h
AGI Tx 200 result=0
AGI Rx NOOP agi_priority: 1
AGI Tx 200 result=0
AGI Rx NOOP agi_enhanced: 0.0
AGI Tx 200 result=0
AGI Rx NOOP agi_accountcode:
AGI Tx 200 result=0
-- AGI Script /var/tmp/callback.lua completed, returning 0
-- Hungup 'Zap/1-1'
-- Attempting call on Zap/1/123456 for start@callback:1 (Retry 1)
[Feb 24 09:29:58] NOTICE[6097]: channel.c:2863 __ast_request_and_dial:
Unable to request channel Zap/1/123456
[Feb 24 09:29:58] NOTICE[6097]: pbx_spool.c:341 attempt_thread: Call
failed to go through, reason (0) Call Failure (not BUSY, and not
NO_ANSWER, maybe Circuit busy or down?)
Thank you for any help.
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Re: [asterisk-users] [1.4] Still can't get it to call back
In article jg9cm6pqkit0q3oi5aacabi7dfql7st...@4ax.com, Gilles codecompl...@free.fr wrote: Hello No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. The whole thing works fine when the original call that triggers Asterisk is from an internal extension (Xlite), but it fails when it's from my cellphone ringing through the FXO/Zaptel port and I want to wait a few seconds and call back through the FXO/Zaptel. Could it that even though the Zap channel is dead when calling the script in the h extension, for some reason, the channel is still in use at that point, so even an AGI script won't be able to make an outgoing call through the FXO? Yes, that is the reason. The easiest thing is probably to put in a delay if os.execute allows full shell syntax: os.execute(mv /var/tmp/callback.call /var/tmp/asterisk/outgoing) os.execute((sleep 2;mv /var/tmp/callback.call /var/tmp/asterisk/outgoing)) Alternatively, you can set the mtime of the file you created in /var/tmp to two or three seconds in the future - does lua have something like file.utime or os.utime? The pbx spooler in Asterisk will not execute call files with an mtime still in the future. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Registration failed though configured.
Hi list, Currently, one of my phones registers fine, and the other does not, though for me they have the same config... Can somebody help debug/understand why? The logs in asterisk say: [Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642 handle_request_register: Registration from 'IMSI208300618462231 sip:IMSI20830061@127.0.0.1' failed for '127.0.0.1' - No matching peer found Thanks in /etc/asterisk/extensions.conf: exten = 2102,1,Macro(dialSIP,IMSI2081) ; this one registers ok exten = 2111,1,Macro(dialSIP,IMSI20830061) ; fails In sip.conf: [IMSI2081] ; callerid=2102 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic [IMSI20830061] callerid=2111 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Assigning an extension to a roaming phone
So you have an IP network, with SIP agents (cell phones ?), some of those are manually setup in you sip.conf file, but you want to allow unknown cell phones users to self register in your system ? Yes, exactly. Someone enter your network, dial 3001@your ipbx and get/set a temporary internal number. Then other phone can dial his ? Yes, right. I don't think it's possible, although ... What you need is to mimic the SIP registration process, by fetching the following informations from during the setup call: * IP of the phone * UDP/TCP Port of the SIP process * Some SIP user ID Then you store thoses in your DB in the form SIP/user@IP:port and then you could be able to Dial this string, (if the phone is ok to be dialed by an unknown party this way) Mmm. yes, but I don't have a clue how I could do that. Perhaps also if I were able to retrieve the IMSI of the roaming user I might be able to work out something. But I don't know how to get it... -- Axelle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Assigning an extension to a roaming phone
Hi Danny, That's a nice log I'll try and do the same with a higher verbosity level on my side too. Just to make sure - who called 3001? the roaming phone that had no extension yet? -- Executing [3001@default:1] Verbose(SIP/sipuser-006f, Create roaming extension) in new stack - when you called 4144 (from another phone), it triggers 144 - which I understand - but did that 144 actually have the roaming phone ring? -- Executing [4144@default:2] Set(SIP/sipuser-0070, ROAMEXT=144) in new stack Thanks Axelle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP (voice) issue. STUN server
Hi,all I configured two Asterisk PBXs: 1.4.X and 1.6.X. All relevant ports are opened, externip is configured in sip.conf. I think, that all relevant configurations are checked. But, no voice hear between local and remote extension. What I need to check, configure in router and PBX for resolving this issue ? How I can to install and configure STUN server ? Thanks, Oleg . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Carrying context from one server to another?
On Thu, Feb 24, 2011 at 11:38:17AM +, Roger Burton West wrote: The relevant part of my setup is something like: SIP phones - local server - remote server - SIP-to-PSTN provider I want _some_ of the SIP phones on the local server to be able to get access to SIP-to-PSTN, but not all of them. The local-to-remote connection is IAX2 over VPN. The way I would to this is by blocking them on the localserver (with different contexts). An other solution would be to set prefixes on the extension when dialing from local to remote and use these to filter, not very elegant but works over any transport. I use this to do multitenant billing on the remote server in places where I only want 1 IAX trunk. Whether this is effective depends on your control of the local server. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP (voice) issue. STUN server
Try something like this, [general] localnet=192.168.0.0/255.255.0.0 ; or your subnet externip=x.x.x.x ; use your address [YOURREMOTEPEER] ; your peer's name nat=yes qualify=yes; Force keepalives On Thu, Feb 24, 2011 at 7:12 PM, Oleg Botvinkin ole...@gmail.com wrote: Hi,all I configured two Asterisk PBXs: 1.4.X and 1.6.X. All relevant ports are opened, externip is configured in sip.conf. I think, that all relevant configurations are checked. But, no voice hear between local and remote extension. What I need to check, configure in router and PBX for resolving this issue ? How I can to install and configure STUN server ? Thanks, Oleg . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown calls
Do you have PRI card or FXO card? -- Sent from my iPhone On Feb 24, 2011, at 5:28 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Thats what im unsure about. I think the calls maybe going to the user directly through sip uri or some other method. How can i test that. I have already tried to call those customers with direct sip uri dial but does not work. On Thu, Feb 24, 2011 at 3:19 PM, Roger Burton West ro...@firedrake.org wrote: On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote: Still last night there was a call to a customer. Plz help me figure out the solution for this problem. Can you be sure that the call _is_ coming through your Asterisk server, rather than being the result of random scanning for your customers' phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown calls
Its a pure VoIP setup. no cards. On Thu, Feb 24, 2011 at 7:12 PM, Satish Patel satish...@hotmail.com wrote: Do you have PRI card or FXO card? -- Sent from my iPhone On Feb 24, 2011, at 5:28 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Thats what im unsure about. I think the calls maybe going to the user directly through sip uri or some other method. How can i test that. I have already tried to call those customers with direct sip uri dial but does not work. On Thu, Feb 24, 2011 at 3:19 PM, Roger Burton West ro...@firedrake.org ro...@firedrake.org wrote: On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote: Still last night there was a call to a customer. Plz help me figure out the solution for this problem. Can you be sure that the call _is_ coming through your Asterisk server, rather than being the result of random scanning for your customers' phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.comrizwanhas...@gmail.com W: http://www.axvoice.com/www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice outbound Caller ID broken
Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk caller ID
We are getting a lot of calls identified as Asterisk or out of area in the middle of the night. From other posts on the list, I have assumed these are null Caller ID calls and Asterisk is plugging in pseudo ID. Is that correct? It seems to me that Asterisk should simply say no caller ID or No ID or something besides Asterisk. In any case, we are trying to filter them with little success. When we do a LEN(CALLERID(num) we get 13, when we expect 10 The call pattern is 1 call followed by a second abut 1 minute later followed by 1 about 10 minutes later. Does anyone have any ideas to contribute? Thanks Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice outbound Caller ID broken
What kind of broken are you seeing. It could be the ID is pseudo ID and may never reflect the actual caller. CF _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle Sent: Thursday, February 24, 2011 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Google Voice outbound Caller ID broken Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice outbound Caller ID broken
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle
Sent: Thursday, February 24, 2011 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Google Voice outbound Caller ID broken
Anybody else noticed that caller id for outbound calls via Google Voice seems
to be broken? It seems to be a Google Voice problem though, not an asterisk
issue.
Yes.. google it
This is what I have done to resolve it (I posted a few days ago on this)
exten = _9NXXNXX,1,Dial(gtalk/(value in
gtalk.conf)/+1(googlevoice#)@voice.google.com,30,D(ww2www${EXTEN:1}#w))
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Re: [asterisk-users] Google Voice outbound Caller ID broken
On Thu, Feb 24, 2011 at 9:08 AM, William Stillwell
will...@stillwellsoft.com wrote:
Yes.. google it
I did. :)
This is what I have done to resolve it (I posted a few days ago on this)
exten = _9NXXNXX,1,Dial(gtalk/(value in gtalk.conf)/+1(googlevoice#)@
voice.google.com,30,D(ww2www${EXTEN:1}#w))
I must have missed that posting. I'll go back and dig it up. Thanks.
--
Chris
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Re: [asterisk-users] [1.4] Still can't get it to call back
On Thu, 24 Feb 2011, Gilles wrote: No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. I don't think it has anything to do with the method used to create the call file. AGI script #!/var/tmp/lua --Must first empty stdin while true do local line = io.read() if line == then break end -- Without line below, script never ends io.write(NOOP ,line,\n) end This script violates the AGI protocol. In addition to suggesting to use an established library, I'd suggest picking a language and sticking with it. Personally, I use C because it's the sharpest tool in my toolbox. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Carrying context from one server to another?
you can also set some kind of authentication on the extensions for example ask for a pin to dialout. etc On Thu, Feb 24, 2011 at 6:51 PM, Daniel Tryba dan...@tryba.nl wrote: On Thu, Feb 24, 2011 at 11:38:17AM +, Roger Burton West wrote: The relevant part of my setup is something like: SIP phones - local server - remote server - SIP-to-PSTN provider I want _some_ of the SIP phones on the local server to be able to get access to SIP-to-PSTN, but not all of them. The local-to-remote connection is IAX2 over VPN. The way I would to this is by blocking them on the localserver (with different contexts). An other solution would be to set prefixes on the extension when dialing from local to remote and use these to filter, not very elegant but works over any transport. I use this to do multitenant billing on the remote server in places where I only want 1 IAX trunk. Whether this is effective depends on your control of the local server. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4.39.2] Simple AGI doesn't reply
On Thu, 24 Feb 2011, Gilles wrote: = /var/tmp/basic.agi #!/bin/bash #Ripped from #http://lists.digium.com/pipermail/asterisk-users/2003-July/008554.html while read -e ARG [ $ARG ] ; do done echo NOOP Here while read line do On Thu, 24 Feb 2011, Gilles wrote: Turns out Bash doesn't allow empty loops. Bash has a thing about syntax too. Note you're not 'done' with your second loop. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Debug Dropped Audio
Hi List, We have 3 Asterisk servers (1.4 1.6) with about 200 users all connecting over the internet. Our biggest problem is with dropped audio. My question is what is the best way to debug this? Searching on the internet does not turn up a lot of results for dropped audio. It seems most people, when they have problems, the problems are related to quality but our quality is pretty good. What happens with us is at some random point in the conversation the audio just disappears. Whats the best way to start tracking this down? -- Jesse Cloutier Network Administrator Cronomagic Canada 5143411579 x210 je...@cronomagic.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice outbound Caller ID broken
Chris, Can you please provide more details. What do you exactly mean by broken? Do your call recipients get a random CID? Have you tried to call from the GMail WEB interface? Are you getting the same result? -Vladimir On 2/24/2011 8:51 AM, Chris Gentle wrote: Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI FullyBooted issue
On Feb 23, 2011, at 4:39 AM, Ishfaq Malik wrote: Hi We're still testing out asterisk 1.8 (using 1.8.2.2 from rpm package) before putting it into production and I'm observing an odd issue when using the AMI Every request I send to the AMI just results in a FullyBooted response rather than the expected response. Here are some examples from my logs -- Call started: 22/02/2011 11:34:03 -- action: command command: core show channels Event: FullyBooted Privilege: system,all SequenceNumber: 1706 File: manager.c Line: 2937 Func: action_login Status: Fully Booted -- Call started: 22/02/2011 10:28:15 -- action: command command: sip show peers Event: FullyBooted Privilege: system,all SequenceNumber: 1610 File: manager.c Line: 2937 Func: action_login Status: Fully Booted Has anyone else experienced anything like this? Is this a new AMI connection for each command? If so, that is the problem. Whenever a connection is made, Asterisk informs the connector that it is FullyBooted and it is safe to start sending commands. If it didn't do this, if multiple machines are making a new connection, only one of them would get notified. The proper thing to do would be to maintain an open connection for sending commands, or to expect the command upon connection and only send after that. You are probably getting the message you expect after the FullyBooted response, so this would be a parsing error on your part. Assign an ActionID to the event you send and the response you are looking for will have that ActionID in it as well. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIAL through Specific number in PRI
On Thu, Feb 24, 2011 at 1:31 AM, Faisal Hanif fai...@vopium.com wrote:
PRI start from 3055 to 30550100 i have purchased a 100 number from
telco and our pilot number is 3055, now if some caller want to dial any
number but caller should shown is 30550008 like this.
Set the ${CALLERID(num)} variable before you make your outbound call.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] extend the timout on ringing for pri or sip
On Thu, Feb 24, 2011 at 5:01 AM, Israel Gottlieb isr...@gmail.com wrote: sorry i wasnt clear enough i meen inbound You could always Answer() the call in your dialplan before you do anything else, then Dial() whoever you're trying to reach and set your own timeouts there. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration failed though configured.
On Thu, Feb 24, 2011 at 7:24 AM, Axelle aaforti...@gmail.com wrote: Hi list, snip in /etc/asterisk/extensions.conf: exten = 2102,1,Macro(dialSIP,IMSI2081) ; this one registers ok exten = 2111,1,Macro(dialSIP,IMSI20830061) ; fails These lines have nothing to do with endpoint registration. These are outbound dial Macro's. In sip.conf: [IMSI2081] ; callerid=2102 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic [IMSI20830061] callerid=2111 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic Do these IMSI names / numbers match what your phone is trying to register as? Are there actual at the end of the numbers, or are you attempting to obfuscate? Show us the actual logs and the actual sip.conf entries with only passwords removed, leave the usernames intact, and maybe we'll be able to see what's actually happening. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Assigning an extension to a roaming phone
On Thu, Feb 24, 2011 at 7:31 AM, Axelle aaforti...@gmail.com wrote:
Hi Danny,
Try this code, it uses the SIPCHANINFO function to get the peername of the
device that's attempting to create the roaming extension, instead of the
callerid. Basically, you need to store some kind of contact info for the
roaming phone for it to be any use at all. If you don't have a peername for
the phone, you could try SIPCHANINFO(peerip) instead.
[roaming-ext]
;Create a new roaming extension
exten = 3001,1(readop),Verbose(Create roaming extension)
exten = 3001,n,Read(digito,beep,3)
exten = 3001,n,Playback(you-entered)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Verbose(Setting roaming extension 4${digito} to call
${SIPCHANINFO(peername)})
exten = 3001,n,Set(DB(roam/${digito})=${SIPCHANINFO(peername)})
exten = 3001,n,Playback(vm-goodbye)
exten = 3001,n,Hangup()
;Dial a roaming extension
exten = _4XXX,1,Verbose(Calling roaming extension ${EXTEN})
exten = _4XXX,n,Set(ROAMEXT=${DB(roam/${EXTEN:1})})
exten = _4XXX,n,Dial(SIP/${ROAMEXT},30)
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] AMI FullyBooted issue
Hi, I have this same behaviour on version 1.8.2.3 build from source. We are using AMI to originate call from our CRM software, but we ignore that message. The patch for the bug at https://issues.asterisk.org/view.php?id=18168 has been committed (thanks FeyFre!). The FullyBooted event will no longer be broadcast to every connection each time someone connects. It will still be sent to the individual connections upon successful authentication. AMI applications should all really listen for the FullyBooted event before doing anything of consequence since it is possible to connect to AMI and send a command before the module that would normally handle the command is loaded. This would result in missed commands. Terry-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debug Dropped Audio
On Thu, Feb 24, 2011 at 10:32 AM, Jesse Cloutier je...@cronomagic.comwrote: Whats the best way to start tracking this down? Collect proper debug information[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paging with Polycom 3.3.x
Hi, My phones stopped auto-answering when being paged, since I moved on to Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk 1.6.2.16. I looked at the wiki but nothing I try there works, even if I cut and paste the same setup. Any one has any idea of what I should change from my old 3.2.3 setup? My older phone (501) still using 3.1.6 still auto-answer correctly. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
If you compare a working config with a non-working you will see something with the answer type. I had that issue until I down rev'd. Look for something like Ring Answer, I forget the exact details now. From: asterisk-users-boun...@lists.digium.com on behalf of Mike Sent: Thu 2/24/2011 1:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Paging with Polycom 3.3.x Hi, My phones stopped auto-answering when being paged, since I moved on to Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk 1.6.2.16. I looked at the wiki but nothing I try there works, even if I cut and paste the same setup. Any one has any idea of what I should change from my old 3.2.3 setup? My older phone (501) still using 3.1.6 still auto-answer correctly. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Thursday, February 24, 2011 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging with Polycom 3.3.x If you compare a working config with a non-working you will see something with the answer type. I had that issue until I down rev'd. Look for something like Ring Answer, I forget the exact details now. _ From: asterisk-users-boun...@lists.digium.com on behalf of Mike Sent: Thu 2/24/2011 1:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Paging with Polycom 3.3.x Hi, My phones stopped auto-answering when being paged, since I moved on to Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk 1.6.2.16. I looked at the wiki but nothing I try there works, even if I cut and paste the same setup. Any one has any idea of what I should change from my old 3.2.3 setup? My older phone (501) still using 3.1.6 still auto-answer correctly. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
From: asterisk-users-boun...@lists.digium.com on behalf of Mike Sent: Thu 2/24/2011 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike Looking for it now -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
On Thu, Feb 24, 2011 at 1:41 PM, Mike l...@net-wall.com wrote: Hi, My phones stopped auto-answering when being paged, since I moved on to Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk 1.6.2.16. I looked at the wiki but nothing I try there works, even if I cut and paste the same setup. Any one has any idea of what I should change from my old 3.2.3 setup? My older phone (501) still using 3.1.6 still auto-answer correctly. Polycom changed some of the config file options as outlined in the UC Software upgrade guide. I am using the following for paging. voIpProt.SIP voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.3.class=ringAutoAnswer voIpProt.SIP.alertInfo.3.value=Ring Answer /voIpProt.SIP.alertInfo /voIpProt.SIP /voIpProt Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
From: asterisk-users-boun...@lists.digium.com on behalf of Mike Sent: Thu 2/24/2011 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=2000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ where the timeout is the ampount of time on milliseconds before it goes to speaker. These values are in the sip.cfg, so in your server it may be sip_316.cfg. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
Thank you Terry and Ryan, I will try those things and see if I can find my problem. Will definitely come back with my solution in case it helps somebody else. Mike Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=2000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ where the timeout is the ampount of time on milliseconds before it goes to speaker. These values are in the sip.cfg, so in your server it may be sip_316.cfg. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
Is 3.3.x downloadable for non-paying people yet? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, February 24, 2011 2:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Thank you Terry and Ryan, I will try those things and see if I can find my problem. Will definitely come back with my solution in case it helps somebody else. Mike Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=2000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ where the timeout is the ampount of time on milliseconds before it goes to speaker. These values are in the sip.cfg, so in your server it may be sip_316.cfg. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
Polycom are at 3.3.1 now, so 3.3.0 should be fair game. It has nothing to do with paying or not, the company that sold you the phone should be able to give you the latest version no? Unless you bought from a guy who found a box that fell off a truck.or some third-rate reseller. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 24, 2011 3:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Is 3.3.x downloadable for non-paying people yet? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, February 24, 2011 2:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Thank you Terry and Ryan, I will try those things and see if I can find my problem. Will definitely come back with my solution in case it helps somebody else. Mike Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=2000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ where the timeout is the ampount of time on milliseconds before it goes to speaker. These values are in the sip.cfg, so in your server it may be sip_316.cfg. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extensions.lua with luasql.mysql.
Hi to all!
I'm trying to create a context for integration with extensions.lua and
libsql.mysql, but I'm not getting to run. When I reload the module
pbx_lua.so the following error appears:
[Feb 24 16:59:29] ERROR[30749]: pbx_lua.c:1249 exec: Error executing lua
extension: error loading module 'luasql.mysql' from file
'/usr/lib/lua/5.1/luasql/mysql.so':
/usr/lib/lua/5.1/luasql/mysql.so: undefined symbol: lua_getfield
stack traceback:
[C]: ?
[C]: in function 'require'
[string extensions.lua]:205: in function [string
extensions.lua]:204
I tested my script with a file.lua and works ok and the extensions.lua works
fine too. My extensions.lua:
extensions = {
luatest = {
[302] = function()
require(luasql.mysql)
app.Answer()
app.Log(NOTICE, Trying to connect in MySQL)
app.Wait(2)
env = assert(luasql.mysql())
sql = assert
(env:connect(asterisk_teste,root,*,localhost,3306))
sel = sql:execute('SELECT * FROM cdr;')
sel:fetch(Fetcharray)
app.Noop(Fetcharray[1])
end;
h = function()
app.Hangup()
end;
};
}
Does anyone know what is happening?
Thansk in advance,
--
Rodrigo Lang
Opening your mind - Just another Open Source
sitehttp://openingyourmind.wordpress.com/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Paging with Polycom 3.3.x
Sorry, I realize my tone might not go down well. I didn't mean to sound like a jerk, but I was just stating that resellers are also authorized to distribute the firmware to their customers if I recall correctly, so everybody can get the firmware for free, just not directly from Polycom. And I don't actually think this is the best way for Polycom to do things, but that`s the way things are. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, February 24, 2011 3:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Polycom are at 3.3.1 now, so 3.3.0 should be fair game. It has nothing to do with paying or not, the company that sold you the phone should be able to give you the latest version no? Unless you bought from a guy who found a box that fell off a truck.or some third-rate reseller. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 24, 2011 3:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Is 3.3.x downloadable for non-paying people yet? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, February 24, 2011 2:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Thank you Terry and Ryan, I will try those things and see if I can find my problem. Will definitely come back with my solution in case it helps somebody else. Mike Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=2000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ where the timeout is the ampount of time on milliseconds before it goes to speaker. These values are in the sip.cfg, so in your server it may be sip_316.cfg. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using a Virtual IP Line
Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work. I have traces of xlite for the invite and register this done to see if someone can help me to use this line with my asterisk. These are the traces of my Xllite REGISTER sip:Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-e322ee549824f666-1---d8754z-;rport Max-Forwards: 70 Contact: sip:888777@10.0.0.221:22818;rinstance=570ac597afa82c9a To: 888777sip:888777@Xlite release 1100l stamp 49022 From: 888777sip:888777@Xlite release 1100l stamp 49022;tag=fb1acd4f Call-ID: NjUyN2Q5ODEzZmZiNjRhY2FmNzRlZjljNDRjMjg5Yjk. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: Xlite release 1100l stamp 49022 Authorization: Digest username=888777,realm=192.168.50.20,nonce=d999e9471b1d,uri=sip:Xlite release 1100l stamp 49022,response=ba26805d2f0b97a70565c37e81444e44,cnonce=820e1f348b49cd73d92e1bc793be5ad7,nc=0001,qop=auth,algorithm=MD5 Content-Length: 0 REGISTER sip:Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-c605aa61ac248834-1---d8754z-;rport Max-Forwards: 70 Contact: sip:888777@33.33.33.33:22818;rinstance=da357fa09f45cdda To: 888777sip:888777@Xlite release 1100l stamp 49022 From: 888777sip:888777@Xlite release 1100l stamp 49022;tag=fb1acd4f Call-ID: NjUyN2Q5ODEzZmZiNjRhY2FmNzRlZjljNDRjMjg5Yjk. CSeq: 4 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: Xlite release 1100l stamp 49022 Authorization: Digest username=888777,realm=192.168.50.20,nonce=d999e9471b1d,uri=sip:Xlite release 1100l stamp 49022,response=b0858d0b5914f054faf8f0b0eed22400,cnonce=659200e211cc5023724817d04c14cb3a,nc=0003,qop=auth,algorithm=MD5 Content-Length: 0 SUBSCRIBE sip:888777@Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-23698d60215c9f07-1---d8754z-;rport Max-Forwards: 70 Contact: sip:888777@33.33.33.33:22818 To: 888777sip:888777@Xlite release 1100l stamp 49022 From: 888777sip:888777@Xlite release 1100l stamp 49022;tag=f5062e32 Call-ID: Y2Y5MjFjNWFlM2QzNWFiZjgwYWQxYTc5ZmRmZTVhOWE. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: Xlite release 1100l stamp 49022 Event: message-summary Content-Length: 0 INVITE sip:18094713172@Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-8c57153848230175-1---d8754z-;rport Max-Forwards: 70 Contact: sip:888777@33.33.33.33:22818 To: 18094713172sip:18094713172@Xlite release 1100l stamp 49022 From: 888777sip:888777@Xlite release 1100l stamp 49022;tag=0337ad04 Call-ID: NWNlMzIyNDZiNjUxNjA4NjQ4ZjM3ZDhjM2E3NmViNjQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Xlite release 1100l stamp 49022 Content-Length: 386 v=0 o=- 3 2 IN IP4 10.0.0.221 s=CounterPath eyeBeam 1.5 c=IN IP4 10.0.0.221 t=0 0 m=audio 48758 RTP/AVP 18 100 106 6 0 105 8 3 5 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:BB752EE94E6C4F5E870B02DB4DA411D5 Any help or any sugestion will be so appreciated. TIA *---* *-Edwin Quijada *- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
Actually, I don't think that has been the case for quite a while. Anyone can get the latest firmware directly from polycom. Including, 3.3.1F http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html On 02/24/2011 03:32 PM, Mike wrote: Sorry, I realize my tone might not go down well. I didn't mean to sound like a jerk, but I was just stating that resellers are also authorized to distribute the firmware to their customers if I recall correctly, so everybody can get the firmware for free, just not directly from Polycom. And I don't actually think this is the best way for Polycom to do things, but that`s the way things are. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, February 24, 2011 3:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Polycom are at 3.3.1 now, so 3.3.0 should be fair game. It has nothing to do with paying or not, the company that sold you the phone should be able to give you the latest version no? Unless you bought from a guy who found a box that fell off a truck.or some third-rate reseller. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 24, 2011 3:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Is 3.3.x downloadable for non-paying people yet? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, February 24, 2011 2:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Thank you Terry and Ryan, I will try those things and see if I can find my problem. Will definitely come back with my solution in case it helps somebody else. Mike Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=2000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ where the timeout is the ampount of time on milliseconds before it goes to speaker. These values are in the sip.cfg, so in your server it may be sip_316.cfg. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- DO NOT SEND WITH THIS ACCOUNT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recieve_Fax caused crash 1.8.2.3
I had an issue today where receive_fax caused an asterisk switch to crash. The switch is 1.8.2.3 version. The call was coming from a fax machine. The call started receive_fax answered and then asterisk stopped responding. I was able to log into asterisk but it would not do a core restart now nor would it take any calls or show an peer registrations. I had to kill the asterisk process and restart it. As best we can tell there was no attempt by the sender to intentionally send any malformed packets that should have caused this. I see there is a security patch 1.8.2.4 that lists some RTP security issues. is it possible that this fix may address what I ran into as well? Thanks zktech -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3
On Thu, Feb 24, 2011 at 6:05 PM, Bryant Zimmerman brya...@zktech.com wrote: I had an issue today where receive_fax caused an asterisk switch to crash. The switch is 1.8.2.3 version. The call was coming from a fax machine. The call started receive_fax answered and then asterisk stopped responding. I was able to log into asterisk but it would not do a core restart now nor would it take any calls or show an peer registrations. I had to kill the asterisk process and restart it. As best we can tell there was no attempt by the sender to intentionally send any malformed packets that should have caused this. I see there is a security patch 1.8.2.4 that lists some RTP security issues. is it possible that this fix may address what I ran into as well? Thanks zktech There are many updates in 1.8.2.4 that may fix your issue. If you are running any version of 1.8 it should be a quick update. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4.39.2] Simple AGI doesn't reply
On Thu, 24 Feb 2011 08:27:13 -0800 (PST), Steve Edwards asterisk@sedwards.com wrote: Bash has a thing about syntax too. Note you're not 'done' with your second loop. Sorry about this :-/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] missing argument on AGI
Hi All,
I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan:
[callback-outbound]
exten = _00.,1,Macro(callout|${EXTEN})
[macro-callout]
exten = s,1,AGI(getchannel.php|${ARG1})
exten = s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr)
exten = s,3,Hangup()
but for some reason i am not receiving the argument:
Executing [s@macro-callout:2] Dial(SIP/201-0004,
Local/@from-internal/nj||tr) in new stack
[Feb 24 21:47:11] NOTICE[1901] chan_local.c: No such extension/context
@from-internal while calling Local channel
the number is missing, i get the number from the agi, below is the debug:
21:47:10]-- Executing [006583232393-1-201@callback-outbound:1]
Macro(SIP/201-0004, callout|006583232393-1-201) in new stack
21:47:10]-- Executing [s@macro-callout:1] AGI(SIP/201-0004,
getchannel.php|006583232393-1-201) in new stack
21:47:10]-- Launched AGI Script /var/lib/asterisk/agi-bin/getchannel.php
21:47:10]AGI Tx agi_request: getchannel.php
21:47:10]AGI Tx agi_channel: SIP/201-0004
21:47:10]AGI Tx agi_language: en
21:47:10]AGI Tx agi_type: SIP
21:47:10]AGI Tx agi_uniqueid: 1298555228.12
21:47:10]AGI Tx agi_callerid: unknown
21:47:10]AGI Tx agi_calleridname: unknown
21:47:10]AGI Tx agi_callingpres: 0
21:47:10]AGI Tx agi_callingani2: 0
21:47:10]AGI Tx agi_callington: 0
21:47:10]AGI Tx agi_callingtns: 0
21:47:10]AGI Tx agi_dnid: unknown
21:47:10]AGI Tx agi_rdnis: unknown
21:47:10]AGI Tx agi_context: macro-callout
21:47:10]AGI Tx agi_extension: s
21:47:10]AGI Tx agi_priority: 1
21:47:10]AGI Tx agi_enhanced: 0.0
21:47:10]AGI Tx agi_accountcode:
21:47:10]AGI Tx
21:47:10]AGI Rx EXEC Noop
21:47:10]-- AGI Script Executing Application: (Noop) Options:
((null)) == THIS SHOULD DISPLAY THE ARGUMENT
21:47:10]AGI Tx 200 result=0
21:47:11]AGI Rx EXEC Set CALLERID(num)=
21:47:11]-- AGI Script Executing Application: (Set) Options:
(CALLERID(num)=)
21:47:11]AGI Tx 200 result=0
21:47:11]AGI Rx EXEC Set OUTBOUND=
21:47:11]-- AGI Script Executing Application: (Set) Options: (OUTBOUND=)
21:47:11]AGI Tx 200 result=0
my php code include something:
#!/usr/bin/php-cgi -q
?php
include('phpagi/phpagi.php');
$agi=new AGI();
$param = $argv[1];
$agi - exec(Noop,$param);
.
.
.
.
?
not sure where to check next i'm stumped, hope somebody can help. thanks
in advance.
Regards
Ron
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Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3
On 11-02-24 04:08 PM, Andrew Latham wrote: There are many updates in 1.8.2.4 that may fix your issue. If you are running any version of 1.8 it should be a quick update. I wouldn't say many. There is one fix in 1.8.2.4 over 1.8.2.3. From the ChangeLog: * Asterisk 1.8.2.4 Released. * AST-2011-002: Multiple array overflow and crash vulnerabilities in UDPTL code The release announcement for AST-2011-002 is here: http://downloads.asterisk.org/pub/security/AST-2011-002.pdf Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3
On Thu, Feb 24, 2011 at 7:43 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 11-02-24 04:08 PM, Andrew Latham wrote: There are many updates in 1.8.2.4 that may fix your issue. If you are running any version of 1.8 it should be a quick update. I wouldn't say many. There is one fix in 1.8.2.4 over 1.8.2.3. From the ChangeLog: * Asterisk 1.8.2.4 Released. * AST-2011-002: Multiple array overflow and crash vulnerabilities in UDPTL code The release announcement for AST-2011-002 is here: http://downloads.asterisk.org/pub/security/AST-2011-002.pdf Leif. And I go back to triple check and compare revision numbers... You are 100% correct, the revision numbers in our local repository are wrong, someone pushed the 1.8.3 RC3 into our local 1.8.2 branch. I apologize and will work to better control my trust of other engineers as this is twice in one week I have looked like an ass. International bandwidth limits change how you work and as a business force the mirroring of as many sources as possible. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
On Tue, Feb 22, 2011 at 12:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Has this issue been fixed in this release of 1.8 (or even in the previous 1.8.2.3)? https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 Thanks Ish Ishfaq, I spoke to soon and was looking at the wrong checkout. The 1.8.2.4 does NOT have the patch from issue 18403. Asterisk Branch 1.8.3 does have the patch which happened just 1 day after the 1.8.2.4 release. I must have lost the release email because I can only find the tag in SVN. I was confused and hope I did not cause you any confusion. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] missing argument on AGI
On Feb 24, 2011, at 5:27 PM, Ron wrote:
Hi All,
I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan:
[callback-outbound]
exten = _00.,1,Macro(callout|${EXTEN})
[macro-callout]
exten = s,1,AGI(getchannel.php|${ARG1})
exten = s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr)
exten = s,3,Hangup()
but for some reason i am not receiving the argument:
Executing [s@macro-callout:2] Dial(SIP/201-0004,
Local/@from-internal/nj||tr) in new stack
[Feb 24 21:47:11] NOTICE[1901] chan_local.c: No such extension/context
@from-internal while calling Local channel
the number is missing, i get the number from the agi, below is the debug:
21:47:10]-- Executing [006583232393-1-201@callback-outbound:1]
Macro(SIP/201-0004, callout|006583232393-1-201) in new stack
21:47:10]-- Executing [s@macro-callout:1] AGI(SIP/201-0004,
getchannel.php|006583232393-1-201) in new stack
21:47:10]-- Launched AGI Script /var/lib/asterisk/agi-bin/getchannel.php
21:47:10]AGI Tx agi_request: getchannel.php
21:47:10]AGI Tx agi_channel: SIP/201-0004
21:47:10]AGI Tx agi_language: en
21:47:10]AGI Tx agi_type: SIP
21:47:10]AGI Tx agi_uniqueid: 1298555228.12
21:47:10]AGI Tx agi_callerid: unknown
21:47:10]AGI Tx agi_calleridname: unknown
21:47:10]AGI Tx agi_callingpres: 0
21:47:10]AGI Tx agi_callingani2: 0
21:47:10]AGI Tx agi_callington: 0
21:47:10]AGI Tx agi_callingtns: 0
21:47:10]AGI Tx agi_dnid: unknown
21:47:10]AGI Tx agi_rdnis: unknown
21:47:10]AGI Tx agi_context: macro-callout
21:47:10]AGI Tx agi_extension: s
21:47:10]AGI Tx agi_priority: 1
21:47:10]AGI Tx agi_enhanced: 0.0
21:47:10]AGI Tx agi_accountcode:
21:47:10]AGI Tx
21:47:10]AGI Rx EXEC Noop
21:47:10]-- AGI Script Executing Application: (Noop) Options: ((null))
== THIS SHOULD DISPLAY THE ARGUMENT
21:47:10]AGI Tx 200 result=0
21:47:11]AGI Rx EXEC Set CALLERID(num)=
21:47:11]-- AGI Script Executing Application: (Set) Options:
(CALLERID(num)=)
21:47:11]AGI Tx 200 result=0
21:47:11]AGI Rx EXEC Set OUTBOUND=
21:47:11]-- AGI Script Executing Application: (Set) Options: (OUTBOUND=)
^ --- This
You are relying on the channel variable ${OUTBOUND} to be set when you invoke
the Dial() string, but the AGI script is not setting OUTBOUND correctly. In
both cases you illustrate (the Noop and the Set) the variable isn't making it
from PHP to AGI. Check your PHP script to make sure the data is what you think
it is.
/BAK/
--
Ben Klang
bkl...@mojolingo.com
404.475.4841
Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com
Twitter: @MojoLingo
21:47:11]AGI Tx 200 result=0
my php code include something:
#!/usr/bin/php-cgi -q
?php
include('phpagi/phpagi.php');
$agi=new AGI();
$param = $argv[1];
$agi - exec(Noop,$param);
..
..
..
..
?
not sure where to check next i'm stumped, hope somebody can help. thanks in
advance.
Regards
Ron
--
_
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Re: [asterisk-users] missing argument on AGI
On 2/24/2011 9:10 PM, Ben Klang wrote:
On Feb 24, 2011, at 5:27 PM, Ron wrote:
Hi All,
I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial
plan:
[callback-outbound]
exten = _00.,1,Macro(callout|${EXTEN})
[macro-callout]
exten = s,1,AGI(getchannel.php|${ARG1})
exten = s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr)
exten = s,3,Hangup()
but for some reason i am not receiving the argument:
Executing [s@macro-callout:2] Dial(SIP/201-0004,
Local/@from-internal/nj||tr) in new stack
[Feb 24 21:47:11] NOTICE[1901] chan_local.c: No such
extension/context @from-internal while calling Local channel
the number is missing, i get the number from the agi, below is the debug:
21:47:10]-- Executing [006583232393-1-201@callback-outbound:1]
Macro(SIP/201-0004, callout|006583232393-1-201) in new stack
21:47:10]-- Executing [s@macro-callout:1] AGI(SIP/201-0004,
getchannel.php|006583232393-1-201) in new stack
21:47:10]-- Launched AGI Script
/var/lib/asterisk/agi-bin/getchannel.php
21:47:10]AGI Tx agi_request: getchannel.php
21:47:10]AGI Tx agi_channel: SIP/201-0004
21:47:10]AGI Tx agi_language: en
21:47:10]AGI Tx agi_type: SIP
21:47:10]AGI Tx agi_uniqueid: 1298555228.12
21:47:10]AGI Tx agi_callerid: unknown
21:47:10]AGI Tx agi_calleridname: unknown
21:47:10]AGI Tx agi_callingpres: 0
21:47:10]AGI Tx agi_callingani2: 0
21:47:10]AGI Tx agi_callington: 0
21:47:10]AGI Tx agi_callingtns: 0
21:47:10]AGI Tx agi_dnid: unknown
21:47:10]AGI Tx agi_rdnis: unknown
21:47:10]AGI Tx agi_context: macro-callout
21:47:10]AGI Tx agi_extension: s
21:47:10]AGI Tx agi_priority: 1
21:47:10]AGI Tx agi_enhanced: 0.0
21:47:10]AGI Tx agi_accountcode:
21:47:10]AGI Tx
21:47:10]AGI Rx EXEC Noop
21:47:10]-- AGI Script Executing Application: (Noop) Options:
((null)) == THIS SHOULD DISPLAY THE ARGUMENT
21:47:10]AGI Tx 200 result=0
21:47:11]AGI Rx EXEC Set CALLERID(num)=
21:47:11]-- AGI Script Executing Application: (Set) Options:
(CALLERID(num)=)
21:47:11]AGI Tx 200 result=0
21:47:11]AGI Rx EXEC Set OUTBOUND=
21:47:11]-- AGI Script Executing Application: (Set) Options:
(OUTBOUND=)
^ --- This
You are relying on the channel variable ${OUTBOUND} to be set when you
invoke the Dial() string, but the AGI script is not setting OUTBOUND
correctly. In both cases you illustrate (the Noop and the Set) the
variable isn't making it from PHP to AGI. Check your PHP script to
make sure the data is what you think it is.
/BAK/
--
Ben Klang
bkl...@mojolingo.com mailto:bkl...@mojolingo.com
404.475.4841
Mojo Lingo -- /Voice applications that work like magic/
http://mojolingo.com http://mojolingo.com/
Twitter: @MojoLingo
21:47:11]AGI Tx 200 result=0
my php code include something:
#!/usr/bin/php-cgi -q
?php
include('phpagi/phpagi.php');
$agi=new AGI();
$param = $argv[1];
$agi - exec(Noop,$param);
..
..
..
..
?
not sure where to check next i'm stumped, hope somebody can help.
thanks in advance.
Regards
Ron
--
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Hi,
Just my 2 cents...Did you try , instead | in
AGI(getchannel.php|${ARG1})?
Regards
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Re: [asterisk-users] Google Voice outbound Caller ID broken
Chris, Let me summarize: 1. GV Outbound CID shows Unknown, Unavailable, Out of area (depending on a recipient's carrier) starting some time around 02/15/2011 if a call is placed via Google Chat/Google Talk/Google Mail/Asterisk GTalk channel. See http://www.google.com/support/forum/p/voice/thread?tid=49c21d292e80ff65hl=enstart=40 for other users' accounts. 2. This CID feature failure affects only some GV phones, some still work fine as of 02/24/2011. 3. Calls placed with GV call-back facility work fine for the phones affected by the issue described in #1. 4. A workaround is to set Caller ID (incoming) to Display my Google Voice number As expected it will suppress an incoming CID. So it is not a perfect workaround. 5. Another workaround is to trigger a GV callback facility per William Stillwell's posting. Connection time increases with this workaround. Historically it took 15 days to a month for Google to fix similar problems. -Vladimir On 2/24/2011 8:51 AM, Chris Gentle wrote: Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice outbound Caller ID broken
Further analysis showed that a call placed using a GTalk channel which came as Restricted was not recorded under History / Placed in Google Voice. A call placed using the same GTalk trunk an hour later was terminated to the same recipient's phone with the proper CID. It looks like a call routing issue on the Google Voice end to me. -Vladimir On 2/24/2011 10:40 PM, Vladimir Mikhelson wrote: Chris, Let me summarize: 1. GV Outbound CID shows Unknown, Unavailable, Out of area (depending on a recipient's carrier) starting some time around 02/15/2011 if a call is placed via Google Chat/Google Talk/Google Mail/Asterisk GTalk channel. See http://www.google.com/support/forum/p/voice/thread?tid=49c21d292e80ff65hl=enstart=40 for other users' accounts. 2. This CID feature failure affects only some GV phones, some still work fine as of 02/24/2011. 3. Calls placed with GV call-back facility work fine for the phones affected by the issue described in #1. 4. A workaround is to set Caller ID (incoming) to Display my Google Voice number As expected it will suppress an incoming CID. So it is not a perfect workaround. 5. Another workaround is to trigger a GV callback facility per William Stillwell's posting. Connection time increases with this workaround. Historically it took 15 days to a month for Google to fix similar problems. -Vladimir On 2/24/2011 8:51 AM, Chris Gentle wrote: Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
