Re: [asterisk-users] touch command not behaving for future calls in asterisk 1.4.41
Yeah, that's what I was saying J good it fixed it. BR Gohar From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Friday, July 06, 2012 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] touch command not behaving for future calls in asterisk 1.4.41 Thanks Gohar, I found the issue was copy file to outbound folder not moving. that's why after making future time asterisk start reading file. On Fri, Jul 6, 2012 at 11:16 AM, SamyGo govoi...@gmail.com wrote: Hi, Did you get anything working on it !! See the permission for the user running asterisk process and see if that user can touch files like that. Regards, Sammy On Thu, Jul 5, 2012 at 10:47 PM, virendra bhati virbh...@gmail.com wrote: Hi All, It's small issue but making a big problem for my application. I have CentOS release 5.8 (Final) with asterisk 1.4.41 installed. I am using 1.4.41 because Flite work in this version. problem is that when I make changes on .call file to make it future call file with touch command then it not changed. [root@server tmp]# touch -t 201207052137 1341509545.39.call [root@server tmp]# ll -rw-r--r-- 1 root root 52 Jul 5 2012 1341509545.39.call .call file's time is missed with year only that's asterisk make call after move to outgoing folder. please give your suggestion. If I am wrong then correct me ... -- Thanks and regards Virendra Bhati +91-9718300881 E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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 - Integration with building intercom systems
2012/7/5, giovanni.v i...@keybits.org: The matter becomes more difficult approaching a building install as there are no devices to handle properly that. I think the snom PA-1 may be a good candidate to play with because of a versatile I/O that could be interfaced to a custom door-phone bridge to IP. Handling both audio and video seems difficult. -- _ -- 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 - Multi Function Printer with one touch scanning/emailing
2012/7/5, C F shma...@gmail.com: snip no sure if you can have it function such that any number entered will actually be send to a gateway. To me, that is the key selling point : people are used to just dial a number and then press a Fax button that I can't succeed in anything more than teaching them to press a Scan button instead. -- _ -- 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 - Multi Function Printer with one touch scanning/emailing
2012/7/5, C F shma...@gmail.com: I searched a bit more, http://www.muratec.com/catalog/F320_config.html#email The above model supports t.37 That's very interesting to know. I quickly googled for t.37 and found several other vendors mentioning this (some from rather old documents). The strange thing is some vendors seem to completely ignore t.37 and t.38 (google for hp t.37 or hp t.38 ). -- _ -- 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 - Integration with building intercom systems
I think you are supposed to have a IP based terminal in order to carry both video and audio. If budget is accept, it is possible to setup several iOS devices as dedicated SIP terminals On Jul 6, 2012, at 8:49 AM, Olivier wrote: 2012/7/5, giovanni.v i...@keybits.org: The matter becomes more difficult approaching a building install as there are no devices to handle properly that. I think the snom PA-1 may be a good candidate to play with because of a versatile I/O that could be interfaced to a custom door-phone bridge to IP. Handling both audio and video seems difficult. -- _ -- 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] Timer1 RFC and SIP.CONF
Hello, Thank you for the clarification. Just a few questions: 1. What is the Timer1 used for? 2. Since timerb is for all responses, final and provisional, the comment in sip.conf perhaps should point that out instead of implying only for provisional responses: If a provisional response is not received in this amount of time, the call will autocongest Thanks, Elliot On Thu, Jul 5, 2012 at 12:31 PM, Olle E. Johansson o...@edvina.net wrote: 4 jul 2012 kl. 13:32 skrev Elliot Murdock: Hello, I am trying to get clarity with the sip.conf timer configuration. The current configuration states: ;--- SIP timers ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 However, according to RFC 3261: (EXCERPT 17.1.1.1) T1 is an estimate of the round-trip time (RTT), and it defaults to 500 ms. Nearly all of the transaction timers described here scale with T1, and changing T1 adjusts their values. The request is not retransmitted over reliable transports. After receiving a 1xx response, any retransmissions cease altogether, and the client waits for further responses. The server transaction can send additional 1xx responses, which are not transmitted reliably by the server transaction. Eventually, the server transaction decides to send a final response. (EXCERPT 13.2.2.4 2xx Responses) The UAC core considers the INVITE transaction completed 64*T1 seconds after the reception of the first 2xx response. According to the RFC, the 64*t1 timeout is not for provisional responses, but for final responses. This seems to be in contradiction to what is stated in the sip.conf file. Unless you have PRACK support, which Asterisk not yet has, there's no timeout in the SIP protocol for provisional responses more than the need to update your provisional response at least every minute not to hit a 3 minute timeout in the SIP transaction state in a proxy. Now, the timerb is used to get ANY response from a server out there, including provisional responses. If we don't get ANY response within TIMERB, the SIP transaction dies and in a UA with a transaction layer we generate a local 408 timeout. In Asterisk, we congest the call. So the 64*T1 is for any response, including final response. It's there to decide whether or not you have intelligent SIP life forms handling your SIP request in the network universe. I hope this clears up your confusion. Regards, /Olle -- _ -- 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] OT - Integration with building intercom systems
Il 06/07/2012 8.49, Olivier ha scritto: Handling both audio and video seems difficult. Sure, I don't know a device being able to handle all problems involved in that project. Another brick near the wall, but no versatile I/O: - http://www.grandstream.com/products/ip-video-surveillance/gxv3500 -- _ -- 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] sip and extensions
Hello, If you would like to make out bound call (from Asterisk to SIP provider), it is fine. But if you want have inbound call (from SIP provider to Asterisk). I think you are supposed to have something like this sip.conf register = 5552530146:your_password@sip3.voipvoip.com/5552530146 [5552530146] ... context=incoming extensions.conf [incoming] ;first creating extensions for your local users exten = 5552530146,1,Goto(5552530146_incomming,s,1) [5552530146_incomming] ;more logic wish it would help. On Jul 5, 2012, at 11:44 PM, Thomas Perron wrote: I am new. Here is the code that I am playing with on CentOS 6.x When I dial the number that corresponds w/ my SIP account I get a recording: reached a non-working number I built Asterisk a few times last year and am now back working on a similar project. In my view, there is something wrong in sip.conf I don't remember using a file that long to get a basic call set up. The format was provided to me by voipvoip.com (the SIP provider). Does anyone have any comments please? I just want a very simple config to get my machine to recognize a call to the SIP provider. Here is results of sip show registry: Hostdnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060 N 5552530146 285 Registered Thu, 05 Jul 2012 21:39:56 1 SIP registrations. Here is sip and extensions.conf sip.conf [general] register = 5552530146:funnytiger...@sip3.voipvoip.com ; [sip3.voipvoip.com] [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no extensions.conf [general] ; ; [incoming] ;first creating extensions for your local users exten= s,1,Dial(SIP/1703717) exten= s,2,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 -- _ -- 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] call file and NFS server
Hello,
I have 3 server, 2 running with asterisk and another one generate call
files say some directory callfile/serverA and callfile/serverB (NFS
Sharing) and mounted this directory to respectively on Server A (Asterisk)
and Server B(Asterisk) on /var/spool/asterisk/outgoing.
Server A has Asterisk 1.8.0-rc2 and Server B has asterisk version 1.8.9.0,
and both asterisk compile ./configure --without-inotify
Callfile will execute call successfully on both machine, but got the
following problem
*[Jul 6 16:15:04] WARNING[26921]: pbx_spool.c:278 safe_append: Unable to
set utime on /var/spool/asterisk/outgoing/15.call: Operation not
permitted
*
I have set the folder (callfile/Server{A/B}) permission to 777 as well as
call file permission to 777.
--
Regards,
Chandrakant Solanki
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Re: [asterisk-users] call file and NFS server
Why don't you just generate call files for each of the servers on the same
server? Anyhow you are not sharing one single pool of call files among servers,
I suspect that's where network drive would come in handy.
Sent from my iPhone
On Jul 6, 2012, at 6:56 PM, Chandrakant Solanki solanki.chandrak...@gmail.com
wrote:
Hello,
I have 3 server, 2 running with asterisk and another one generate call files
say some directory callfile/serverA and callfile/serverB (NFS Sharing) and
mounted this directory to respectively on Server A (Asterisk) and Server
B(Asterisk) on /var/spool/asterisk/outgoing.
Server A has Asterisk 1.8.0-rc2 and Server B has asterisk version 1.8.9.0,
and both asterisk compile ./configure --without-inotify
Callfile will execute call successfully on both machine, but got the
following problem
[Jul 6 16:15:04] WARNING[26921]: pbx_spool.c:278 safe_append: Unable to set
utime on /var/spool/asterisk/outgoing/15.call: Operation not permitted
I have set the folder (callfile/Server{A/B}) permission to 777 as well as
call file permission to 777.
--
Regards,
Chandrakant Solanki
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Re: [asterisk-users] call file and NFS server
On Friday 06 July 2012, Chandrakant Solanki wrote:
I have 3 server, 2 running with asterisk and another one generate call
files say some directory callfile/serverA and callfile/serverB (NFS
Sharing) and mounted this directory to respectively on Server A (Asterisk)
and Server B(Asterisk) on /var/spool/asterisk/outgoing.
Server A has Asterisk 1.8.0-rc2 and Server B has asterisk version 1.8.9.0,
and both asterisk compile ./configure --without-inotify
Callfile will execute call successfully on both machine, but got the
following problem
*[Jul 6 16:15:04] WARNING[26921]: pbx_spool.c:278 safe_append: Unable to
set utime on /var/spool/asterisk/outgoing/15.call: Operation not
permitted
*
I have set the folder (callfile/Server{A/B}) permission to 777 as well as
call file permission to 777.
The problem is that root on one machine doesn't have full root access to other
users' files on NFS shares. A user logged in as root on a local machine and
accessing an NFS share on a remote machine ordinarily has *fewer* privileges,
and even world write doesn't allow remote root write. This is by design; as
otherwise, a local privilege escalation on one machine can lead to a whole-
network privilege escalation.
(By the way, you should have permissions 666 for a callfile, not 777.
Callfiles
should not be executable.)
You could either recompile all the NFS stuff (not really recommended); or
have the callfile generated and re-timed by a CGI script on the remote machine
(where /var/spool/asterisk/outgoing actually is), fired off by `wget` on the
local machine.
--
AJS
Answers come *after* questions.
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[asterisk-users] Can I install Asterisk normally and then installing the GUI asterisk now
Hello; Is it possible if I have already asterisk installed on Fedora machine to install the GUI asterisk now without doing a fresh installation using the Asterisk Now CD? Which version of the GUI that should be selected to work with the asterisk version? For example, if I have asterisk 1.8 then which GUI version to select? I am talking about compatibility. Can I say that Freepbx is Asterisk + Asterisk Now? 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
Re: [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing
There are some appliances that support it is well. But those don't have a scanner, just thru the computer. MultiTech FaxFinder comes to mind, for the price they are excellent. On Fri, Jul 6, 2012 at 3:13 AM, Olivier oza_4...@yahoo.fr wrote: 2012/7/5, C F shma...@gmail.com: I searched a bit more, http://www.muratec.com/catalog/F320_config.html#email The above model supports t.37 That's very interesting to know. I quickly googled for t.37 and found several other vendors mentioning this (some from rather old documents). The strange thing is some vendors seem to completely ignore t.37 and t.38 (google for hp t.37 or hp t.38 ). -- _ -- 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] Can I install Asterisk normally and then installing the GUI asterisk now
- Original Message - Hello; Is it possible if I have already asterisk installed on Fedora machine to install the GUI asterisk now without doing a fresh installation using the Asterisk Now CD? Which version of the GUI that should be selected to work with the asterisk version? For example, if I have asterisk 1.8 then which GUI version to select? I am talking about compatibility. Can I say that Freepbx is Asterisk + Asterisk Now? You could, but it would be wrong. :) AsteriskNow is Asterisk+FreePBX in a nutshell. Of course there is better package management (RPM repos from Digium vs source installs or building your own packages), etc. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk trying to call a queue with no members
Hi, I am trying to configure some static queues in asterisk, it's almost working, the problem is that asterisk is not verifying if the queue has logged members. For example, if I create queue called test, which has no members logged in, and try to place a call using Queue(test) I get into the queue, even if all phones are turned off, I tried to verify it with the QUEUE_MEMBER function, using the ready parameter, it shows me that all members are logged in. Here is my queues.conf: [general] persistentmembers = yes monitor-type = MixMonitor [default-queue](!) musicclass=default joinempty=yes leavewhenempty=no strategy=random timeout=10 retry=3 timeoutpriority=conf autofill=yes announce-position = yes ringinuse=no [recepcao](default-queue) member = SIP/100 member = SIP/101 member = SIP/102 I haven't showed the other queues because there is no need to do that. I've enabled sip qualifying in sip.conf, Must I do anything else to get it working properly? Thanks in advance. 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] call file and NFS server
On Friday 06 July 2012, Chandrakant Solanki wrote:
I have set the folder (callfile/Server{A/B}) permission to 777 as well
as call file permission to 777.
On Fri, 6 Jul 2012, A J Stiles wrote:
(By the way, you should have permissions 666 for a callfile, not 777.
Callfiles should not be executable.)
Whenever I see 777 (or it's Satanic cousin, 666) I see 'I don't really
understand ownership and permissions so let's just allow everything and
hope for the best.'
Do you really intend to allow every user and exploited program to be able
to create call files? (And if you've done this, you've probably created
other holes in your system's security.)
While 'opening the flood gates' is (IMO) a valid temporary debugging
technique to identify the source of the problem, the directories and files
should be owned by the user executing Asterisk and permissions should
limit reading to only users and groups that need reading and limit writing
to only users and groups that need writing.
I don't have any need or experience with call files on my production
boxes, but I suspect a successful implementation would include NTP and
creating the call file in another directory on the shared device and then
moving the call file to the outgoing spool directory.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Asterisk trying to call a queue with no members
On 07/06/2012 10:15 AM, Antonio Modesto wrote: Hi, I am trying to configure some static queues in asterisk, it's almost working, the problem is that asterisk is not verifying if the queue has logged members. For example, if I create queue called test, which has no members logged in, and try to place a call using Queue(test) I get into the queue, even if all phones are turned off, I tried to verify it with the QUEUE_MEMBER function, using the ready parameter, it shows me that all members are logged in. Here is my queues.conf: [general] persistentmembers = yes monitor-type = MixMonitor [default-queue](!) musicclass=default joinempty=yes This setting will cause Queue() to allow callers to join the queue even if the queue has no members. Since you are saying you don't want that, you should turn it off. leavewhenempty=no strategy=random timeout=10 retry=3 timeoutpriority=conf autofill=yes announce-position = yes ringinuse=no [recepcao](default-queue) member = SIP/100 member = SIP/101 member = SIP/102 You have statically defined the members of this queue, there is no 'logging in' required. The Queue() app does not try to determine whether the devices you list as members are currently connected to Asterisk or not. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI DTMF problem?
- Original Message - It has a Digium Wildcard TE122 If it has an onboard echo canceler, try disabling it and retrying. Just a shot in the dark, going from my experience with other cards and same symptoms. If the card is new(ish) I would think Digium could provide support to you for determining the DTMF problems. --Tim -- _ -- 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 DTMF problem?
On Fri, Jul 06, 2012 at 11:10:43AM -0500, Tim Nelson wrote: - Original Message - It has a Digium Wildcard TE122 If it has an onboard echo canceler, try disabling it and retrying. Just a shot in the dark, going from my experience with other cards and same symptoms. If the card is new(ish) I would think Digium could provide support to you for determining the DTMF problems. Bill, To repeat what Tim said, if you're eligible I would recommend contacting Digium's support department. There are many variables with a new install and Digium support would be happy to help you troubleshoot. Some other things to try in order to isolate the drivers / hardware change from the Asterisk change. You could: a) Install the exact same versions of Asterisk / DAHDI that you used previously on the new server with Centos 5.8. b) Run a local pattern test to verify the host - card communication is valid. If you have problems here there may be a framebuffer configured or a disk controller running in combined mode preventing the cards interrupt handler from running in a timely fashion. c) Use dahdi_monitor to record the audio on the channel you're testing with and open it up with audacity and verify that the DTMF looks correct. If it does, then most likely there is a problem with chan_dahdi reading the audio from the drivers quickly enough. d) Ensure that you are only loading the Asterisk modules you need in case you're running on a system with limited memory and the asterisk process is dropping audio while paging in code. (See DAHLIN-241 [1]) [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI DTMF problem?
On Fri, Jul 06, 2012 at 11:07:26AM -0500, Bill Dunn - VCI Internet Services wrote: It has a Digium Wildcard TE122 I've asked Digium about the card below. They say the -1 in the bipolar and CRC errors is ok. They don't change. Description AlarmsIRQbpviol CRC Fra Codi Options LBO Wildcard TE122 Card 0 OK1 -1 -1 ESF B8ZS0 db (CSU)/0-133 feet (DSX-1) Some background on the -1s for the bipolar and CRC errors: The driver for the TE122 returns -1's now for the error counters to flag that the driver does not actually collect them versus returning 0 which was leading users to believe no errors existed. This change was made in r10212 wcte12xp: Set uncollected performance counters to -1 [1]. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10212 -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI DTMF problem?
Bill Dunn - VCI Internet Services wrote: I have an Asterisk server configured to run as voicemail with a T1 and SMDI. It has 1.6.1.6 (dahdi 2.1.0.4) and Centos 5.6 and has worked great for a few years. I am configuring a new server with Asterisk 1.8.13 (dahdi 2.6.1) on Centos 5.8 The problem I am having appears to be related to DTMF detection. When the test phone number is called (2704083000) Asterisk only receives a portion of the dialed number. It varies as to what numbers are detected. Sometimes it sees a single digit, sometimes 3 or 4 of the digits of the dialed number. When I compare this to the old server the debug below is similar but there isn't any mention of the sig_analog.c lines shown below. I am told the T1's on the old server and the new server are configured the same. I can make outgoing calls on the T1 from Asterisk. Can someone give me a clue as to what could be causing this? Bill Dunn Try setting: relaxdtmf=yes We used to have that same problem on most of our servers. Setting relaxdtmf to yes solved the problem for us. -- Ron Bergin Network Operations Administrator Fry's Electronics, Inc. -- _ -- 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 trying to call a queue with no members
On Fri, 2012-07-06 at 11:09 -0500, Kevin P. Fleming wrote: On 07/06/2012 10:15 AM, Antonio Modesto wrote: Hi, I am trying to configure some static queues in asterisk, it's almost working, the problem is that asterisk is not verifying if the queue has logged members. For example, if I create queue called test, which has no members logged in, and try to place a call using Queue(test) I get into the queue, even if all phones are turned off, I tried to verify it with the QUEUE_MEMBER function, using the ready parameter, it shows me that all members are logged in. Here is my queues.conf: [general] persistentmembers = yes monitor-type = MixMonitor [default-queue](!) musicclass=default joinempty=yes This setting will cause Queue() to allow callers to join the queue even if the queue has no members. Since you are saying you don't want that, you should turn it off. I've set the two options to 'strict', it is working now. One more question, I selected the random strategy, but sometimes asterisk randomly selects a member that is on a call, then it starts to announce an incoming call to this member, and there are other members available. leavewhenempty=no strategy=random timeout=10 retry=3 timeoutpriority=conf autofill=yes announce-position = yes ringinuse=no [recepcao](default-queue) member = SIP/100 member = SIP/101 member = SIP/102 You have statically defined the members of this queue, there is no 'logging in' required. The Queue() app does not try to determine whether the devices you list as members are currently connected to Asterisk or not. I don't want the users to manually login in the queue, I want they join the queue when they turn on their phone. I thought that this was the right way of doing it, how can I do it? -- _ -- 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 DTMF problem?
Thanks Ron. I have had my chan_dahdi.conf file set as follows with the same result. [trunkgroups] [channels] switchtype=national usecallerid=yes callerid=asreceived cidsignalling=smdi echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 usesmdi=yes smdiport=/dev/ttyS0 signalling = em_w immediate = no group = 1 channel = 1-3 Bill Dunn - Original Message - From: Ron Bergin To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, July 06, 2012 12:34 PM Subject: Re: [asterisk-users] DAHDI DTMF problem? Bill Dunn - VCI Internet Services wrote: I have an Asterisk server configured to run as voicemail with a T1 and SMDI. It has 1.6.1.6 (dahdi 2.1.0.4) and Centos 5.6 and has worked great for a few years. I am configuring a new server with Asterisk 1.8.13 (dahdi 2.6.1) on Centos 5.8 The problem I am having appears to be related to DTMF detection. When the test phone number is called (2704083000) Asterisk only receives a portion of the dialed number. It varies as to what numbers are detected. Sometimes it sees a single digit, sometimes 3 or 4 of the digits of the dialed number. When I compare this to the old server the debug below is similar but there isn't any mention of the sig_analog.c lines shown below. I am told the T1's on the old server and the new server are configured the same. I can make outgoing calls on the T1 from Asterisk. Can someone give me a clue as to what could be causing this? Bill Dunn Try setting: relaxdtmf=yes We used to have that same problem on most of our servers. Setting relaxdtmf to yes solved the problem for us. -- Ron Bergin Network Operations Administrator Fry's Electronics, Inc. -- _ -- 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] DAHDI DTMF problem?
I used the dahdi_monitor to record the audio on the T1 channel of the working server and the new server. The audio stream of the working server allowed me to hear the audio I heard over the phone call plus the DTMF at the very beginning. The audio of the new server was completely messed up. I could barely make out the where the DTMF and the audio were in the file. If I didn't have a working sample I wouldn't know what it was. And, the beginning of the new server sample always contains a hum or tone in it whereas the working server does not have that. What does this tell me? Bill Dunn - Original Message - From: Shaun Ruffell To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, July 06, 2012 12:16 PM Subject: Re: [asterisk-users] DAHDI DTMF problem? On Fri, Jul 06, 2012 at 11:10:43AM -0500, Tim Nelson wrote: - Original Message - It has a Digium Wildcard TE122 If it has an onboard echo canceler, try disabling it and retrying. Just a shot in the dark, going from my experience with other cards and same symptoms. If the card is new(ish) I would think Digium could provide support to you for determining the DTMF problems. Bill, To repeat what Tim said, if you're eligible I would recommend contacting Digium's support department. There are many variables with a new install and Digium support would be happy to help you troubleshoot. Some other things to try in order to isolate the drivers / hardware change from the Asterisk change. You could: a) Install the exact same versions of Asterisk / DAHDI that you used previously on the new server with Centos 5.8. b) Run a local pattern test to verify the host - card communication is valid. If you have problems here there may be a framebuffer configured or a disk controller running in combined mode preventing the cards interrupt handler from running in a timely fashion. c) Use dahdi_monitor to record the audio on the channel you're testing with and open it up with audacity and verify that the DTMF looks correct. If it does, then most likely there is a problem with chan_dahdi reading the audio from the drivers quickly enough. d) Ensure that you are only loading the Asterisk modules you need in case you're running on a system with limited memory and the asterisk process is dropping audio while paging in code. (See DAHLIN-241 [1]) [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 trying to call a queue with no members
On 07/06/2012 12:36 PM, Antonio Modesto wrote: I don't want the users to manually login in the queue, I want they join the queue when they turn on their phone. I thought that this was the right way of doing it, how can I do it? That's a reasonable way to do it if you like, although it's pretty uncommon for users to have 'turn on' their phones. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk trying to call a queue with no members
He's probably using softphones -Original Message- From: Kevin P. Fleming kpflem...@digium.com Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 06 Jul 2012 13:32:20 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk trying to call a queue with no members On 07/06/2012 12:36 PM, Antonio Modesto wrote: I don't want the users to manually login in the queue, I want they join the queue when they turn on their phone. I thought that this was the right way of doing it, how can I do it? That's a reasonable way to do it if you like, although it's pretty uncommon for users to have 'turn on' their phones. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 DTMF problem?
Bill Dunn - VCI Internet Services wrote: Thanks Ron. I have had my chan_dahdi.conf file set as follows with the same result. [trunkgroups] [channels] switchtype=national usecallerid=yes callerid=asreceived cidsignalling=smdi echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 usesmdi=yes smdiport=/dev/ttyS0 signalling = em_w immediate = no group = 1 channel = 1-3 Bill Dunn - Original Message - From: Ron Bergin To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, July 06, 2012 12:34 PM Subject: Re: [asterisk-users] DAHDI DTMF problem? Try setting: relaxdtmf=yes We used to have that same problem on most of our servers. Setting relaxdtmf to yes solved the problem for us. Are you using SIP? If so, relaxdtmf can also be set in sip.conf as well as a dtmfmode setting that you can adjust. What type of phones are you using? In our case, part of this problem was due our low end 2.4ghz cordless phones. -- Ron Bergin Network Operations Administrator Fry's Electronics (408) 487-4600 -- _ -- 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 trying to call a queue with no members
On Fri, 2012-07-06 at 13:32 -0500, Kevin P. Fleming wrote: On 07/06/2012 12:36 PM, Antonio Modesto wrote: I don't want the users to manually login in the queue, I want they join the queue when they turn on their phone. I thought that this was the right way of doing it, how can I do it? That's a reasonable way to do it if you like, although it's pretty uncommon for users to have 'turn on' their phones. Yes, it's not very common, but is interesting when the phone is with a problem that prevents it from registering in the pbx. (What already happened with me.) -- _ -- 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] Maximum concurrent calls using call files
I am planning on building a testing module which would spawn about 500 calls in order to test the performance of the network by transferring audio/speech files to end points at that juncture.Is it possible to spawn as many concurrent calls (or nearly concurrent calls) using just call files.Is there a limit as to the maximum number that could be spawned.? I tried doing this for about 20 calls and found that there is autofallthrough after a point of time.Is this a problem with my dialplan or is it because of the call files (i also get a warning which states that the ast_queue_frame:Exceptionally long queue length) Thanks, Sathiish -- _ -- 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] Can I install Asterisk normally and then installing the GUI
OK, is there asterisk-gui that differs that freepbx? Or Freepbx is a GUI for asterisk? In other words, if I have asterisk and I need to add for it a GUI, is there asterisk-gui which is differs than freepbx or it is the same? Regards Bilal - Hello; Is it possible if I have already asterisk installed on Fedora machine to install the GUI asterisk now without doing a fresh installation using the Asterisk Now CD? Which version of the GUI that should be selected to work with the asterisk version? For example, if I have asterisk 1.8 then which GUI version to select? I am talking about compatibility. Can I say that Freepbx is Asterisk + Asterisk Now? You could, but it would be wrong. :) AsteriskNow is Asterisk+FreePBX in a nutshell. Of course there is better package management (RPM repos from Digium vs source installs or building your own packages), etc. --Tim -- _ -- 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] Trixbox or FreePBX?
Hi All; Based on what I have to use Trixbox or FreePBX? Can someone advise? 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
[asterisk-users] sip.conf and binaddr issue
Hi there. i am seriously stuck with an asterisk and sip problem. the following sip.conf works with respect to some_peer: [general] bindaddr = x.y.z.w nat = no [some_peer] type=peer host=somehost secret=somesecret some other unrelated options here x.y.z.w is the ip address of the interface pointing to the network containing somehost. more precisely its the address of tun0 and route -n prints Destination Gateway Genmask Flags Metric RefUse Iface [..] x.y.z.0 0.0.0.0 255.255.255.0 U 0 0 0 tun0 [..] here 'it works' implies that i have to change and reload sip.conf after ifup tun0, or anything that forces tun0 to go down, like my dsl provider. also, the bindaddr line is suboptimal for the other peers... the same thing -- without the bindaddr part -- doesnt work. more precisely it almost works. its just incoming sound that doesnt. this must have something to do with how asterisk picks up interface addresses and communicates them to the peer in question. inspecting the packages sent to somehost, gave me the impression that asterisk uses the ip adress of ppp0 (a dsl modem) instead. how am i supposed to tell asterisk to use tun0 as the interface for [some_peer] so i can remove the bindaddr line? i've found many nat-related options in the manual, but there is no nat involved here. also, i couldnt find anything similar to iface=tun0, although the sip dialogue apparently relies on ip adresses and routing. this is about asterisk on sid, version 1:1.8.13.0~dfsg-1, but of course i'm going to switch to whatever you might suggest. regards and thanks felix -- _ -- 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 DTMF problem?
On Fri, Jul 06, 2012 at 01:28:01PM -0500, Bill Dunn - VCI Internet Services wrote: I used the dahdi_monitor to record the audio on the T1 channel of the working server and the new server. The audio stream of the working server allowed me to hear the audio I heard over the phone call plus the DTMF at the very beginning. The audio of the new server was completely messed up. I could barely make out the where the DTMF and the audio were in the file. If I didn't have a working sample I wouldn't know what it was. And, the beginning of the new server sample always contains a hum or tone in it whereas the working server does not have that. What does this tell me? Bill, to close out this thread: After logging into your system, I'm nearly certain you'll need to contact Digium technical support for assistance if moving the card into a different system doesn't change the results. I ran the same commands I ran on your server on a test server of mine and this is what I would have expected on your server: $ cat system.em.conf span=1,0,0,esf,b8zs em=1-24 loadzone= us defaultzone = us $ modprobe --first-time wcte12xp vpmsupport=0 latency=6 debug=1 $ dahdi_cfg -c system.em.conf $ dahdi_maint -s 1 --loopback localhost Span 1: local host loopback ON $ patlooptest 1 -t 10 patlooptest 1 -t 10 Using Timeout of 10 Seconds Going for it... Timeout achieved Ending Program Test ran 33 loops of 2039 bytes/loop with 0 errors Using Timeout of 10 Seconds Going for it... Timeout achieved Ending Program Test ran 34 loops of 2039 bytes/loop with 0 errors On your server this same sequence would produce repeated patloop errors even though there were not any indications in dmesg of latency bumps, which would be case if something else was preventing the wcte12xp interrupt service routine from running. The patlooptest errors would explain the bad audio you recorded with dahdi_monitor and the DTMF detection problems. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I install Asterisk normally and then installing the GUI
- Original Message - OK, is there asterisk-gui that differs that freepbx? Or Freepbx is a GUI for asterisk? In other words, if I have asterisk and I need to add for it a GUI, is there asterisk-gui which is differs than freepbx or it is the same? There have been a handful of other GUIs around, but none have the market share, support, or feature-set of FreePBX. It is pretty much the defacto standard. If you're feeling adventurous, give asterisk-gui a try. Last I checked, the latest code was available from Digium SVN and 'worked better' than from the tar.gz on downloads.digium.com. But again, that was some time ago, YMMV... --Tim -- _ -- 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] Trixbox or FreePBX?
- Original Message - Hi All; Based on what I have to use Trixbox or FreePBX? Can someone advise? Trixbox includes FreePBX as it's GUI. However, keep in mind it is a bastardized, forked version of FreePBX that has seen nary any new development or innovation in some time. At this point, for a standard PBX installation, my recommendations would be (in this order): 1. Elastix 3. AsteriskNOW 2. PBX In a Flash --Tim -- _ -- 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] FreePBX: How to hangup if the caller did not press # after the voicemail message
I did one try and did not hangup: I called the extension and when the voicemail answered, I did not leave any message and stayed waiting .. waiting .. waiting .. It did not hangup from it self ! How much it stay waiting me to leave a message? Why I am trying this? OK, the answer is: Because I am trying to see if the voicemail hangup even if it did not get from the caller the hangup, because this is happening when the caller is calling from outside to the dahdi (analoge) and when the caller hangup his mobile, the hangup does not reach (it is PSTN issue), so I beleive the voicemail should hangup from it self after timeout .. but I do not see this. Anyadvise? Regards Bilal --- Asterisk, and by extension FreePBX, automatically end the voicemail recording when the caller hangs up. You have some OTHER issue. Perhaps Asterisk is not detecting the hangup? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, July 05, 2012 6:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FreePBX: How to hangup if the caller did not press # after the voicemail message Dears; In FreePBX, when I select voicemail for the extension, and if the caller sent for the voicemail, and he leaved (or did not leave) a voice message, and did not press #, so the channel will stay open and this is not good specially if the call was coming from outside via the analoge lines (because the caller might hangup and the dahdi does not detect the hangup, so the channel will stay openned). How to let the voicemail hangup automatically after waiting for certain seconds (for example after 30 or 40 second), then to hangup or jump for the next line to run it? What is the parameter or the setting field in the freepbx that can resolve this (the voice mail message to be maximum for 30 or 40 second, after that to hangup even without pressing #). From where? 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
Re: [asterisk-users] Maximum concurrent calls using call files
I haven't used it, so can't recommend it per se; but as I understand it, iperf is a tool that can do that kind of simulation for you: http://iperf.sourceforge.net/ might be worth trying before you build your own modules. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Fri, Jul 6, 2012 at 4:01 PM, sathiish kumar sathiish.ku...@gmail.com wrote: I am planning on building a testing module which would spawn about 500 calls in order to test the performance of the network by transferring audio/speech files to end points at that juncture.Is it possible to spawn as many concurrent calls (or nearly concurrent calls) using just call files.Is there a limit as to the maximum number that could be spawned.? I tried doing this for about 20 calls and found that there is autofallthrough after a point of time.Is this a problem with my dialplan or is it because of the call files (i also get a warning which states that the ast_queue_frame:Exceptionally long queue length) Thanks, Sathiish -- _ -- 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] Maximum concurrent calls using call files
I've previously used iperf for my project and It can only simulate TCP/UDP traffic.and the thing is I'm testing this on a platform which does only RTP/SIP.I am not sure they've this facility.Anyhoo i wanted to know if it was possible to make such concurrent calls using Asterisk On Fri, Jul 6, 2012 at 3:00 PM, Stephen J Alexander sjalexan...@mpbx.comwrote: I haven't used it, so can't recommend it per se; but as I understand it, iperf is a tool that can do that kind of simulation for you: http://iperf.sourceforge.net/ might be worth trying before you build your own modules. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Fri, Jul 6, 2012 at 4:01 PM, sathiish kumar sathiish.ku...@gmail.com wrote: I am planning on building a testing module which would spawn about 500 calls in order to test the performance of the network by transferring audio/speech files to end points at that juncture.Is it possible to spawn as many concurrent calls (or nearly concurrent calls) using just call files.Is there a limit as to the maximum number that could be spawned.? I tried doing this for about 20 calls and found that there is autofallthrough after a point of time.Is this a problem with my dialplan or is it because of the call files (i also get a warning which states that the ast_queue_frame:Exceptionally long queue length) Thanks, Sathiish -- _ -- 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] FreePBX: using context other than the default context and the generation for the configuration
Dears;
Thanks for all the replies and help.
First of all, I am not looking to have the custom context only for outbound, I
need this also to separate the extensions into partitions, so I can have same
extensions in different contexts, also extensions in context A can not call
extensions in context B (as example).
Secondly, regarding to have the calls in the CDR (the outside and internal
calls), really i did not understand in the below example what I have to do?
For example:
exten = _2XXX,1,Verbose(Internal extension-to-extension call)
same = n,Goto(${EXTEN},from-internal,1)
Why to use the Goto? How the call will be done using my manual configuration?
As you are going to use from-internal then why you used the Verbose?
We need to know if we did a manual script, how the calls will be logged in the
CDR without using the from-internal? In other words, we will write manually the
Dial function and will not depend on the auto generated script.
Also why u used same? And u let it go for from-internal at sequence 1. Well, is
it always to go for sequence 1? Or I have to check the right sequence? What if
changed?
Regards
Bilal
umm Warren, yes including from-internal is the way of
getting all the
features,,,but in my experience the calls going out
using the dialplan
script we manually enter in our custome context don't
get inserted into the
FreePBX CDR and recording stuff !!
Okay, if you're writing custom dialplan to control outbound
calling, but
you want to utilize the FreePBX standard features, without
using custom
modules, you can do something like the following, adjusting
for your
specific situations of course:
[custom-local-only]
; local NANPA calling for area code 281
exten = _281NXX,1,Verbose(Outbound call from
local-only context)
same = n,Goto(${EXTEN},from-internal,1)
; extension-to-extension (internal) calling, assuming 2XXX
internal
extension plan
exten = _2XXX,1,Verbose(Internal extension-to-extension
call)
same = n,Goto(${EXTEN},from-internal,1)
[custom-long-distance]
; long distance NANPA calling, dial a 1 to dial anything
outside of a local
number
exten = _1NXXNXX,1,Verbose(Outbound call from local
and long-distance
context)
same = n,Goto(${EXTEN},from-internal,1)
; allow local calls also, without having to dial a 1
include = custom-local-only
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] FreePBX: How to hangup if the caller did not press # after the voicemail message
- Original Message - From: bilal ghayyad bilmar...@yahoo.com To: asterisk-users@lists.digium.com Sent: Friday, July 6, 2012 4:56:02 PM Subject: Re: [asterisk-users] FreePBX: How to hangup if the caller did not press # after the voicemail message I did one try and did not hangup: I called the extension and when the voicemail answered, I did not leave any message and stayed waiting .. waiting .. waiting .. It did not hangup from it self ! How much it stay waiting me to leave a message? There are settings in voicemail.conf to control this behavior. The maxsilence setting can be used to automatically end a recording: ; How many seconds of silence before we end the recording maxsilence=10 Note that depending on the situation, you may have to also tweak the silencethreshold setting. If VoiceMail detects sound on the voice frames being passed through it, it will treat that as someone 'talking' and not end the call. ; Silence threshold (what we consider silence: the lower, the more sensitive) silencethreshold=128 Why I am trying this? OK, the answer is: Because I am trying to see if the voicemail hangup even if it did not get from the caller the hangup, because this is happening when the caller is calling from outside to the dahdi (analoge) and when the caller hangup his mobile, the hangup does not reach (it is PSTN issue), so I beleive the voicemail should hangup from it self after timeout .. but I do not see this. Anyadvise? Regards Bilal --- Asterisk, and by extension FreePBX, automatically end the voicemail recording when the caller hangs up. You have some OTHER issue. Perhaps Asterisk is not detecting the hangup? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, July 05, 2012 6:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FreePBX: How to hangup if the caller did not press # after the voicemail message Dears; In FreePBX, when I select voicemail for the extension, and if the caller sent for the voicemail, and he leaved (or did not leave) a voice message, and did not press #, so the channel will stay open and this is not good specially if the call was coming from outside via the analoge lines (because the caller might hangup and the dahdi does not detect the hangup, so the channel will stay openned). How to let the voicemail hangup automatically after waiting for certain seconds (for example after 30 or 40 second), then to hangup or jump for the next line to run it? What is the parameter or the setting field in the freepbx that can resolve this (the voice mail message to be maximum for 30 or 40 second, after that to hangup even without pressing #). From where? Regards Bilal -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox or FreePBX?
Hi Tim, How about AsteriskNow? Thanks and BR, Anam On 7/7/12, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - Hi All; Based on what I have to use Trixbox or FreePBX? Can someone advise? Trixbox includes FreePBX as it's GUI. However, keep in mind it is a bastardized, forked version of FreePBX that has seen nary any new development or innovation in some time. At this point, for a standard PBX installation, my recommendations would be (in this order): 1. Elastix 3. AsteriskNOW 2. PBX In a Flash --Tim -- _ -- 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 -- Sent from my mobile device -- _ -- 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] Trixbox or FreePBX?
- Original Message - Hi Tim, How about AsteriskNow? Thanks and BR, Anam On 7/7/12, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - Hi All; Based on what I have to use Trixbox or FreePBX? Can someone advise? Trixbox includes FreePBX as it's GUI. However, keep in mind it is a bastardized, forked version of FreePBX that has seen nary any new development or innovation in some time. At this point, for a standard PBX installation, my recommendations would be (in this order): 1. Elastix 3. AsteriskNOW 2. PBX In a Flash Did you read #2 above? Erm, wait, #3 I guess? The list is in proper order but apparently my ability to make a numbered list is somewhat lacking today. :) --Tim -- _ -- 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] Trixbox or FreePBX?
Opss, sorry not read it carefully :(( On 7/7/12, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - Hi Tim, How about AsteriskNow? Thanks and BR, Anam On 7/7/12, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - Hi All; Based on what I have to use Trixbox or FreePBX? Can someone advise? Trixbox includes FreePBX as it's GUI. However, keep in mind it is a bastardized, forked version of FreePBX that has seen nary any new development or innovation in some time. At this point, for a standard PBX installation, my recommendations would be (in this order): 1. Elastix 3. AsteriskNOW 2. PBX In a Flash Did you read #2 above? Erm, wait, #3 I guess? The list is in proper order but apparently my ability to make a numbered list is somewhat lacking today. :) --Tim -- _ -- 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 -- Sent from my mobile device -- _ -- 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] call file and NFS server
Hi,
I have 100+ call file generated in other directory, and by using program, I
have moved 10-10 files in /var/spool/asterisk/outgoing, and call made
successfully.
Once all call completed, I found following error for all files...
[Jul 7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to
open /var/spool/asterisk/outgoing/100097_172.18.100.72.call: No such
file or directory, deleting
[Jul 7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to
open /var/spool/asterisk/outgoing/100098_172.18.100.72.call: No such
file or directory, deleting
[Jul 7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to
open /var/spool/asterisk/outgoing/100099_172.18.100.72.call: No such
file or directory, deleting
On Fri, Jul 6, 2012 at 8:47 PM, Steve Edwards asterisk@sedwards.comwrote:
On Friday 06 July 2012, Chandrakant Solanki wrote:
I have set the folder (callfile/Server{A/B}) permission to 777 as well
as call file permission to 777.
On Fri, 6 Jul 2012, A J Stiles wrote:
(By the way, you should have permissions 666 for a callfile, not 777.
Callfiles should not be executable.)
Whenever I see 777 (or it's Satanic cousin, 666) I see 'I don't really
understand ownership and permissions so let's just allow everything and
hope for the best.'
Do you really intend to allow every user and exploited program to be able
to create call files? (And if you've done this, you've probably created
other holes in your system's security.)
While 'opening the flood gates' is (IMO) a valid temporary debugging
technique to identify the source of the problem, the directories and files
should be owned by the user executing Asterisk and permissions should limit
reading to only users and groups that need reading and limit writing
to only users and groups that need writing.
I don't have any need or experience with call files on my production
boxes, but I suspect a successful implementation would include NTP and
creating the call file in another directory on the shared device and then
moving the call file to the outgoing spool directory.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Regards,
Chandrakant Solanki
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