Re: [asterisk-users] Asterisk SMS()
On Tuesday 19 February 2013, Nicholas Johnson wrote: Thanks for the help. Right now I'm running asterisk on a raspberry pi using a phone number from flowroute. Is using a company like flowroute the same as connecting to the PSTN? Also i've tried to install smsq but I couldn't find any good documentation to get it setup properly. So no, I'm not using smsq. The bad news: You need a GSM modem to send SMS messages. The good news: It is not so. You can send SMS messages on POTS or ISDN lines See the voip-wiki about it In the US (and other parts of the world) there are SMS gateways the providers offer to reach their subscribers They are free and since you are using a raspberry which means no direct pstn interface might be a good approach with the help of some AGI/bash scripting -- Stelios S. Koroneos -- _ -- 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 question
2013/2/20 Nguyễn Công nguyencong.1...@gmail.com Hello everyone, I’m new to Asterisk and I have a question. There is a phone call between two users, then they are talking to each other directly or by the server. I mean all packets from the user A to user B will be send directly to each other or will those packets from user A must be send to server and server will send to user B. Thanks. -- Both cases can happens. In a VoIP call we have two connections, one is used for signaling, usually port 5060 for SIP protocol, UDP transport and one is used for media (voice), usually random port. When the call starts the asterisk server sits in the middle of the media path, meaning all voice packets from phone A go to asterisk server and they are rerouted to phone B. After few milliseconds, if configured this way, asterisk server instructs the phone A to send the media directly to phone B to save bandwidth. It is named reinvite Leandro -- _ -- 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] crossed channels
Ist one channel significant louder than the other? Maybe it is some sort of crosstalking. Take a look here: http://es.wikipedia.org/wiki/Diafon%C3%ADa Am 19.02.2013 16:25, schrieb Juan Carlos Agudelo: El 19/02/13 03:59, Thorsten Göllner escribió: What exactly do you mean by crossing channels? Mixed audio? Can callers hear each other? Am 19.02.2013 02:07, schrieb Juan Carlos Agudelo: Hi, I have installed Asterisk 1.6.2.17-rc2 and I have a strange behavior, because sometimes they are crossing channels, thus producing unwanted calls connections...Any suggestions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme and MEETME_EXIT_CONTEXT
Hello,
using Asterisk 1.8.12.2
I am having trouble with exiting the conference room by entering a
single digit.
option X of the Meetme()-application should do this.
I have following in extensions.conf :
/exten = _1000X,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)//
//exten = _1000X,n,MeetMe(${CONFNO},dMX)//
//
//
//[dynamic-nway-invite]//
//exten = 0,1,NoOp(confno = ${CONFNO})//
//exten = 0,n,Read(DEST,dial,,i)//
//exten = 0,n,Set(DYNAMIC_FEATURES=nway-inv#nway-noinv)//
//exten = 0,n,Dial(Local/${DEST}@${LocalContext},,g)//
//exten = 0,n,Set(DYNAMIC_FEATURES=)//
//exten = 0,n,NoOp(tralalala)//
//exten = 0,n,Goto(dynamic-nway1,${CONFNO},1)//
//exten = i,1,Goto(dynamic-nway1,${CONFNO},1)//
/
So by pressing 0 (zero) while in the conference room, I should be able
to exit and continue in the context [dynamic-nway-invite] . Correct ?
But nothing happens when pressing 0 (zero).
What am I missing ??
Kind regards,
Jonas.
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Re: [asterisk-users] Meetme and MEETME_EXIT_CONTEXT
- Original Message - From: Jonas Kellens jonas.kell...@telenet.be But nothing happens when pressing 0 (zero). Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it. You can also push those to the console and watch what happens when you press zero. On the console be sure to turn up verbosity with core set verbose 5 If you can't tell what is happening, post a pastebin link to the log and point out (via timestamp or otherwise) where you would expect to see the DTMF digit. Maybe someone will be able to take a look. I'd also really recommend using ConfBridge which is newer than MeetMe. If you switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ERROR: chan_dahdi.c: PRI Span: 3 PROBLEM: General: Badly Structured Component
Hi all, has anybody ever encountered this ERROR before? It happens frequently on my debian6-based pbx. I'm using Asterisk 1.8.11 with dahdi-linux-2.4.1 and a quadBRI card. ERROR: chan_dahdi.c: PRI Span: 3PROBLEM: General: Badly Structured Component I tried to google but without success. Do you know what it means? Should I worry? Thank You Giorgio -- _ -- 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] Meetme and MEETME_EXIT_CONTEXT
Hello, I don't really see anything when pressing '0' (zero). It's like the '0' (zero) does not reach Asterisk. However the password to enter the conference does reach Asterisk well. Kind regards, Jonas. On 02/20/2013 03:32 PM, Rusty Newton wrote: - Original Message - From: Jonas Kellens jonas.kell...@telenet.be But nothing happens when pressing 0 (zero). Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it. You can also push those to the console and watch what happens when you press zero. On the console be sure to turn up verbosity with core set verbose 5 If you can't tell what is happening, post a pastebin link to the log and point out (via timestamp or otherwise) where you would expect to see the DTMF digit. Maybe someone will be able to take a look. I'd also really recommend using ConfBridge which is newer than MeetMe. If you switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] exten = h,n,AGI(generateCall.php,${NEXT})
not able to run my php from AGIi am using asterisk 1.8.13 (debian)i am able to
make call file using php command line..but when executing php from AGI, it is
not working..kindly see the attachment if bellow text is not
readable...___ File:
/etc/asterisk/extensions.conf[call]exten = call,1,Answerexten =
call,n,Playback(hello-world)exten = call,n,Hangup()
exten = h,1,Set(NEXT=$[${NEXT}+1])exten =
h,n,AGI(generateCall.php,${NEXT})exten =
h,n,Hangup()___File:
/usr/share/asterisk/agi-bin/generateCall.php
#!/usr/bin/php -q?php$fileName = /var/www/consumer.txt;$next = $argv[1];$f =
fopen($fileName,'r');$fileContent =
file_get_contents($fileName);$outdialNumbers = explode(\n,$fileContent);
if($outdialNumbers[$next]) {$callFile =
/var/spool/asterisk/outgoing/.$outdialNumbers[$next]..call;$f
= fopen($callFile,'w');$callFileContent= \nChannel:
dongle/dongle0/.$outdialNumbers[$next].\nContext: call\nExtension:
call\nPriority: 1\nSet: NEXT=.$next.\n;fwrite($f, $callFileContent);
fclose($f);chmod($callFile,
0777);}?mac@almighty
~ $ ls /usr/share/asterisk/agi-bin/ -ltotal 4-rwxrwxrwx 1 root root 1166 Feb
20 15:48
generateCall.phpmac@almighty
~ $ ls /var/spool/asterisk/ -ltotal 28drwxrwxrwx 2 root users 4096 Sep 13
06:59 dictatedrwxrwxrwx 2 root users 4096 Sep 13 06:59 meetmedrwxrwxrwx 2 root
users 4096 Sep 13 06:59 monitordrwxrwxrwx 2 root users 4096 Feb 20 20:39
outgoingdrwxrwxrwx 2 root users 4096 Sep 13 06:59 systemdrwxrwxrwx 2 root users
4096 Sep 13 06:59 tmpdrwxrwxrwx 2 root users 4096 Sep 13 06:59
voicemailmac@almighty ~
$almighty*CLI[Feb
20 20:39:32] WARNING[2007]: pbx_spool.c:278 safe_append: Unable to set utime
on /var/spool/asterisk/outgoing/9033544852.call: Operation not permitted--
Attempting call on dongle/dongle0/9033544852 for call@call:1 (Retry 1)
Channel Dongle/dongle0-01000d was answered.-- Executing [call@call:1]
Answer(Dongle/dongle0-01000d, ) in new stack-- Executing
[call@call:2] Playback(Dongle/dongle0-01000d, silence/1) in new stack
-- Dongle/dongle0-01000d Playing 'silence/1.gsm' (language 'en')--
Executing [call@call:3] Playback(Dongle/dongle0-01000d, hello-world) in
new stack-- Dongle/dongle0-01000d Playing 'hello-world.gsm' (language
'en')-- Executing [call@call:4] SayDigits(Dongle/dongle0-01000d, 0)
in new stack-- Dongle/dongle0-01000d Playing 'digits/0.gsm' (language
'en')-- Executing [call@call:5] Hangup(Dongle/dongle0-01000d, ) in
new stack == Spawn extension (call, call, 5) exited non-zero on
'Dongle/dongle0-01000d'-- Executing [h@call:1]
Set(Dongle/dongle0-01000d, NEXT=1) in new stack-- Executing
[h@call:2] AGI(Dongle/dongle0-01000d, generateCall.php,1) in new stack
-- Launched AGI Script
/usr/share/asterisk/agi-bin/generateCall.phpDongle/dongle0-01000dAGI Tx
agi_request: generateCall.phpDongle/dongle0-01000dAGI Tx
agi_channel: Dongle/dongle0-01000dDongle/dongle0-01000dAGI Tx
agi_language: enDongle/dongle0-01000dAGI Tx agi_type:
DongleDongle/dongle0-01000dAGI Tx agi_uniqueid:
1361372972.13Dongle/dongle0-01000dAGI Tx agi_version:
1.8.13.1~dfsg-1Dongle/dongle0-01000dAGI Tx agi_callerid:
unknownDongle/dongle0-01000dAGI Tx agi_calleridname:
unknownDongle/dongle0-01000dAGI Tx agi_callingpres:
0Dongle/dongle0-01000dAGI Tx agi_callingani2:
0Dongle/dongle0-01000dAGI Tx agi_callington:
0Dongle/dongle0-01000dAGI Tx agi_callingtns:
0Dongle/dongle0-01000dAGI Tx agi_dnid:
unknownDongle/dongle0-01000dAGI Tx agi_rdnis:
unknownDongle/dongle0-01000dAGI Tx agi_context:
callDongle/dongle0-01000dAGI Tx agi_extension:
hDongle/dongle0-01000dAGI Tx agi_priority:
2Dongle/dongle0-01000dAGI Tx agi_enhanced:
0.0Dongle/dongle0-01000dAGI Tx
agi_accountcode:Dongle/dongle0-01000dAGI Tx agi_threadid:
1129301104Dongle/dongle0-01000dAGI Tx agi_arg_1:
1Dongle/dongle0-01000dAGI Tx Dongle/dongle0-01000dAGI Rx Could
not open input file: 1Dongle/dongle0-01000dAGI Tx 510 Invalid or
unknown command-- Dongle/dongle0-01000dAGI Script generateCall.php
completed, returning 0[Feb 20 20:39:44] NOTICE[2672]: pbx_spool.c:366
attempt_thread: Call completed to dongle/dongle0/9033544852almighty*CLI
my detail is bellow.
i am using asterisk 1.8.13
___
File: /etc/asterisk/extensions.conf
[call]
exten = call,1,Answer
Re: [asterisk-users] exten = h,n,AGI(generateCall.php,${NEXT})
On Wed, Feb 20, 2013 at 9:23 AM, Mahendra Dobariya
mahendra_mahen...@hotmail.com wrote:
File: /etc/asterisk/extensions.conf
[call]
exten = call,1,Answer
exten = call,n,Playback(hello-world)
exten = call,n,Hangup()
exten = h,1,Set(NEXT=$[${NEXT}+1])
exten = h,n,AGI(generateCall.php,${NEXT})
Try
exten =
h,n,AGI(/usr/bin/php,/usr/share/asterisk/agi-bin/generateCall.php,${NEXT})
exten = h,n,Hangup()
Dongle/dongle0-01000dAGI Rx Could not open input file: 1
This is indicating that, for whatever reason, php is seeing 1 as argv[1],
not the name of your script file. I reproduced this by making a php shebang
that looks like
#!/usr/bin/php 1
Not sure why, though. The above should be a workaround for now.
--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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Re: [asterisk-users] exten = h,n,AGI(generateCall.php,${NEXT})
On Wed, 20 Feb 2013, Mahendra Dobariya wrote: not able to run my php from AGI Your PHP script is not an AGI. It does not conform to the Asterisk Gateway Interface protocol. Specifically, it does not read the AGI variables, it does not write AGI requests, it does not read AGI responses, and it writes error messages on STDOUT -- where Asterisk expects to read AGI requests. Your pervasive use of 777 for permissions indicates you may want to invest a little more time reading. For example, suppose I can gain local shell access to your host or trick some service into executing: echo 'rm -f -r /*' /usr/share/asterisk/agi-bin/generateCall.php Unless you can restore the ownership and permissions of your filesystem to their original values, I'd suggest un-installing Asterisk, deleting any remaining files and directories and then installing from scratch. Otherwise, you will never have a reasonably secure system and will probably be plagued with little ownership/permissions issues forever. Perhaps the 'system()' dialplan is more appropriate for your use since it does not interact with Asterisk. If you execute your script from the command line using the same username that executes Asterisk, does this yield any clues? Where does the error message 'Could not open input file: 1' come from? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ERROR: chan_dahdi.c: PRI Span: 3 PROBLEM: General: Badly Structured Component
has anybody ever encountered this ERROR before? It happens frequently on my debian6-based pbx. I'm using Asterisk 1.8.11 with dahdi-linux-2.4.1 and a quadBRI card. ERROR: chan_dahdi.c: PRI Span: 3PROBLEM: General: Badly Structured Component I tried to google but without success. Do you know what it means? Should I worry? It means that the peer has rejected a facility message sent by Asterisk. Facility messages are mainly used to implement supplementary services. Supplementary services are things like call-completion, explicit-call-transfer, call-diversion/redirection, and advice-of-charge. The supplementary service that Asterisk was attempting to invoke was rejected and thus failed. It could be that the peer does not support the service, does not recognize the format used, or does not handle the message correctly. A pri set debug on span x trace is needed to give any more information. Richard -- _ -- 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] Meetme and MEETME_EXIT_CONTEXT
- Original Message - From: Jonas Kellens jonas.kell...@telenet.be Hello, I don't really see anything when pressing '0' (zero). It's like the '0' (zero) does not reach Asterisk. However the password to enter the conference does reach Asterisk well. Please don't top post (https://www.asterisk.org/community/discuss). Also, you didn't pastebin any debug, so I can't confirm that there is not some other issue upon a possible DTMF reception. If it is the case that Asterisk doesn't detect a DTMF 0 when you send it from the endpoint, then you probably want to look at a SIP packet capture to verify the endpoint is actually sending the DTMF to Asterisk. What you look for in the capture or audio will depend on what kind of DTMF you are sending with the endpoint. Does Asterisk detect the digit 0 at any other time outside of MeetMe? Can you setup an extension matching for 1234567890 and dial that? Do you see DTMF debug for all those digits? If you do end up trying ConfBridge - I've never used it in 1.8. Others have made me aware that ConfBridge wasn't the best in 1.8, and that it's much better in 10 or preferably 11. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Blips at end of Record() - 1.8.18
Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -- _ -- 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 with inbound calls
When you add a card, it adds channels, so what used to be dahdi channel 1 is now probably channel 49 or 97. Look at /etc/dahdi/system.conf and /etc/asterisk/dahdi-channels.conf to see how you have it configured. I'm not sure what the zaptel equivalents are - my guess would be /etc/zaptel/system.conf and /etc/asterisk/zaptel-channels.conf -Justin Killen From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Wednesday, February 20, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] issue with inbound calls hello list, i add a new diguim card in my server i use asterisk 1.4 with zaptel .conf after that i can't receive the calls in my server with outbound calls there is no problem i have all time this error msg [Feb 20 18:15:48] WARNING[28582]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! [Feb 20 18:15:52] WARNING[28582]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! [Feb 20 18:15:56] WARNING[28582]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! any help please thank you [cid:image002.gif@01CE0F5D.5B69AB30] inline: image002.gif-- _ -- 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] Meetme and MEETME_EXIT_CONTEXT
Hello, what is the equivalent parameter of X in the ConfBridge()-command ? How can you exit ConfBridge by pressing a digit ? Concerning MeetMe() : Verbosity is 25 and I still don't see anything on the console or in the logs when pressing '0' (zero). Kind regards, Jonas. On 02/20/2013 03:32 PM, Rusty Newton wrote: - Original Message - From: Jonas Kellens jonas.kell...@telenet.be But nothing happens when pressing 0 (zero). Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it. You can also push those to the console and watch what happens when you press zero. On the console be sure to turn up verbosity with core set verbose 5 If you can't tell what is happening, post a pastebin link to the log and point out (via timestamp or otherwise) where you would expect to see the DTMF digit. Maybe someone will be able to take a look. I'd also really recommend using ConfBridge which is newer than MeetMe. If you switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remove Abandoned call
hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same time from two different server_A and server_B and hit into server_X and one call is abandoned and another one have taken by the agent But i don't want to abandoned the call, I want to set the priority, supposed to server_A and server_B call originate same time server_X take the call from server_A first and then take the call server_B after 1 sec please guide me Regards Akhilesh -- _ -- 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
