Re: [asterisk-users] Quoting error with gotoiftime
Error doesn't occur in 11.2.1
-- Executing [1260@default:1] Answer(SIP/sipuser-0001, ) in new
stack
-- Executing [1260@default:2] Goto(SIP/sipuser-0001,
scottsdale#queues-account,s,1) in new stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing [s@scottsdale#queues-account:1]
GotoIfTime(SIP/sipuser-0001, 8:00-16:55,mon-fri,*,*?queue) in new
stack
-- Goto (scottsdale#queues-account,s,3)
-- Executing [s@scottsdale#queues-account:3]
ExecIf(SIP/sipuser-0001, 1?Queue(azenglish):Queue(azspanish)) in new
stack
== Spawn extension (scottsdale#queues-account, s, 3) exited non-zero on
'SIP/sipuser-0001'
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Friday, January 25, 2013 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Quoting error with gotoiftime
I'm getting the following error, and none of us can figure out why:
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting
$end; Input:
=
^
Here is the code that generates it:
[scottsdale#queues-account]
exten = s,1,GotoIfTime(8:00-16:55,mon-fri,*,*?queue)
exten = s,n,Goto(scottsdale#queues-closed,s,1)
exten = s,n(queue),ExecIf($[${prefix} =
]?Queue(azenglish):Queue(azspanish))
exten = s,n(callback),Playback(scottsdale-q/${prefix}callbackmessage)
exten = s,n,Voicemail(@scottsdale,s)
exten = s,n,Hangup
Here is the rest of the call progress surrounding it, which seems to be
working anyway:
-- Executing [2@scottsdale#queues-aax:1]
Goto(SIP/televolve-1-1c7d, scottsdale#queues-account,s,1) in new
stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing [s@scottsdale#queues-account:1]
GotoIfTime(SIP/televolve-1-1c7d, 8:00-16:55,mon-fri,*,*?queue) in
new stack
-- Goto (scottsdale#queues-account,s,3)
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting
$end; Input:
=
^
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [s@scottsdale#queues-account:3]
ExecIf(SIP/televolve-1-1c7d, ?Queue(azenglish):Queue(azspanish))
in new stack
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- 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] Quoting error with gotoiftime
Where possible you should have a VM to try these things as needed. Where not, it isn't too difficult to duplicate the contexts and do something like this [default] . . Exten = 1260,1,answer Exten = 1260,n,goto(test-context,s,1) . From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Friday, January 25, 2013 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Quoting error with gotoiftime On Fri, Jan 25, 2013 at 9:27 AM, Eric Wieling ewiel...@nyigc.com wrote: Don't do that. Set(prefix=) You are setting the prefix to have two quotes. You WANT prefix to be empty. I'll give that a try during non-production hours. Odd that the same code works in earlier versions and later, but not this one. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 11 / Missing Application SetCallerPres
Simplest question first. Does it show up in core show applications or core show application SetCallerPres? If not, do a make menuselect and see if something broke in the ability to make the application. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten Göllner Sent: Thursday, January 24, 2013 8:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 11 / Missing Application SetCallerPres Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI == Using SIP RTP CoS mark 5 -- Executing [100@sip:1] AGI(SIP/userid-001e, test.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script Executing Application: (SetCallerPres) Options: (prohib_not_screened) [Jan 24 15:20:04] WARNING[15507][C-0030]: res_agi.c:2527 handle_exec: Could not find application (SetCallerPres) Why is the application not found, please? I think it should exist: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_SetCaller Pres Best regards, -Thorsten- -- _ -- 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] Integration with Social Media, Email and Web call center
Yes it might be “hacky”, but anything that isn’t somewhat is going to come at a premium price. Today’s motto is “get her done as quick and cheap as possible”. It is a luxury to have a well-trained, professional staff providing solid solutions when folks want Top Quality at slave wage labor prices. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Thursday, January 24, 2013 8:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Integration with Social Media, Email and Web call center And how would you have this working together with Asterisk queueing? I have seen solutions like this using agent pauses and then making everyithing happen outside the normal ACD flow, but it's a bit of a hack l. 2013/1/22 Danny Nicholas da...@debsinc.com For just the messaging part, you should be able to use wget or curl to interface and create messages. You might have to go a little higher level like C or Perl, but it sounds very doable. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, January 22, 2013 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integration with Social Media, Email and Web call center Dears; Can someone advise me where to find a technology (open source) that let us able to integrate with social media like whatsapp and facebook? And use this in call center (queuing the messages and routing it for agent)? Anyone give me a light to start? Regards Bilal -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] Integration with Social Media, Email and Web call center
This is how I would see the process working 1. use curl/wget to query Facebook (etc.) 2. determine whether we are to drop a call into the queue or just process a message 3. determine agent availability through AMI process or asterisk -rx process. 4. drop the call into the queue or place the message if the agent is available 5. if the agent is unavailable, do alternate process. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, January 24, 2013 9:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Integration with Social Media, Email and Web call center They advised me to check jabber.org. Yes, jabber.org has a client that can send/receive and integrate with other social media (facebook, msn, twitter, ... etc). But, as an Agent who can login/logout and take a calls, how can I make it to be single login for voice and messages. So, if the agent is not available, he will not get a calls and will not get a messages. Those who used jabber.org or who used other than jabber.org for such requirement, what do you suggest? Regards Bilal -- For just the messaging part, you should be able to use wget or curl to interface and create messages. You might have to go a little higher level like C or Perl, but it sounds very doable. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, January 22, 2013 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integration with Social Media, Email and Web call center Dears; Can someone advise me where to find a technology (open source) that let us able to integrate with social media like whatsapp and facebook? And use this in call center (queuing the messages and routing it for agent)? Anyone give me a light to start? Regards Bilal -- _ -- 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] question on SIP trunk and AMI to place call
Not the greatest solution, but since you are most likely using a script for the AMI process, you could do an Asterisk –rx “core show channels verbose”|grep SIP/testmachine-000d And get the dialed number from that. Actually you could issue the AMI command core show channels verbose. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada Sent: Thursday, January 24, 2013 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call Have you tried and looked up all events generated when you place the call? some of them are bound to have the variable callerid set On 24 January 2013 16:46, Jerry Geis ge...@pagestation.com wrote: When I am monitoring the AMI I see the following event for a call I just made over a SIP trunk. Event: Newchannel Privilege: call,all Channel: SIP/testmachine-000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: testmachine Uniqueid: 1359035395.20 In this event or any event following I do not see the phone number that I dialled. How do I correlate the SIP/testmachine-000d to the number I just dialed (purpose is to hangup the call later if I need to interrupt it) Now if I am using a machine with actual hardware cards, the phone number is included as part of the Channel so I can look that up. but for a SIP trunk the phone number dialled does not come over the AMI. How do I match up the call I just started (using AMI over SIP trunk) to the number I called? Thanks, jerry -- _ -- 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] question on SIP trunk and AMI to place call
This might have changed but IIRC /etc/asterisk/manager.conf controls what events you have access to. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, January 24, 2013 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call You probably want the Dial event. It is raised both at the beginning of the Dial, as well as when the Dial completes. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial Note that the Channel: field will contain the name initiating the Dial, the Destination: field will contain the channel name being dialled, and the Dialstring: field will contain the non-technology specific portion of the thing being dialled. I get that even on the system with the PRI card and using DAHDI however I am not getting that event on the system with the SIP trunk . Is there something to enable to get that??? Both systems are running Asterisk 11.0.2. Thanks, Jerry -- _ -- 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] clicking sound with alaw codec
Your sounds might be too loud. We use a lot of custom sounds here and when the volume approaches 0 db (asterisk standard is -3 db) we get fuzz and clicks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner Sent: Thursday, January 24, 2013 2:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] clicking sound with alaw codec I'm trying to interface Asterisk with an Alcatel PABX and trying to find a code that works well. It says it doesn't support ulaw, though it doesn't reject it. It supports G.729, and that works fine, but we'd prefer not to use compression. When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. The outgoing SDP looks like this: v=0 o=root 1691755711 1691755711 IN IP4 205.232.38.178 s=Asterisk PBX 10.7.1 c=IN IP4 205.232.38.178 t=0 0 m=audio 11432 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv The reply SDP is: v=0 o=default 1359060187 1359060187 IN IP4 10.10.22.246 s=Asterisk PBX 10.7.1 c=IN IP4 10.10.22.246 t=0 0 m=audio 32000 RTP/AVP 8 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=maxptime:90 Any suggestions on how to debug what's causing 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 -- _ -- 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] Execute a script outside Asterisk
I would vote for system() on two accounts. #1 AGI requires more overhead and protocol #2 you are not expecting a result to return to the dialplan. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January 23, 2013 4:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Execute a script outside Asterisk Hello, at certain time inside my dialplan I would like to have an external php script executed. Asterisk should not wait for the end of the execution to continue with the rest of the dialplan. It should just start the execution of the php script (which inserts an entry into a remote mysql-DB). What is the best way to work ? - with AGI inside the dialplan ? - with the system()-command inside the dialplan ? Kind regards, Jonas. -- _ -- 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] Execute a script outside Asterisk
Let's assume you're using this snippet [default] Exten = s,1,answer() Exten = s,n,playback(tt-monkeys) Exten = s,n,waitexten(6) Exten = s,n,hangup() Exten = 1,1,AGI(Jonas.php) Exten = 1,n,playback(vm-goodbye) Exten = 1,n,hangup() Exten = 2,1,system(Jonas.php) Exten = 2,n,playback(vm-goodbye) Exten = 2,n,hangup() Both of these do the exact same thing - pick up the line, play tt-monkeys, run Jonas.php if you press 1 or 2, play vm-goodbye and hangup. The failure of Jonas.php due to database or any other problem would not affect the execution of the dialplan. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January 23, 2013 8:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Execute a script outside Asterisk Hello, thank you for your answer. The most important here is that Asterisk continues with the rest of the dialplan, in case the database-connection fails or hangs or ... I don't think the System()-command makes this true. Jonas. On 01/23/2013 03:27 PM, Danny Nicholas wrote: I would vote for system() on two accounts. #1 AGI requires more overhead and protocol #2 you are not expecting a result to return to the dialplan. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January 23, 2013 4:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Execute a script outside Asterisk Hello, at certain time inside my dialplan I would like to have an external php script executed. Asterisk should not wait for the end of the execution to continue with the rest of the dialplan. It should just start the execution of the php script (which inserts an entry into a remote mysql-DB). What is the best way to work ? - with AGI inside the dialplan ? - with the system()-command inside the dialplan ? Kind regards, Jonas. -- _ -- 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] Execute a script outside Asterisk
Here is the way I got it to do what I think you want. '1250' = 1. answer() [pbx_config] 2. setMusiconhold(jazz) [pbx_config] 3. AGI(wait10.sh) [pbx_config] 4. playback(vm-goodbye) [pbx_config] 5. setMusiconhold(monkey) [pbx_config] 6. system(/var/lib/asterisk/agi-bin/wait10.sh ) [pbx_config] 7. playback(vm-goodbye) [pbx_config] 8. hangup() [pbx_config] Without the , AGI and system both execute and wait for completion of wait10.sh. with the , the system command returns control to the dialplan immediately. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January 23, 2013 8:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Execute a script outside Asterisk Hello, will this : Exten = 2,n,playback(vm-goodbye) be executed even when Exten = 2,1,system(Jonas.php) is still executing ?? The exact snippet would be : Exten = s,1,answer() Exten = s,n,system(Jonas.php) ; script that may take a minute Exten = s,n,do something Exten = s,n,Dial(SIP/peer1,,10) ; dial peer 1 Exten = s,n,system(Jonas.php) ; script that may take a minute Exten = s,n,do something Exten = s,n,Dial(SIP/peer2,,10) ; dial peer 2 Exten = s,n,system(Jonas.php) ; script that may take a minute Exten = s,n,do something Exten = s,n,Dial(SIP/peer3,,10) ; dial peer 3 Exten = s,n,hangup() The peer MUST be dialed even if the script Jonas.php is still running. Jonas. On 01/23/2013 03:44 PM, Danny Nicholas wrote: Let's assume you're using this snippet [default] Exten = s,1,answer() Exten = s,n,playback(tt-monkeys) Exten = s,n,waitexten(6) Exten = s,n,hangup() Exten = 1,1,AGI(Jonas.php) Exten = 1,n,playback(vm-goodbye) Exten = 1,n,hangup() Exten = 2,1,system(Jonas.php) Exten = 2,n,playback(vm-goodbye) Exten = 2,n,hangup() Both of these do the exact same thing - pick up the line, play tt-monkeys, run Jonas.php if you press 1 or 2, play vm-goodbye and hangup. The failure of Jonas.php due to database or any other problem would not affect the execution of the dialplan. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January 23, 2013 8:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Execute a script outside Asterisk Hello, thank you for your answer. The most important here is that Asterisk continues with the rest of the dialplan, in case the database-connection fails or hangs or ... I don't think the System()-command makes this true. Jonas. On 01/23/2013 03:27 PM, Danny Nicholas wrote: I would vote for system() on two accounts. #1 AGI requires more overhead and protocol #2 you are not expecting a result to return to the dialplan. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, January 23, 2013 4:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Execute a script outside Asterisk Hello, at certain time inside my dialplan I would like to have an external php script executed. Asterisk should not wait for the end of the execution to continue with the rest of the dialplan. It should just start the execution of the php script (which inserts an entry into a remote mysql-DB). What is the best way to work ? - with AGI inside the dialplan ? - with the system()-command inside the dialplan ? Kind regards, Jonas. -- _ -- 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
Re: [asterisk-users] two steps when calling from web!
Originate is the answer here. Let’s say your X-lite is SIP/100 and you’re dialing 555-1212. From the x-lite you dial 555-1212 and Asterisk does a dial command to execute the call. From the web, we “originate” the call from SIP/100 to 555-1212. Asterisk makes sure SIP/100 is available then dials the call. sendcommand( Action = 'Originate', Channel = SIP/100, Exten = 5551212, Context = 'default', priority = 1, Number = 5551212 ); I use this in my office with Apache 1.X and 2.X. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Wednesday, January 23, 2013 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] two steps when calling from web! On Wed, Jan 23, 2013 at 1:09 AM, Muhammad mohammad.ghaz...@gmail.com wrote: -1 in normal way, when I type the number in softphone, it call the number and show me just End bottom. 2- when I calling the number through the web, it show me Answer bottom and I have to click answer to calling then number. it is 2 steps to calling from web. For Asterisk, there is no way to bring a device in on a call unless Asterisk dials out to it first. That device needs to accept the Asterisk-originated call as if a normal call were incoming. When I was referring to headers, I was talking about SIP headers that allow many hardware SIP phones to go into what is effectively an intercom mode, not requiring an explicit answer function. I don't know (off of the top of my head) how to set SIP headers from the AMI originate action, but I suppose there probably is some way to do it. Then question then becomes whether or not your softphone supports it. Otherwise, there may be an option to configure your softphone to simply automatically answer all incoming calls. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Is there a need to secure RTP ports?
As I am going to mis-explain this, an Asterisk SIP call originates on port 5060 (incoming or outgoing) then uses two RTP ports for audio in and audio out. Police and Hackers can tap into the RTP ports to monitor your conversations (I don't really know if the capabilities stop there) but you can limit your exposure by changing the default 1-2 range to a range of 4 per anticipated calls simultaneously. If you have 5 phones in your shop, you aren't going to make 2500 simultaneous calls (just seems like telemarketers can do this). Change the 1-2 to 10001-10040 for a 5 phone shop. This lets all 5 phones have two calls going at once. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Arcus Sent: Wednesday, January 23, 2013 11:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is there a need to secure RTP ports? I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open at my end to the Internet. However, I have the RTP ports open (as SIP has some trouble with my NAT). My question is - what are the vulnerabilities in this scenario at my end? I suppose some man-in-the-middle or eavesdropping attack is always a possibility - but that aside, is there anything that will attack RTP ports on Asterisk when there are no SIP ports open? I was looking into installing fail2ban - until I realised that there is no SIP port exposed for an attacker to poke at. Searching on Google for secure RTP ports keeps on bringing up results about SRTP - which is not exactly the answer to my question. Thank you -- _ -- 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 with no voice
You are obviously getting the call connected, so the subnet issue is moot.
What this sounds like (pardon the pun) to me is an rtp skip issue. The
working calls are generating rtp connections in the allowed range; the
other calls have one or more ports outside of your rtp range. Verify that
all of your ports defined in rtp.conf (1-2 by default) are open in
the firewall.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
Sent: Tuesday, January 22, 2013 10:18 AM
To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Google voice with no voice
Chris,
I covered the whole 74.125.225.* subnet.
Even if I open the ports mentioned below for all (not limited to IP
addresses) I still have the same issue.
Have anyone ever succeeded in such configuration? :
Digium phones on 2 different private networks (2 different buildings)
Asterisk server in the internet with a public IP Use Google Voice
Even if you have asterisk on a private network, but have the same kind of
solution working for you, I'd love to hear your story..
On 1/22/13 9:55 AM, Christopher Harrington wrote:
On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com
mailto:fr...@efirehouse.com wrote:
Actually, the funny thing is that it works randomly.
This may be due to the fact that voice.google.com
http://voice.google.com actually resolves to a range of IP addresses.
When you set up your firewall, it may not be including all of the
possible resolutions for voice.google.com...
voice.l.google.com http://voice.l.google.com.300INA74.125.225.36
voice.l.google.com http://voice.l.google.com.300INA74.125.225.46
voice.l.google.com http://voice.l.google.com.300INA74.125.225.33
voice.l.google.com http://voice.l.google.com.300INA74.125.225.32
voice.l.google.com http://voice.l.google.com.300INA74.125.225.41
voice.l.google.com http://voice.l.google.com.300INA74.125.225.38
voice.l.google.com http://voice.l.google.com.300INA74.125.225.35
voice.l.google.com http://voice.l.google.com.300INA74.125.225.39
voice.l.google.com http://voice.l.google.com.300INA74.125.225.40
voice.l.google.com http://voice.l.google.com.300INA74.125.225.34
voice.l.google.com http://voice.l.google.com.300INA74.125.225.37
(ie 74.125.225.32-41 and 74.125.225.46)
Since these are short TTL values (the 300 means 5 minutes) there may be
a brief period where your devices and your firewall agree, before one or
both change their mind about the IP address behind that hostname.
I just tried out of the blue calling from D70 through Google Voice
to a cell phone, and it worked. I hung up, redial, and no audio at
all.
On 1/21/13 10:38 PM, Frank wrote:
Greetings all,
I was reading the documentation tonight, and decided to try
Google voice
with my asterisk.
I was able to setup iksemel, connect to google using jabber, and
connect
to google voice using gtalk.
Here is my physical configuration:
Digium D70 -- private network 192.168.1.x -- Airport express
--
Internet -- Asterisk with public IP
My asterisk has the following ports open:
5060 tcp/udp from my Airport Express public IP and from
voice.google.com http://voice.google.com
10,000:20,000 from my Airport Express public IP and from
voice.google.com http://voice.google.com
My issue is that when I place a call with google voice, I have
no audio
path at all in both way.
When a call is received on google voice (and sent to the D70),
if I pick
up, nothing happen, and the caller still hear the ringing tone.
My D70 is setup as follow in the sip.conf:
[D70]
type=friend
nat=yes
qualify=yes
directmedia=no
host=dynamic
secret=takapoum
disallow=all
allow=ulaw
context=LocalSets
mailbox=D70@default
my gtalk.conf is setup as follow:
[general]
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=gtalk_incoming
connection=asterisk
and finally, the interesting parts in my extensions.conf are
setup as
follow:
;Dialing out on google voice:
exten =
_1zxxzxx,1,Dial(Gtalk/__asterisk/+${EXTEN}@voice.__google.com
mailto:exten...@voice.google.com)
same = n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten = r...@gmail.com mailto:r...@gmail.com,1,Verbose(0,
Incoming gtalk from ${CALLERID(all)})
same = n,Answer()
same = n,Wait(2)
same = n,Dial(SIP/D70)
same = Hangup()
I would appreciate if
Re: [asterisk-users] Google voice with no voice
Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com http://voice.l.google.com.300INA74.125.225.36 voice.l.google.com http://voice.l.google.com.300INA74.125.225.46 voice.l.google.com http://voice.l.google.com.300INA74.125.225.33 voice.l.google.com http://voice.l.google.com.300INA74.125.225.32 voice.l.google.com http://voice.l.google.com.300INA74.125.225.41 voice.l.google.com http://voice.l.google.com.300INA74.125.225.38 voice.l.google.com http://voice.l.google.com.300INA74.125.225.35 voice.l.google.com http://voice.l.google.com.300INA74.125.225.39 voice.l.google.com http://voice.l.google.com.300INA74.125.225.40 voice.l.google.com http://voice.l.google.com.300INA74.125.225.34 voice.l.google.com http://voice.l.google.com.300INA74.125.225.37 (ie 74.125.225.32-41 and 74.125.225.46) Since these are short TTL values (the 300 means 5 minutes) there may be a brief period where your devices and your firewall agree, before one or both change their mind about the IP address behind that hostname. I just tried out of the blue calling from D70 through Google Voice to a cell phone, and it worked. I hung up, redial, and no audio at all. On 1/21/13 10:38 PM, Frank wrote: Greetings all, I was reading the documentation tonight, and decided to try Google voice with my asterisk. I was able to setup iksemel, connect to google using jabber, and connect to google voice using gtalk. Here is my physical configuration: Digium D70 -- private network 192.168.1.x -- Airport express -- Internet -- Asterisk with public IP My asterisk has
Re: [asterisk-users] Google voice with no voice
Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com http://voice.l.google.com.300INA74.125.225.36 voice.l.google.com http://voice.l.google.com.300INA74.125.225.46 voice.l.google.com http://voice.l.google.com.300INA74.125.225.33 voice.l.google.com http://voice.l.google.com.300INA74.125.225.32 voice.l.google.com http://voice.l.google.com.300INA74.125.225.41 voice.l.google.com http://voice.l.google.com.300INA74.125.225.38 voice.l.google.com http://voice.l.google.com.300INA74.125.225.35 voice.l.google.com http://voice.l.google.com.300INA74.125.225.39 voice.l.google.com http://voice.l.google.com.300INA74.125.225.40 voice.l.google.com http://voice.l.google.com.300INA74.125.225.34 voice.l.google.com http://voice.l.google.com.300INA74.125.225.37 (ie
Re: [asterisk-users] Google voice with no voice
If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com http://voice.l.google.com.300INA74.125.225.36 voice.l.google.com http://voice.l.google.com.300INA74.125.225.46 voice.l.google.com http://voice.l.google.com.300INA74.125.225.33 voice.l.google.com http
Re: [asterisk-users] Google voice with no voice
Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network
Re: [asterisk-users] Google voice with no voice
What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more
Re: [asterisk-users] Google voice with no voice
This is incoming, outgoing or idle (no call)? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:21 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI
Re: [asterisk-users] Google voice with no voice
This sounds like a codec issue. Set your verbose to 10 and retry the incoming call. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:26 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice That's idle. If I call from D70 (working scenario) the result of the command is the same. gtalk show channels shows this when I call from D70 (again, working scenario): Channel Jabber ID Resource Read Write Gtalk/+1x@voice.googl +1xx...@voice.google.com srvres-MTAuMjI3 ulaw ulaw When I call google voice, gtalk show channels shows the following: While ringing: *CLI gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw slin 1 active gtalk channel Once I pick up *CLI -- SIP/D70-0004 answered Gtalk/+xxx-2c8e gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw ulaw 1 active gtalk channel The only difference is the WRITE column that changes from SLIN to ULAW On 1/22/13 2:22 PM, Danny Nicholas wrote: This is incoming, outgoing or idle (no call)? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:21 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition
Re: [asterisk-users] Asterisk, Digium phones, and voicemail.
Theoretically you can do this Exten = 370,1,Voicemailmain(D70@default) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk, Digium phones, and voicemail. Hi all, I registered my Digium D70 using a name (D70) instead of a number. Is there a way to program Asterisk (or the phone?) so when I press the MSGS button, it automatically goes to the correct voicemail, with or without asking me for a password ? As of now, it asks me for my mailbox number, which is D70. If I press D70 on the phone (or 370), asterisk does not recognize my phone, of course. Thanks -- _ -- 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] Asterisk, Digium phones, and voicemail.
That one's not in my wheelhouse since I only use Polycom phones. Google D70 Provisioning -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 2:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Asterisk, Digium phones, and voicemail. That worked, thank you. Is there a way to program the keys of the Digiums D70 from asterisk ? Or does everything needs to be done on the phone itself ? On 1/22/13 3:31 PM, Danny Nicholas wrote: Theoretically you can do this Exten = 370,1,Voicemailmain(D70@default) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk, Digium phones, and voicemail. Hi all, I registered my Digium D70 using a name (D70) instead of a number. Is there a way to program Asterisk (or the phone?) so when I press the MSGS button, it automatically goes to the correct voicemail, with or without asking me for a password ? As of now, it asks me for my mailbox number, which is D70. If I press D70 on the phone (or 370), asterisk does not recognize my phone, of course. Thanks -- _ -- 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
Re: [asterisk-users] Capture queue agent drop and put caller back in queue
Not doubting how quickly Nagios can respond, but if the Nagios solution is going to place a call using Asterisk, wouldn’t it be just as efficient (or more so) to depend on Asterisk? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Tuesday, January 22, 2013 3:30 PM To: Benny Amorsen Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Capture queue agent drop and put caller back in queue On Tue, Jan 22, 2013 at 3:01 PM, Benny Amorsen benny+use...@amorsen.dk wrote: Can a Nagios-based solution provide quicker failover than the 90 seconds provided by sip timers or the 10-30 seconds provided by rtptimeout? Certainly; Nagios can detect missed ping responses with a granularity of single seconds. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Capture queue agent drop and put caller back in queue
A qualify value that low would be a resource hog (some phones can't even re-register in 10 seconds). The Nagios solution would require a custom shell, so it would less needy to make the shell be a daemon independent of either. -Original Message- From: Eric Wieling [mailto:ewiel...@nyigc.com] Sent: Tuesday, January 22, 2013 4:12 PM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas Cc: Benny Amorsen Subject: RE: [asterisk-users] Capture queue agent drop and put caller back in queue Using qualify=10 ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Tuesday, January 22, 2013 5:11 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Benny Amorsen Subject: Re: [asterisk-users] Capture queue agent drop and put caller back in queue How do you propose that Asterisk determines that the endpoint has vanished off the network without waiting for a 10-90 second timeout period? On Tue, Jan 22, 2013 at 4:07 PM, Danny Nicholas da...@debsinc.com wrote: Not doubting how quickly Nagios can respond, but if the Nagios solution is going to place a call using Asterisk, wouldn’t it be just as efficient (or more so) to depend on Asterisk? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Integration with Social Media, Email and Web call center
For just the messaging part, you should be able to use wget or curl to interface and create messages. You might have to go a little higher level like C or Perl, but it sounds very doable. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, January 22, 2013 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integration with Social Media, Email and Web call center Dears; Can someone advise me where to find a technology (open source) that let us able to integrate with social media like whatsapp and facebook? And use this in call center (queuing the messages and routing it for agent)? Anyone give me a light to start? Regards Bilal -- _ -- 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] MoH with message on intervals
The simplest way to do it would be to use sox to remix your moh file with the message like this: Let's say you're using the standard file macroform-cold_day.wav. First you split it into two minute segments like so Sox macroform-cold_day.wav seg1.wav trim 0.0 120.0 Sox macroform-cold_day.wav seg2.wav trim 0.0 120.0 Sox macroform-cold_day.wav seg3.wav trim 0.0 120.0 Now put it back together with your message inserted like this: Sox seg1.wav yourmessage.wav seg2.wav yourmessage.wav seg3.wav yourmessage.wav macroform-cold_day.wav -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett Sent: Monday, January 21, 2013 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] MoH with message on intervals I'm talking to somebody who wants to have a recorded message play periodically for people on hold. An example would be interrupting the hold music every two minutes to play a message with business hours and current specials. Seems like you could fake it by breaking the music files into two minute chunks with alphabetical file names, and using sort=alpha. It seems like there might also be possible ways to do in the dialplan with 'Set(CHANNEL(musicclass)=' or a combination of StartMusicOnHold() and StopMusicOnHold(). Can anybody point me in the right direction? -- _ -- 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] How to give users the capability to set CDR userfield for some calls
Since Gosub is technically an application, you should be able to modify this
snippet in features.conf
testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and
callee to play
;;tt-monkeys to the opposite
channel
To this
testfeature = #9,peer,Gosub,play-monkeys,s,1 ;Allow both the caller and
callee to play
;;tt-monkeys to the opposite
channel
And in extensions.conf add
[play-monkeys]
Exten = s,1,playback(tt-monkeys)
Exten = s,n,return()
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, January 18, 2013 3:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give users the capability to set CDR
userfield for some calls
2013/1/17 Kevin Larsen kevin.lar...@pioneerballoon.com
Possibly switch to using subroutines instead of Macros. Macros are being
deprecated in place of subroutines.
Interesting thing to try.
The trouble is I can't find any usable example of calling Gosub routines
from features.conf's application map.
I've found old references explaining that this is not supported but I don't
if it's still valid or not.
Any ex
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From:Olivier oza_4...@yahoo.fr
To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date:01/17/2013 10:29 AM
Subject:[asterisk-users] How to give users the capability to set CDR
userfield for some calls
Sent by:asterisk-users-boun...@lists.digium.com
_
Hello,
To my surprise, with asterisk 1.8 (I've not tried with other versions), it
seems you cannot set CDR's userfield from within a dialplan macro called by
dynamic features.
See :
testfeature = *321,self/callee,Macro,toto
[macro-toto]
exten = s,1,Verbose(0,Into macro-toto with CDR(src) set to ${CDR(src)})
exten = s,n,Set(CDR(userfield)=foobar)
I'm planning to use this feature to let users mark in CDR an ongoing call as
malicious or important or whatever.
Any hint ?
Regards--
_
-- 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
--
_
-- 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] Annoying delay after main server goes down
I think this is a phone problem not an asterisk one. In my experience a SIP (IP) phone takes about 20 seconds to properly negotiate (re)registration (longer for Polycom 501s). The best work-around I could recommend would be to have an intermediate interface like kamailio (sp) that handles the dual registration. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Onur Cem Çelebi Sent: Friday, January 18, 2013 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Annoying delay after main server goes down Hello, we have distributed lots of cisco spa303 IP phones and get them work with Asterisk. I have configured proxy and alternate proxy and enabled dual registration features in provisioning files(xml files). All phones are able to subscribe to both of servers. But the problem is, if main server goes down, i am obliged to wait nearly 20 second in order to place a call over second server. How to get alternate proxy work immediately after first server fails ? Thanks in advance. -- _ -- 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] rtptimeout: how to detect it in dialplan?
As I read it you can do it like this: From http://www.voip-info.org/wiki/view/Asterisk+sip+rtptimeout Exten = s,1,noop (set rtptimeout so we can have 2 timeouts on a dial) Exten = s,n,Set(rtptimeout=60) Exten = s,n,Dial(SIP/peer1,60) Exten = s,n,Dial(SIP/peer2,60) Haven't tested this. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion Sent: Friday, January 18, 2013 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] rtptimeout: how to detect it in dialplan? Hi! I want to forward a call to another destination if the outgoing call leg has an rtptimeout. But as far as I see there is no way to find out if the hangup was due to a rtp timeout or any other reason. I thought that HANGUPCAUSE or DIALSTATUS would be set, but they aren't. Are there any means to detect an rtp timeout in extensions.conf? Thanks Klaus -- _ -- 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] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start
I cloned this ticket for 11.2 https://issues.asterisk.org/jira/browse/ASTERISK-20962 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Wednesday, January 16, 2013 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start On Wed, Jan 16, 2013 at 1:37 PM, Danny Nicholas da...@debsinc.com wrote: Same issue exists with 11.2 I've created issue 20945 to track this, at least for 1.8.20.0. https://issues.asterisk.org/jira/browse/ASTERISK-20945 -- 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] special conference room
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Tuesday, January 15, 2013 6:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] special conference room Hi list, I am in need of a special asterisk conference room with the following constraints: - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how to realize that are highly appreciated.. Thanx in advance, yves Hint 1 - specify your Asterisk version since these capabilities change availabilities between releases. Hint 2 - you can have the moderator enter the conference normally and everyone else enter muted Hint 3 - the moderator should probably have a web interface to accomplish the other tasks Hint 4 - refer to this link - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ConfBridg e -- _ -- 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] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start
Same issue exists with 11.2 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bakko Sent: Wednesday, January 16, 2013 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start me too. regards El 16/01/2013 13:25, Eric Wieling escribió: I am also experiencing this issue. Asterisk is in fact running, you can verify by running asterisk -rvvv (-r connects to an EXISTING asterisk process) or using ps. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Wednesday, January 16, 2013 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the afternoon, I got the notification that asterisk 1.8.20.0 had been released, so today, I downloaded the latest 1.8-current.tar.gz and compiled and installed it (./configure, make menuselect and choose all the same options as my previous install, make, make install). Now, when I start the asterisk service using service asterisk start from the command line, this is the output: [root@pbx ~]# service asterisk start Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Starting asterisk: However, the /var/run/asterisk/asterisk.ctl file is being created and the process is starting: [root@pbx ~]# ls -lh /var/run/asterisk/ total 4.0K srwxr-xr-x 1 root root 0 Jan 16 12:07 asterisk.ctl -rw-r--r-- 1 root root 6 Jan 16 12:07 asterisk.pid However, I'm no longer getting the usual splash message when I connect to the asterisk console...this is what I get: [root@pbx ~]# asterisk -r Verbosity is at least 3 pbx*CLI I don't have any peers setup yet, or even any dialplan configured to test, but when I go through the logs, I don't find any errors or warnings that I'm not expecting. I've gone back to the asterisk 1.8.19.1 install and everything works as expected (no error messages, full splash about license / version on connection to console, etc). I performed make clean in my 1.8.20 source directory, then ./configure, make menuselect, make, make install, and even make config, and I'm still seeing this message pop up when restarting / starting the service. I went through the CHANGELOG.TXT for 1.8.20.0 and it appears there are some items talking about changing the way the process starts up (commit r376428), but I'm not enough of a coder to understand if those would cause what I'm seeing. Is anyone else seeing this issue? Should I open an issue on the tracker? Anyone see something obvious I missed? -- 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 -- _ -- 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] N Priority in Mysql
n priority is a runtime value set by the dialplan. To use it in a database, you would have to update the database using something like dialplan show context. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roy Abshire Sent: Wednesday, January 16, 2013 3:44 PM To: Asterisk Users Subject: [asterisk-users] N Priority in Mysql Why doesn't the n priority work in a mysql database?? This way I don't have to re-number everything when I insert a new line... -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) -- _ -- 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] special conference room
From what I read, neither confbridge or meetme have the whisper feature built-in; This doesn't matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web interface. Let's say Yves' special conference is . The moderator would start using this command Exten = s,1,meetme() The participants would do Exten = s,1,meetme(,m) - muted so they can listen but not talk - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator The moderator would need to be able to enumerate the conference by doing Asterisk -rx core show channels verbose|grep meetme This is supposed to be doable from the dialplan but my google-fu failed me on it. - the moderator must be able to mute and unmute any caller at any time Establish a maximum number of users and set this up for each one Exten = 99,1,meetmeadmin(,M,1) let user 1 talk Exten = 199,1,meetmeadmin(,m,1) turn user 1 back off - the moderator must be able to talk to all callers or to a specific caller. Exten = 901,1,chanspy(SIP/XXX,w) - the modetator must be able to kick off any caller at any time... Exten = 299,1,meetmeadmin(,k,1) kick out user 1 Exten = 666,1,meetmeadmin(,K) shut it down From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Wednesday, January 16, 2013 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] special conference room Sounds like a conference with all attendees permanently muted (except the moderator). The moderator uses whisper to communicate with individuals. --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Wednesday, January 16, 2013 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] special conference room barat and danny, thank you for your input... I am using asterisk 11.2 and i read about meetme. Yes, it has many switches and options and can help me a lot... but as you already said... does _almost_ all features.. unfortunately I need ALL the constraints fulfilled... therefore i admit I have not tried it in deep, because just from reading the doc I realized, that it wont fit all my needs... btw.: I understood the mute switch to disable the callers to talk to the conference.. (so to say it mutes the callers microphone, not his earphones am I wrong? nevertheless... any more hints for my original feature-request? thank you all, yves Am 16.01.2013 19:03, schrieb Bharat Lalcheta: Please study meetme application's options. You will get almost all feature you ask for in it On Jan 16, 2013 5:37 AM, Yves A. yves...@gmx.de wrote: Hi list, I am in need of a special asterisk conference room with the following constraints: - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how to realize that are highly appreciated.. Thanx in advance, yves -- _ -- 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
Re: [asterisk-users] special conference room
You could set up the caller meetme where the user presses 1 to Exit the conference Whisper to the moderator Rejoin the conference From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Wednesday, January 16, 2013 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] special conference room ok, now i have got some very valuable information to start off with. thank you all. i´ll be back to report success or further questions... just one thing, that i think might be a showstopper that i may have not explained clear enough...: muting and unmuting a caller should have the effect, that the caller can talk to the moderator or not... any caller should NEVER hear what other callers are talking... may he be muted or not... yves Am 16.01.2013 23:01, schrieb Danny Nicholas: From what I read, neither confbridge or meetme have the whisper feature built-in; This doesnt matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web interface. Lets say Yves special conference is . The moderator would start using this command Exten = s,1,meetme() The participants would do Exten = s,1,meetme(,m) muted so they can listen but not talk - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator The moderator would need to be able to enumerate the conference by doing Asterisk rx core show channels verbose|grep meetme This is supposed to be doable from the dialplan but my google-fu failed me on it. - the moderator must be able to mute and unmute any caller at any time Establish a maximum number of users and set this up for each one Exten = 99,1,meetmeadmin(,M,1) let user 1 talk Exten = 199,1,meetmeadmin(,m,1) turn user 1 back off - the moderator must be able to talk to all callers or to a specific caller. Exten = 901,1,chanspy(SIP/XXX,w) - the modetator must be able to kick off any caller at any time... Exten = 299,1,meetmeadmin(,k,1) kick out user 1 Exten = 666,1,meetmeadmin(,K) shut it down From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Wednesday, January 16, 2013 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] special conference room Sounds like a conference with all attendees permanently muted (except the moderator). The moderator uses whisper to communicate with individuals. --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Wednesday, January 16, 2013 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] special conference room barat and danny, thank you for your input... I am using asterisk 11.2 and i read about meetme. Yes, it has many switches and options and can help me a lot... but as you already said... does _almost_ all features.. unfortunately I need ALL the constraints fulfilled... therefore i admit I have not tried it in deep, because just from reading the doc I realized, that it wont fit all my needs... btw.: I understood the mute switch to disable the callers to talk to the conference.. (so to say it mutes the callers microphone, not his earphones am I wrong? nevertheless... any more hints for my original feature-request? thank you all, yves Am 16.01.2013 19:03, schrieb Bharat Lalcheta: Please study meetme application's options. You will get almost all feature you ask for in it On Jan 16, 2013 5:37 AM, Yves A. yves...@gmx.de wrote: Hi list, I am in need of a special asterisk conference room with the following constraints: - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how to realize that are highly appreciated.. Thanx in advance, yves -- _ -- 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
Re: [asterisk-users] block one number in incoming calls
I would suggest this Exten = _0666XX,1,answer() Exten = _0666XX,n,playback(tt-monkeys) Exten = _0666XX,n,hangup() You could just hangup on them, but playing the screeching monkeys will get the message to them to leave you alone. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, January 14, 2013 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] block one number in incoming calls Hello list could you please help me about one question. i have asterisk 1.4 installed, i configure the inbound call in my asterisk like below. exten = 520xx,1,Dial(SIP/224, 30). when the customer call my number (520xx) the sip phone 224 works without issue my problem i have a lot of calls coming from this number (0666xx) and i want to block it. if you can give me an example please . thanks and regards -- _ -- 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] block one number in incoming calls
Reverse the 3:4 and you will have the desired effect.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine
Elharit
Sent: Monday, January 14, 2013 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] block one number in incoming calls
hi Zohair Raza
thanks for your replay but this script will allow just this 0666XX to
call my number 520xx what i want is block this number to call 520xx
not allow it
thank you
exten = 520xx,1,NoOp(Caller-ID: ${CALLERID(all)})
exten = 520xx,2,GotoIf($[${CALLERID(num)} = 0666XX ]?3:4)
exten = 520xx,3,Dial(SIP/224, 30)
exten = 520xx,4,hangup
2013/1/14 Salaheddine Elharit salah.elharit...@gmail.com
thanks danny
i think i didn't explain correctly may question
i revive a lot of calls from this number _0666XX and i wants to block it
to call my number 520xx .
2013/1/14 Danny Nicholas da...@debsinc.com
Exten = _0666XX,1,answer()
Exten = _0666XX,n,playback(tt-monkeys)
Exten = _0666XX,n,hangup()
--
_
-- 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] block one number in incoming calls
Yes. This is referred to in the documentation as ex-girlfriend logic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of isr...@gmail.com Sent: Monday, January 14, 2013 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] block one number in incoming calls Can't we. Do this? exten = 520xx/0666XX,1,hangup -Original Message- From: Salaheddine Elharit salah.elharit...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 14 Jan 2013 16:51:11 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] block one number in incoming calls -- _ -- 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
Re: [asterisk-users] block one number in incoming calls
If you're worth the trouble to change my dialplan for, you should suffer some torture for calling me. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Monday, January 14, 2013 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] block one number in incoming calls So I'm not the only one who uses the monkeys as our place to send bad calls to. -- Sent from my iPhone On Jan 14, 2013, at 10:02 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Monday 14 January 2013, Salaheddine Elharit wrote: i think i didn't explain correctly may question i revive a lot of calls from this number _0666XX and i wants to block it to call my number 520xx . Use something like Exten = _520X./0666XX,1,Answer() Exten = _520X./0666XX,n,PlayBack(tt-monkeys) Exten = _520X./0666XX,n,HangUp() Now when a call comes in from 0666XX to _520X. they will get monkey noises. -- AJS Answers come *after* questions. -- _ -- 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
Re: [asterisk-users] Set Language for VoiceMailMain
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted to
answer the phone in English, then do voicemails in different languages, this
should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) - redundant since default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
--
_
-- 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] Which tool to edit custom reports from CDR and queues logs ?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Friday, January 11, 2013 4:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Which tool to edit custom reports from CDR and queues logs ? Hi, I would like to edit reports showing how fast operator and users answer incoming calls. Users are spread over 6 locations, each with its own asterisk instance. Operator is on main site. Users have casual extension but operator logs as queue agent. I've read or/and tried Star2Billing's CDR-Stats and A2Billing, Asternic Call center Stats,. I'm wondering if using a BI tool such as Jasper Reports would be preferable. Which tool would you suggest to build custom reports ? Regards My exposure to these tools seems to indicate that they are all SQL/MYSQL based engines. To combine the data from six sites my inclination would be to use something like Crystal Reports or Perl to roll my own (or possibly even as low tech as Excel). -- _ -- 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] Set Language for VoiceMailMain
No. It is purposely set from the dialplan. In Asterisk 11.X you have the
[zonemessage] section in voicemail.conf that could probably be tweaked to
change the language without dialplan changes. Also in sip.conf you can set
language by peer so you could have something like
[London]
Type = peer
Language=en
[Madrid]
Type=peer
Language=es
[paris]
Type=peer
Language=fr
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of mailbox and
VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted to
answer the phone in English, then do voicemails in different languages, this
should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) - redundant since default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
--
_
-- 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] Set Language for VoiceMailMain
Since the peer language sets CHANNEL(language), I can say yes with
reasonable certainly. Like anything else here, you don't really know until
you try it on your box.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Hello,
are you sure that the language-parameter of the SIP peer will influence
the language used by VoiceMailMain() ?
Jonas.
On 01/11/2013 04:07 PM, Danny Nicholas wrote:
No. It is purposely set from the dialplan. In Asterisk 11.X you have the
[zonemessage] section in voicemail.conf that could probably be tweaked to
change the language without dialplan changes. Also in sip.conf you can set
language by peer so you could have something like
[London]
Type = peer
Language=en
[Madrid]
Type=peer
Language=es
[paris]
Type=peer
Language=fr
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of mailbox and
VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted to
answer the phone in English, then do voicemails in different languages, this
should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) - redundant since default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
--
_
-- 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
Re: [asterisk-users] Set Language for VoiceMailMain
Tried it just now and that is indeed the way it works (100% for me).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Well, I thought you had tried it and thus could tell it with 100% certainty.
Thanks for your help.
Jonas.
On 01/11/2013 04:16 PM, Danny Nicholas wrote:
Since the peer language sets CHANNEL(language), I can say yes with
reasonable certainly. Like anything else here, you don't really know until
you try it on your box.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Hello,
are you sure that the language-parameter of the SIP peer will influence
the language used by VoiceMailMain() ?
Jonas.
On 01/11/2013 04:07 PM, Danny Nicholas wrote:
No. It is purposely set from the dialplan. In Asterisk 11.X you have the
[zonemessage] section in voicemail.conf that could probably be tweaked to
change the language without dialplan changes. Also in sip.conf you can set
language by peer so you could have something like
[London]
Type = peer
Language=en
[Madrid]
Type=peer
Language=es
[paris]
Type=peer
Language=fr
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of mailbox and
VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted to
answer the phone in English, then do voicemails in different languages, this
should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) - redundant since default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
--
_
-- 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
--
_
-- 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] How often to restart Asterisk...
The general rule seems to be, don't restart it unless there's a problem or you hear of memory leaks. I had a version of 1.4 that I restarted every night because I read about memory leaks, but I hear of 1.2 installs that have been running continuously for 10 years. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen Sent: Friday, January 11, 2013 3:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How often to restart Asterisk... Had my Asterisk instance stop responding to incoming/outgoing calls today. Had to kill -9 the asterisk process and restart it to get it back. Not really looking for help on that as the instance is version 1.6 and is due to be replaced with an upgraded version shortly. However, this does make me wonder, do you restart periodically to try to avoid issues or do you just let things run until there is a problem? This box had 119 days of up time on the Asterisk process. I have a client that I installed an Elastix instance on and the last time I checked it, it was up to almost 500 days of up time without an asterisk restart. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 -- _ -- 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] Your thoughts and opinions on Asterisk 11 for production use
I don't presently have 11 in production, but in each case where I've put 11 in on top of 10.X the process has been relatively seamless, so I expect my 10.X boxes will go to 11.X sometime this year. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, January 10, 2013 8:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Your thoughts and opinions on Asterisk 11 for production use Hi, Have you experienced Asterisk 11 in production ? What do you think of it ? Which libpri version, if any, did you then associate with Asterisk 11 ? Regards -- _ -- 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] Streaming/Recording audio
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian Sent: Wednesday, January 09, 2013 4:25 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Streaming/Recording audio Regarding the streaming of audio. I thought of another approach, but I'm not sure if Asterisk will allow it. When playing a file they're read from /var/lib/asterisk/sounds/en/. Is it possible to change this directory to a network directory hosted on a windows environment? This is hypothetically possible. If I say Playback(foo) in the dialplan it will expect foo.wav (or .gsm or .sln44 or .ulaw depending on the codec) to be in /v/l/a/s/e. If I say Playback(/tmp/foo) it will expect the same file to be in /tmp. So if I can make my windows environment file be addressable as /windows I can say Playback(/windows/foo). The better way would be to do a samba script to copy the file from windows to /tmp so you don't have lag issues or other problems. -- _ -- 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] AstDB with Sqlite
The Dialplan DB commands still work using the old Berkley conventions. If
you want to use the sqlite for functions other than AstDB, then you have to
resort to SQL SELECTs and INSERTs. If you need that kind of functionality,
you will probably save yourself some headaches by just using MYSQL.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Robinson
Sent: Wednesday, January 09, 2013 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AstDB with Sqlite
Are you sure that it is correct for sqlite?
For what i know it should contain select or inser operator in dialplan.
Correct if i am wrong.
On Oct 19, 2012 9:18 AM, Danny Nicholas da...@debsinc.com wrote:
Just use the standard dialplan db commands - Read
Set(TESTOP=${DB(Nightop/ext)}) write Set(DB(Nightop/ext)=107). It's not as
robust as Mysql or postgres but does seem to do better than the old Berkley
database.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Robinson
Sent: Thursday, October 18, 2012 10:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AstDB with Sqlite
As you may know, asterisk version 10 and high use sqlite. Are any examples
or documentation how to use in dialplan?
--
_
-- 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] IVR platform for a mobile operator
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of luke devon Sent: Wednesday, January 09, 2013 9:06 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] IVR platform for a mobile operator Hi Friends , I want to setup a IVR platform using asterisk to a mobile operator. Can somebody give me some guides with recommended hardware types ? Thank you Luke. IMO you will be happiest with a SIP trunk handling this as there can be horrible latency in DAHDI/Mobile connections. -- _ -- 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] IVR platform for a mobile operator
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitul Limbani Sent: Wednesday, January 09, 2013 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR platform for a mobile operator On Jan 9, 2013 8:38 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of luke devon Sent: Wednesday, January 09, 2013 9:06 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] IVR platform for a mobile operator Hi Friends , I want to setup a IVR platform using asterisk to a mobile operator. Can somebody give me some guides with recommended hardware types ? Thank you Luke. IMO you will be happiest with a SIP trunk handling this as there can be horrible latency in DAHDI/Mobile connections. Latency on DAHDI - heard this for first time TDM networks have zero latency we face latency only on IP (SIP) networks. Mitul Limbani DAHDI in my reference should be construed as copper POTS, not PRI/T1 trunks. -- _ -- 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] IAX2 support of video
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, January 07, 2013 6:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX2 support of video According to this: https://wiki.asterisk.org/wiki/display/AST/Video+Telephony yes. I have a local server with two video phones - running SIP to each phone. Works. Then I have an IAX2 connection from that local machine to another machine. then a SIP connection from that machine to another machine where the same model video phone is in use. A call to that phone does not show video only audio. All machines have in sip.conf:videosupport=yes Is there something else to get SIP/IAX2/SIP video call to work? Thanks Jerry Make sure you have the H.26X codec enabled at all points. -- _ -- 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] Streaming/Recording audio
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian Sent: Tuesday, January 08, 2013 9:24 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Streaming/Recording audio Hello Users, I've been searching for a couple of hours now but I can't find the answers to my questions, so here they go: 1) Is it possible to stream audio files from a webserver during a call by configuring this in the dialplan? Something like Playback(http://myserver.companynetwork/welcome.alaw)? 2) After recording is finished using the Record application, is the recorded file(the audio stream) accessible to send off to an http handler using HTTP POST? I hope someone could help me out. Thanks, Grant 1 . AFAIK, playback is just for local files, not web files. There are options like mpg123 to play streaming audio files - see http://nerdvittles.com/?p=92 2. As long as you save the audio stream in a normal format, this shouldn't be a problem. -- _ -- 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 unit suggestions
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Monday, January 07, 2013 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Paging unit suggestions We currently have an Asterisk system that is hooked up to our old paging speakers via sound card, plugged into two amps. Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both warehouses are around 30k square feet. Both have a large number of printing presses. The computer system is that is running Asterisk is around 10 years old and starting to fail. I'm looking to replace both the system and hopefully move to a IP paging system, but wanted to reuse the current speakers. I'm looking for suggestions on a IP based amp or similar that could drive the current speakers? I was envisioning a unit that would register as a SIP extension then would handle auto-answer that I could send a sound file to. Thanks for any help! Doug This may be insane, but it seems from what I read that you could replace most 10 year old boxes with a $35 Raspberry Pi. That not being the case, the speakers could probably be adapted to work off of any SIP phone headset/handset. -- _ -- 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] IAX2 support of video
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, January 07, 2013 3:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] IAX2 support of video Does IAX2 support a video call ? According to this: https://wiki.asterisk.org/wiki/display/AST/Video+Telephony yes. -- _ -- 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 for Praying
Whether you use .call file or AMI, you should still do the call/page using a context and that context run the PHP script. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Monday, January 07, 2013 4:30 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Paging for Praying Thanks for the help and it seems I deleted some of my emails by mistake ! I am sorry if I repeated my question. As I see that the call file is used to generate calls, can I use this technique to page the Phones? It is one wave file only that need to be Paged for all the Phones connected on the Asterisk PBX. When I say Paging, I mean that they are going to hear the sound from the speaker (without pickup the handset). By using AMI, then I can build PHP script that will use the AMI to do the Page? Thanks and Regards Bilal -- A call file is a text file that you create. The format is very specific. On Tue, 1 Jan 2013, bilal ghayyad wrote: * How can I know this format? Because I need to know what should I place in this file so it will execute Paging for this group of Phones? This may help: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out How many customers will be receiving these reminders? * It is required that all the employers at the company to hear this on their IP Phones. In my experience, you can't just dump xxx call files into the outgoing directory. If you expect more than a dozen or so, you'll have to move them in blocks as they are processed. Another good reason to use AMI. You can 'schedule' a call file to be processed in the future by setting the file's 'mtime.' * Can you explain for me please? Create a file named fajr containing: application: playback channel: sip/bilal data: fajr-in-10-minutes Copy the file to a directory we assume is on the same file system as /var/spool/asterisk/outgoing/: cp\ fajr\ /var/spool/asterisk/tmp/ Set the file's 'mtime' touch\ --date='now + 2 minutes'\ --time=mtime\ /var/spool/asterisk/tmp/fajr Move it to the outgoing directory: mv\ /var/spool/asterisk/tmp/fajr\ /var/spool/asterisk/outgoing/ Your phone should ring in about 2 minutes. You may want to look into setting 'auto-answer' or some sort of 'overhead paging' with a very discreet sound file like a short, single beep. Please consider AMI if you are looking for a robust service. -- Thanks in advance, -- --- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline -- _ -- 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] faxdetect on/off on the fly?
Don't think you actually can, per se. What you can do is set a variable and
redirect to the line that has this defined or undefined.
Let's say that you have 4 lines; SIP/1001 and DAHDI/1 have faxdetect=yes
defined in sip.conf and chan_dahdi.conf. SIP/1002 and DAHDI/2 have
faxdetect=no in the respective places. Simple Perl AGI to set dialplan
variable:
cat selline.pl
#!/usr/local/bin/perl
#
#
# Send USEFAX to Dialplan
#
#
use strict;
use warnings;
require Asterisk::AGI;
# turn off I/O buffering
$| = 1;
# the AGI object
my $agi = new Asterisk::AGI;
my %input = $agi-ReadParse();
print STDOUT SET VARIABLE USEFAX \ON\ \n;
print STDOUT SET VARIABLE USEFAX \OFF\ \n;
exit;
In the Dialplan
Exten = s,1,Answer()
Exten = s,n,AGI(selline.pl)
Exten = s,n,Gotoif($
exten = s,1,Gotoif($[${USEFAX} = ON]?on:off)
exten = s,n(on),Dial(SIP/1001)
exten = s,n,hangup
exten = s,n(off),Dial(SIP/1002)
exten = s,n,hangup
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Friday, January 04, 2013 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] faxdetect on/off on the fly?
Hi Danny,
Can you please elaborate on how in the dialplan we can set faxdetect on and
off?
We currently have it set on in sip.conf.
Thanks.
On 3 January 2013 17:21, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Thursday, January 03, 2013 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] faxdetect on/off on the fly?
Hello,
We want the ability to choose from an AGI script whether or not to enable
faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone
suggest a workaround?
Thanks for any advice.
You should be able to call the AGI and set a dialplan variable and use
Gotoif to do/not do faxdetect. Reading the .sample files for 11.0 it seems
that normally these are configured until restart/reload but with a little
testing, the default should be overrideable.
--
_
-- 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
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
--
_
-- 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] Calender and EWS with shared calenders
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus Löfqvist Sent: Friday, January 04, 2013 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Calender and EWS with shared calenders Hi all, We want to use res_calender_ews to close users extensions if there are busy in there exchange calenders. It is possible to use shared calenders ? Ie, I have a resource user that have access to the users calenders, so I dont need to maintain every users password/username in asterisk. We are today running 1.8 but are going to upgrade to 11... Best regards / Magnus If it works for you in 1.8 it should work in 11.X the only change is the $Revision number. -- _ -- 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] DAHDI: How to know since when it is used? How to shutdown after max time?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, January 03, 2013 2:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI: How to know since when it is used? How to shutdown after max time? 2013/1/3 bilal ghayyad bilmar...@yahoo.com Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected (stuck) for long time. I know how to write the extensions.conf in a way to handle the hangup properly, also I send the incoming calls to the voicemail to be sure it is hanged up properly. One more thing, I set the rtptimeout in case there is any problem in the sip phone and the network .. But, still after sometime, I am surprised that some channels are stuck and stayed connected and then I have to reset it manually !! This is happening only in the analoge channels. What other than the rtptimeout, the hangup in the extensions.conf, the voicemail? Is there anything I have to take care for it that might cause this stuck and keeping the channel openned? By the way, for such cases, what should I place the value of the rtpkeepalive as currently it is 0? What other things I have to take care for it? Regards Bilal I checked on my PBX and I find no way to identify the duration of a call involving a DAHDI channel like it happens on SIP channels. I think the only way will be to assign a not so huge AbsoluteTimeout to each call. My suggestion would be to either do a cron job that executes asterisk -rx core show channels verbose and kill anything with a duration over 90 minutes or do the same thing with an AMI task (cron optional here). -- _ -- 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] Moving User Agent To Remote Location
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, January 03, 2013 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Moving User Agent To Remote Location Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such * nat (yes): No problem here either * defaultuser (1...@example.com): Does the @example.com have to point to the UA (i.e., (1003@ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing host=dynamic takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is host=dynamic sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick. I'm going to vote for the RTP issue. If you are establishing a call but have no audio, you are getting the 5060 port, but not the 1-2 range that RTP normally expects. A better practice is to allocate 4 ports per line you expect to use in rtp.conf (1-2 would allow 2500 lines; much more that most folks need and more holes to monitor). -- _ -- 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] Moving User Agent To Remote Location
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Thursday, January 03, 2013 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving User Agent To Remote Location On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 The rtpend should be 10008 and rtpstart should be 10005. A SIP call in Asterisk originates on 5060 (5061 for TLS) and then spawns two RTP channels for audio. AFAIK the odd channel is send and the even channel is receive (smarter folks than me like Tzafir can give you the specifics; this was covered at least twice in 2012 threads). If you open 5060 on your NAT/firewall, but open no RTP channels, you will establish a call with no sound. -- _ -- 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] Moving User Agent To Remote Location
Just for grins, run netstat -anp on the call using just Asterisk and then again with OpenSIPS in the mix. It sounds like OpenSIPS or your RTPproxy is block the audio channels. -- _ -- 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] faxdetect on/off on the fly?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Cunningham Sent: Thursday, January 03, 2013 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] faxdetect on/off on the fly? Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. You should be able to call the AGI and set a dialplan variable and use Gotoif to do/not do faxdetect. Reading the .sample files for 11.0 it seems that normally these are configured until restart/reload but with a little testing, the default should be overrideable. -- _ -- 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] Dialing out and recording
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,.)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
--
_
-- 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] Users list email totals by year .
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Wednesday, January 02, 2013 7:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Users list email totals by year . So where has every body else gone? Still here, but mature working systems, still running 1.4.x Doug As the thread said earlier (I think it was Shaun), the response mechanism has moved a good bit into the forums. The users list still is functional for folks who want to contribute but don’t keep a browser window open to monitor the forums. P.S. since the world has now turned twice, Happy New Year to anyone reading. -- _ -- 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] Dialing out and recording
Henrik Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,.)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
Danny Nicholas
Sent: Wednesday, January 02, 2013 8:18 AM
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 02, 2013 9:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and we
announce the call before the transfer is complete. The carrier requires a
diversion header if the ANI is not one of our DIDs. Does someone have
experience with this working?
--
Two suggestions for you, Don. #1 if the Dial is Private the
announcement is taken care of. #2 I'm supposing that you could do a SIP
Header command before the Dial to resolve the diversion header issue.
--
_
-- 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] Top Posting
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, January 02, 2013 9:54 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Wednesday, January 02, 2013 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen benny+use...@amorsen.dk wrote: Gergo Csibra csi...@gmail.com writes: Complaining about top posting on a list where's no moderation, no sanction if somebody top posting is pointless. There is a sanction. People like me will score top posters lower and soon not see their posts at all. I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. -- I personally don't give a rat's rear whether it's at the top or the bottom; If it's relevant, I'll read it and if not it goes to file-13. Quit picking on Outlook and Blackberry users (no, keep it up, the list need the volume?) -- _ -- 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] Top Posting
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner Sent: Wednesday, January 02, 2013 10:00 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Top Posting I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. Wouldn't need to if people trimmed their posts properly. Precisely (e.g., see above)! Indeed, my sense is that top-posting *discourages* properly trimming email and that's my main reason against it. If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. -- Good point. I've found myself having to edit and trim replies to poorly constructed conversations in the past because we got to the N'th iteration using either or both formats. -- _ -- 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] Top Posting
Grow up, follow the rules, have a good day. JohnM PS. Did not intend to imply that it was Steve that hijacked the thread, in case anyone read my comment that way JohnM Steve has waded through enough of these that he should be a hijacker. -- _ -- 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 as answering machine
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler Sent: Wednesday, January 02, 2013 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk as answering machine I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it would be dispatched through the normal process. Currently, as soon as the analogue phone rings, the Asterisk PBX has already answered the call and starts the You have reached. Dial and tries to dispatch the call. This makes it hard to carry on a conversation. Ron In your dialplan, put Wait(10) in front of Answer(). This will give the human 4 rings to pick up before Asterisk does. -- _ -- 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] Dialing out and recording
1.6.2 is a deader soldier than 1.4.X.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 3:20 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dialing out and recording
#2 works for me on Asterisk 1.8.12 when setting the header like this:
exten = _S,n,SipSetHeader(Diversion: ${CALLERID(rdnis)})
I haven't been able to make it work on 1.6 yet though, has anyone else?
/Henrik
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 02, 2013 9:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination
and we announce the call before the transfer is complete. The carrier
requires a diversion header if the ANI is not one of our DIDs. Does
someone have experience with this working?
--
Two suggestions for you, Don. #1 if the Dial is Private the
announcement is taken care of. #2 I'm supposing that you could do a
SIP Header command before the Dial to resolve the diversion header issue.
-- next part -- An HTML attachment was
scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20130102/
459
43b1f/attachment-0001.htm
--
--
_
-- 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] Auto ban IP addresses
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Wednesday, January 02, 2013 4:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Auto ban IP addresses On Wed, Jan 2, 2013 at 3:49 PM, Frank fr...@efirehouse.com wrote: Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18 mailto:sip%3A100@108.161.145.18 ;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? http://www.fail2ban.org/wiki/index.php/Asterisk Fail2ban is a nice program, but deny=108.161.145.18 in sip.conf should satisfy OP's request. -- _ -- 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] Top Posting
*** *** Until Monopolysoft fixes Outlook, I think we should Middle Post - Happy New Year to New Zealand! *** *** -- _ -- 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] Top Posting
My assembler may be a little bit rusty, but wouldn't -1 against rule #5 = +1 for rule #5? -- _ -- 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] Top Posting
Wouldn't the answer to that violate family forum rules (see Charlie Sheen jokes) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Monday, December 31, 2012 11:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Top Posting Tongue firmly in both cheeks? How do you do that? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes Sent: Monday, December 31, 2012 3:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting We should all top *AND* bottom post! On 31 Dec 2012, at 06:03, isr...@gmail.com wrote: Just my pitch in to post From a blackberry you can only top post there is no way of bottom posting So if I would have to wait to get to a computer to bottom post I would just never answer We should all top *AND* bottom post! (tongue firmly in cheek here..) S Tongue firmly in both cheeks? How do you do that? -- _ -- 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] Delaying retry since we're currently running
My best guess is that you are creating the .call file with permissions that don’t allow Asterisk to delete it when it is finished or retries have been exhausted. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Necati Demir Sent: Friday, December 28, 2012 7:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Delaying retry since we're currently running Hi, I am making 200 call concurrently via call files. But i get these messages in asterisk logs: Delaying retry since we're currently running Also, in call files i have the following lines: DelayedRetry: 28662 0 (1356701828) DelayedRetry: 28662 0 (1356702128) DelayedRetry: 28662 0 (1356702428) I set MaxRetries: 0. I did not understand the problem, any idea? -- Necati DEMİR -- _ -- 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 with Cisco 887M
Shouldn't be difficult. You're just setting up the Cisco box as a SIP gateway. Here's a link to get you started. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b06gtw ay.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Edwin Quijada Sent: Thursday, December 27, 2012 10:10 AM To: Asterisk Asterisk Subject: [asterisk-users] Asterisk with Cisco 887M Hi! I am installing asterisk with my ISP but he give me a Cisco 887M router to use for SIP conection. My problem is that I dont know how to link Asterisk with this device because I dont have user/pass to use. Anybody has a cluee to use CISCO 887M with Asterisk ? Thks! -- _ -- 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] stop log/debug messages into /var/log/messages
If it is writing to /v/l/m, then it is coming from somewhere else. All Asterisk messages go to /v/l/asterisk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, December 27, 2012 3:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stop log/debug messages into /var/log/messages please refer logger.conf under /etc/asterisk and stop messages log for full. On Thu, Dec 27, 2012 at 2:43 PM, [Digital^Dude] R millennium@gmail.com wrote: I disabled all logger channels but still it logs to /var/log/messages. Any hints? -- _ -- 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] Paging for Praying
I would set up 5 shell files called pray1.sh, pray2.sh, etc and then set up 5 entries in /etc/crontab to run them at the specified time daily. The file pray1.sh should look something like this: #!/bin/sh cp /pray1/*.call /tmp mv /tmp/*.call /var/spool/asterisk/outgoing the entry in /etc/crontab would look like this 0 8 *** root /usr/bin/pray1.sh This would run pray1.sh at 8 am daily. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharat Lalcheta Sent: Thursday, December 27, 2012 2:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging for Praying I dont think this is existed. However, its easy to build a script in php or perl or any other language which check time from file or database and generate call file which execute paging in asterisk. Just put this script in cron. Thats it... Regards, Bharat Lalcheta On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; How can I have Paging on Asterisk to call for pray? The pray is 5 times in a day and there is a timing for pray (actually it can be existed in a text file or database for the next 2 or 5 years). My question is compound from two parts: How can I have Automatic Page? The automatic page should happens by reading the time and check if the time is same as this time, then do the Page. How? Is it by cron? Someone told me that do a cron that call a script which will check the time, if the time came to do th Page, then do a Page. But really I do not know how this can be done and I do not know if this is already existed? Regards Bilal -- _ -- 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 -- Bharat Lalcheta -- _ -- 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] How do *you* test your changes to dialplans ruled by GotoIfTime?
The simplest way to address this kind of change is to test it a week
(month) or so in advance on your test machine (we have VM's set up to mock
our live machines). A protection against last minute changes is to have
this kind of thing controlled by variables so you can possibly even avoid
dialplan changes by controlling the variables with an AGI script.
In your case, the dialplan could have been written like this:
; Christmas
Exten = s,1,Set(christday=25)
Exten = s,n,Set(eveday=24)
Exten = s,n,Set(boxday=26)
Exten = s,n,Set(christmon=Dec)
Exten = s,n,set(christopen=9:30)
...
; exten =
821192,n,GotoIfTime(${christopen}-${christclose},*,${christday},${christ
mon}?ivr-lightspeed-tech-early,s,1)
exten =
821192,n,GotoIfTime(${eveopen}-${eveclose},*,${eveday},${christmon}?ivr-
lightspeed-day,s,1)
exten =
821192,n,GotoIfTime(*,*,${christday},${christmon}?ivr-lightspeed-after-h
ours,s,1)
exten =
821190,n,GotoIfTime(${boxopen}-${boxclose},*,${boxday},${christmon}?ivr-
lightspeed-day,s,1)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Thursday, December 27, 2012 1:46 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How do *you* test your changes to dialplans ruled
by GotoIfTime?
This past holiday weekend has resulted in some real groaners when it comes
to bugs in our dialplan, making obvious the need for some changes in our
procedures.
First, our hours of operation for Christmas Eve, Christmas, Boxing Day and
New Year's Eve had changed with little to no notice. Okay, fine, whatever, I
fix.
Our Christmas Eve hours (made worse by being Monday this year) dialplan was
broken by me misspelling the correct dialplan to go to. Then our Boxing Day
dialplan was broken when I copied and pasted the correct dialplan from one
similar extension number to the other, like this:
; Christmas
; exten =
821192,n,GotoIfTime(9:30-14:00,*,25,dec?ivr-lightspeed-tech-early,s,1)
exten =
821192,n,GotoIfTime(8:00-17:00,*,24,dec?ivr-lightspeed-day,s,1)
exten =
821192,n,GotoIfTime(*,*,25,dec?ivr-lightspeed-after-hours,s,1)
exten =
821190,n,GotoIfTime(9:00-18:00,*,26,dec?ivr-lightspeed-day,s,1)
then failed to notice the problem until it was too late. Of course, that
only applied on Boxing day and couldn't be noticed earlier, either.
I suppose the first problem where I misspelt the dialplan can be solved by
testing the dialplan in another extension and modifying the time to now + 2
minutes. But how can I avoid stupid errors in the extension number, when
testing by definition requires that I change the extension number to and
fro?
This appears to boil down to always remember to test it at the time that
it becomes relevant. But if I was the kind of person who always remembered
to do things at the right time, then there would never be a need for
computers to do jobs like this in the first place.
--
_
-- 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] How do *you* test your changes to dialplans ruled by GotoIfTime?
I would say that the database method has the advantage over GotoIfTime in that it should stay the same between releases. More headache on the front end, but easier once it is up and running. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn Sent: Thursday, December 27, 2012 2:06 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime? We bypass this problem by storing business hours and holidays in a database table. We use an ODBC call to return whether or not to play the day or night greeting based on the database. We also store the name of a custom greeting file to play. The database is fairly easy to manipulate with test data. Mitch On 12/27/2012 01:46 PM, Ernie Dunbar wrote: This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our procedures. First, our hours of operation for Christmas Eve, Christmas, Boxing Day and New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan was broken by me misspelling the correct dialplan to go to. Then our Boxing Day dialplan was broken when I copied and pasted the correct dialplan from one similar extension number to the other, like this: ; Christmas ; exten = 821192,n,GotoIfTime(9:30-14:00,*,25,dec?ivr-lightspeed-tech-early, s,1) exten = 821192,n,GotoIfTime(8:00-17:00,*,24,dec?ivr-lightspeed-day,s,1) exten = 821192,n,GotoIfTime(*,*,25,dec?ivr-lightspeed-after-hours,s,1) exten = 821190,n,GotoIfTime(9:00-18:00,*,26,dec?ivr-lightspeed-day,s,1) then failed to notice the problem until it was too late. Of course, that only applied on Boxing day and couldn't be noticed earlier, either. I suppose the first problem where I misspelt the dialplan can be solved by testing the dialplan in another extension and modifying the time to now + 2 minutes. But how can I avoid stupid errors in the extension number, when testing by definition requires that I change the extension number to and fro? This appears to boil down to always remember to test it at the time that it becomes relevant. But if I was the kind of person who always remembered to do things at the right time, then there would never be a need for computers to do jobs like this in the first place. -- _ -- 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
Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through
#1 I assume you have spandsp installed #2 I'm guessing you got some hints from this thread - https://issues.asterisk.org/jira/browse/ASTERISK-18394 ? -- _ -- 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] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through
Not certain that you actually do. I do know that T.38 can be a dental experience with Asterisk, but some folks have succeeded with it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, December 27, 2012 3:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through I was not aware you needed SpanDSP for T.38 passthrough.. How will that work with the UDPTL packets not going through Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, December 27, 2012 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through #1 I assume you have spandsp installed #2 I'm guessing you got some hints from this thread - https://issues.asterisk.org/jira/browse/ASTERISK-18394 ? -- _ -- 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] Called Party Name between Asterisk systems
It may depend on the asterisk version, but in theory you should just be able to set callerid(num) and callerid(name) before doing the IAX2 dial. You should always specify the Asterisk version you are using as features often change or are available/not available between different releases. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chet W. Stevens Sent: Friday, December 21, 2012 9:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Called Party Name between Asterisk systems I apologize in advance if I do not ask the question well, but, here goes. I have SIP extensions (Digium phones) on one Asterisk system. Once I have set sendrpid=pai in sip.conf, they see the called party name on their displays when calling eachother. Is there any method to extend this capability between Asterisk systems that are connected via IAX2? I do not see this information being passed when I look at the IAX2 debug. Is there something that could be set in the dialplan prior to making the call then retrieved on the other side? I have done extensive searches online, in the documentation, etc. I appreciate your help. Thank you. Chet Stevens -- _ -- 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 11 and DAHDI/i4
The Asterisk 11 part is irrelevant. You need to use an AGI or local call to use the ChanIsAvail function. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 20, 2012 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 11 and DAHDI/i4 It is Action: ExtensionState Exten: 5551212 Context: fubar This will return the status of the dialplan exten hint. and Action: Command Command: ChanIsAvail Parameters: DAHDI/1 says Error No such command ChanIsAvail ChanIsAvail is a dialplan application not a CLI command. It also will not work for what you want in this case. I'm clearly missing something? Quite possibly. :) Richard OK - so what I am trying to do is through the AMI interface ask if channel DAHDI/1 is busy, on hook or available. How do I tell that In the past I simply did a core show channels and see if DAHDI/1 was present. It it was I new it was in use... How do check now in asterisk 11 if the channels are reported as DAHDI/i4 etc... Thanks, Jerry -- _ -- 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 11 and DAHDI/i4
IMO the local channel call should be the lowest overhead option available. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 20, 2012 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 11 and DAHDI/i4 You should just cache the AMI DAHDIChannel event information in your program. If you really must you could use the CLI command pri show channels. However, it is not intended to be repeatedly run for performance reasons. It blocks processing of ISDN messages while it is running. I am not continually logged in to the AMI to catch those events... Can I make a call to a local channel, run some context+ extension, there call ChanIsAvail for the channel I am interested in - but they how do I get that info back to my C program? Also is that a big overhead? jerry -- _ -- 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 11 and DAHDI/i4
Just for grins, do you have a softphone like xlite that you can try the
outgoing call on? I think it's an outgoing issue, not a polycom one.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 20, 2012 1:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 11 and DAHDI/i4
I have a little dialplan context now...
[check-chanisavail]
exten = s,1,ChanIsAvail(${agi_channel}) exten = s,n,System(/bin/echo
${AVAILCHAN} /tmp/${agi_file}) exten = s,n,Hangup()
and a call file:
Channel: Local/s@check-chanisavail/n
Context: check-chanisavail
Extension: s
Priority: 1
SetVar: agi_file=jerry
SetVar: agi_channel=DAHDI/1
To tell me if a channel is busy or not.
When no channels are busy I execute the dialplan above and it correctly gave
me DAHDI/1-1 in my file. As expected.
When I call in from the outside into my asterisk box, then I execute my
dialplan above and I query (DAHDI/1) I get AVAILCHAN = which is what I
expect.
However, if I use a polycom phone to dial out, and then I execute my
dialplan above and I query (DAHDI/1) it says its still available.
Should it not say AVAILCHAN = as I am using that line it is not
available.
Thanks,
Jerry
--
_
-- 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] It's possible a redudant Queue?
In my experience, you should set up two identical queues and configurations. With a little work, you should be able to let server 1 know the phone is in use by server 2 and vice versa. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi Sent: Friday, December 14, 2012 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] It's possible a redudant Queue? Hi all, I have a doubt. I have to create a queue with 3 phones, these phones can be reached via two redudant Asterisk server. I can pass a variable (the sip trunks) to the queue or should I do two queues with the different trunks? Danilo -- -- _ -- 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] Digital accoustics trying to register to asterisk 1.4.43
Please post the sip.conf entry with any confidential data xxx'ed out. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 2:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 I am trying to get a digital accoustics talkmaster to register to asterisk 1.4.43 I am getting the 401 unauthorized. I have host=dynamic I have verified the passwords match What else is there? I dont see any further clues in sip set debug. all it says is using request as basis request What do I try? -- _ -- 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] Digital accoustics trying to register to asterisk 1.4.43
The two things I would try are changing type from friend to peer and sendrpid from no to yes. The no matching peer usually means the device username isn't matching the sip.conf username=. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 [5001] type=friend username=5001 secret=XXX dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw context=incoming host=dynamic canreinvite=no qualify=no trustrpid=yes sendrpid=no nat=no I did notice one more thing: chan_sip.c:17045 handle_request_register: Registration from '5001sip:5001%4010.239.46.200@10.239.46.200' failed for '137.52.88.195' - No matching peer found Why is there no matching peer I have it defined. I shows in my sip show peers? jerry -- _ -- 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] Digital accoustics trying to register to asterisk 1.4.43
This animal might be like the OBI110 box where you set it up in users.conf instead of sip.conf. Something like this: [5001] transfer=yes call-limit=5 registersip=no host = 1.2.3.4 context=default hasvoicemail=no dtmfmode=inband threewaycalling=no hasdirectory=no callwaiting=no hasmanager=no managerread = system,call,log,verbose,command,agent,user,config managerwrite = system,call,log,verbose,command,agent,user,config hasagent = no hassip=yes hasiax=no secret=x nat=no canreinvite=no dtmfmode=rfc2833 insecure=port,invite pickupgroup=1 callgroup=1 disallow = all allow = ulaw,gsm You still do sip reload to get it connected. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 The two things I would try are changing type from friend to peer and sendrpid from no to yes. The no matching peer usually means the device username isn't matching the sip.conf username=. I have tried both friend and peer. I changed the sendrpid to yes and made no difference either. Still get 401 Unauthorized. Jerry -- _ -- 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] Digital accoustics trying to register to asterisk 1.4.43
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 13, 2012 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital accoustics trying to register to asterisk 1.4.43 This animal might be like the OBI110 box where you set it up in users.conf instead of sip.conf. Something like this: [5001] transfer=yes call-limit=5 registersip=no host = 1.2.3.4 context=default hasvoicemail=no dtmfmode=inband threewaycalling=no hasdirectory=no callwaiting=no hasmanager=no managerread = system,call,log,verbose,command,agent,user,config managerwrite = system,call,log,verbose,command,agent,user,config hasagent = no hassip=yes hasiax=no secret=x nat=no canreinvite=no dtmfmode=rfc2833 insecure=port,invite pickupgroup=1 callgroup=1 disallow = all allow = ulaw,gsm You still do sip reload to get it connected. That worked - it registered. Why would it not register the other way? Jerry n It's supposed to work both ways. It depends on how you have it set up on the remote side. It's been two years since I went through the process so it isn't fresh on my brain. -- _ -- 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] disconnect supervision
Could be, but I'd check the easier to fix polarity settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] disconnect supervision Most of the time my phone line are working OK but at time to time when I run: asterisk -rx core show channels it show: Channel Location State Application(Data) SIP/pstn--00 (None) Up AppDial((Outgoing Line)) SIP/pstn-9998-00 7807586576@internal: Up Dial(SIP/97807807586576@pstn-4 2 active channels 1 active call even though nobody is using any line. I'm using Audiocodes gateway. Does it have anything to do with disconnect supervision on analog line in Audiocodes gateway? -- Joseph -- _ -- 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] disconnect supervision
In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and hanguponpolarityswitch lines. If they aren't present, the default values are being used. If they are, tweak them and restart asterisk and dahdi. I do this - service asterisk stop; service dahdi restart; service asterisk start. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 11, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] disconnect supervision On 12/11/12 11:30, Danny Nicholas wrote: Could be, but I'd check the easier to fix polarity settings. How do I do that? Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week. -- Joseph -- _ -- 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] MACRO_CONTEXT equivalent for GoSub
You don't state version, but I'm pretty sure this animal doesn't exist.
What I did in 1.4 was to set a variable before the gosub so I could track
it. Something like this
Exten = s,n,Set(from=foo)
Exten = s,n,gosub(showfoo,s,1)
Exten = s,n,Set(from=bar)
Exten = s,n,gosub(showfoo,s,1)
[showfoo]
Exten = s,1,verbose(called from ${from})
Exten = s,n,return()
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn
Sent: Tuesday, December 11, 2012 3:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MACRO_CONTEXT equivalent for GoSub
Is there an equivalent of MACRO_CONTEXT for a GoSub? Looking for a way
to determine the name of the calling context.
--
Mitch
--
_
-- 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] deadagi on 11 and 1.4
Put a GLOBAL in extensions.conf with the version and use GOTOIF to run AGI/DEADAGI dependent on it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, December 10, 2012 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] deadagi on 11 and 1.4 How can extensions.conf be changed to work with both Asterisk 11 and 1.4.X such that 1.4.X calls deadagi and 11 just calls agi as deadagi is no more. Thanks, jerry -- _ -- 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] Is there an issue with 11.0.2 and registration
Sounds like a registration timeout issue. What does the sip.conf entry look like for these? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, December 10, 2012 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is there an issue with 11.0.2 and registration When you say two, is it two every time? The same two? Is there something different about the two that show this behavior? There isn't enough information in your message. Yes it is the same two devices every time. I have the server running 11.0.2 , I have 8 asterisk devices (1.4.43), I have two polycom phones. They all seem fine. Then I have two other devices (IPSpeakers) that run fine on 1.4.43 and some time after inititally starting 11.0.2 they change from showing the IP address in sip show peers to showing unspecified. They work in the beginning until such time they show unspecified. Then if I stop asterisk 11.0.2 again, and restart it they start working again for some time. Jerry -- _ -- 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 with SIP trunk I've set up between two * boxes.
Does each box show up in the others SIP SHOW PEERS?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request - nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
72.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 --- ACK sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Proxy-Authorization: Digest username=6110, realm=asterisk,
nonce=16883b72, uri=sip:7444@172.17.0.17,
response=b75389c5938b4f185b3d31bd4463abf3, algorithm=MD5
Content-Type: application/sdp
Re: [asterisk-users] BLF and call-limit in 1.8
Not sure about this since I use the 10/11 branches and not 1.8, but I think you need to use the deprecated call-limit for BLF and the new busylimit for the other features you need. http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B. Christensen Sent: Thursday, December 06, 2012 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] BLF and call-limit in 1.8 Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This change has other implications, however. Call waiting stops working, queues don't offer calls if the user is in a private call etc. We have customers that require both BLF and call waiting at the same time. We are running Asterisk 1.8.11-cert7 I've made the following additions to sip.conf [general]: callcounter=yes counteronpeer=yes (undocumented? Supposed to replace limitonpeers?) (old relevant values, unchanged) allowsubscribe=yes subscribecontext=blf notifyringing=yes notifyhold=yes limitonpeers=yes I also tried may other suggestions I've found like placing the hints in the same context as the extensions and removing subscribecontext. Is there something I'm missing? Is something not working correctly? Thanks in advance, Pan -- _ -- 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] [OT] Polycom IP450 Firmware Issues
What happens if you reinitialize a phone, then do the update? (I keep a bottle of Ibuprophen on hand just for Polycom issues). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Thursday, December 06, 2012 3:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [OT] Polycom IP450 Firmware Issues I have a site with Polycom handsets on all the desks, mostly IP650s, some IP550s, and some IP450s as well. I need to update the firmware on the IP450s. However, the firmware simply won't load. The latest firmware (4.0.3 Rev F) supports all phones at this site, and was downloaded from here: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html The phone pulls the firmware from the PBX via TFTP (as expected), but always results in 'Error: Image is not compatible with the phone'. As a troubleshooting step, *ALL* firmware has been removed from the TFTP root, and *ONLY* the new firmware placed there. So, is the Polycom firmware matrix wrong about this phone/firmware compatibility, or am I missing something? The bootrom has also been upgraded to the latest without any problems. Thoughts? My head is getting sore from banging it on my desk... :/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 -- _ -- 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
