FreePBX: How To Hangup If The Caller Did Not Press # After The Voicemail Message

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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…

Asterisk Users 3.2 years ago 3 Answers

FreePBX: Using Context Other Than The Default Context And The Generation For The Configuration

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Hi All;

If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal).

Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I have to write the configuration in my hand and it will not be autogeneration, correct?

In this case, the Phone will not have any of the features that I am going to add…

Asterisk Users 3.2 years ago 18 Answers

Sip And Extensions

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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…

Asterisk Users 3.2 years ago 2 Answers

AST-2012-011: Remote Crash Vulnerability In Voice Mail Application

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If a single voicemail account is manipulated by two parties simultaneously, a condition can occur where memory is freed twice causing a crash. Management of the memory in question has been reworked so that double frees and out of bounds array access do not occur. Upgrade to the latest release. Affected Versions

  • Product Release Series
  • Asterisk Open Source 1.8.x 1.8.11 and newer
  • Asterisk Open Source 10.x 10.3 and newer
  • Certified Asterisk 1.8.11-certx All versions
  • Asterisk Digiumphones 10.x.x-digiumphones All versions
Corrected In
  • Product Release
  • Asterisk Open Source 1.8.13.1, 10.5.2
  • Certified Asterisk 1.8.11-cert4
  • Asterisk Digiumphones 10.5.2-digiumphones

VoIP News 3.2 years ago 0 Answers

AST-2012-010: Possible Resource Leak On Uncompleted Re-invite Transactions

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Asterisk Project Security Advisory - AST-2012-010

Product Asterisk Summary Possible resource leak on uncompleted re-invite transactions Nature of Advisory Denial of Service Susceptibility Remote authenticated sessions Severity Minor Exploits Known No Reported On June 13, 2012 Reported By Steve Davies Posted On July 5, 2012 Last Updated On July 5, 2012 Advisory Contact Terry Wilson CVE Name TBD

Description If Asterisk sends a re-invite and an endpoint responds to the re-invite with a provisional response but never sends a final response, then the SIP dialog structure is never freed and the RTP ports for…

Asterisk Users 3.2 years ago 0 Answers

Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, 10.5.2-digiumphones Now Available (Security Release)

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The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones.

These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones resolve the following two issues:

* If Asterisk sends a re-invite and an endpoint responds to the re-invite with a provisional response but never sends a final response, then the SIP dialog structure is never freed and the RTP ports for the call are never released. If an attacker has the ability to place a…

Asterisk Users 3.2 years ago 0 Answers

Touch Command Not Behaving For Future Calls In Asterisk 1.4.41

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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…

Asterisk Users 3.2 years ago 3 Answers

OT - Multi Function Printer With One Touch Scanning/emailing

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Hi,

I'm curious about the availability of Multi Function Printers with the following feature : - user feeds paper sheets in - user dials a phone number (0123456, for instance) then a hits single button - the result is that the paper sheets are scanned into a file which is emailed to a given address such as 0123456@myfaxgateway.com (where myfaxgateway.com is a fixed and configured address).

Is this a common feature ? Last time I checked, MFP's alphanumeric diaplan was either oriented to digits or letters typing, and of course, scanning feature implied letters typing mode.

Regards

Asterisk Users 3.2 years ago 6 Answers

OT - Integration With Building Intercom Systems

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Hi,

Now and then, I'm facing environments in which it could be helpful to integrate building intercom systems with Asterisk.

Those intercom systems are made of : - a main panel, showing company names and equiped with a speaker, a microphone and an optional video cam - a doorstrike - several intercom phone with an optional monitor and with a button triggering door's opening, - maybe other hidden components I'm not aware of.

I've noticed that these intercom phone are connected to the main panel through a 4 or 5-wires cable.

More precisely, for the video case I have in mind, it's a 4-wires…

Asterisk Users 3.2 years ago 5 Answers

Sip Set Debug On Always Showing Error

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dear

please Help. I am continously getting this message after "sip set debug on". and not getting clear voice from both side.

<--- Transmitting (NAT) to 122.163.193.94:1893 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106:5060 ;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received2.163.193.94;rport93 From: "2002" ;tagZ1cc54c To: "2002" ;tag=a64f1f102 Call-ID: 8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0 CSeq: 245 OPTIONS Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0

<------------> Scheduling destruction of SIP dialog '8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '6636e2ee56cce3b0e2287ebe51962c84@0:0:0:0:0:0:0:0' Method: OPTIONS Really destroying SIP dialog '4ec2032c2b9a58e1cf2d1afa70c1970b@0:0:0:0:0:0:0:0' Method: OPTIONS

Asterisk Users 3.2 years ago 2 Answers