What is the best CRM solution for Asterisk, which is easy to deploy and Open Source? Well, there are some good options out there but the reality is that It's not possible to determine which one is "better". Nevertheless you can always evaluate and consider which one fits your needs, that's why among the different CRM solutions around, I would like to call your attention to Zurmo CRM, which have just released their 1.0 GA version. The first thing about Zurmo is that the way they care about the community's feedback, contributions and suggestions leads me to believe…
Counting any Open Source package is difficult for many reasons. There is probably not a reliable answer to this question since there are at least 4 major flavors of Asterisk out there (1.4, 1.6, 1.8, 1.10) and open and commercial source. It is reliably > 10,000 and quite possibly over 100,000 or even over 1 million. The Asterisk folks might be willing to tell you how many downloads have been done from http://www.asterisk.org , but that wouldnt tell you the real number. Maybe a good start point for an estimate would start at 200,000+ if you are including all of the versions and types. But then we might still think…
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
- Product Release
- Asterisk Open Source 18.104.22.168, 10.5.2
- Certified Asterisk 1.8.11-cert4
- Asterisk Digiumphones 10.5.2-digiumphones
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
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…
I will be running an Asterisk SIP Masterclass - the last one - in Barcelona in June. During this week, I will organize a dinner for everyone working with or interested in Asterisk, Kamailio and other Open Source platforms for realtime communication. It's June 13th somewhere in Barcelona - location will be announced later. You pay our own dinner (unless we can find sponsors) and enjoy the geeky company for free!
To join the event, use this Facebook event https://www.facebook.com/events/307548349321608/
See you in Barcelona!
In the Skinny channel driver, KEYPAD_BUTTON_MESSAGE events are queued for processing in a buffer allocated on the heap, where each DTMF value that is received is placed on the end of the buffer. Since the length of the buffer is never checked, an attacker could send sufficient KEYPAD_BUTTON_MESSAGE events such that the buffer is overrun. Now, the length of the buffer is now checked before appending a value to the end of the buffer. Affected Versions:
- Product Release Series
- Asterisk Open Source 1.6.2.x All Versions
- Asterisk Open Source 1.8.x All Versions
- Asterisk Open Source 10.x All Versions
A user of the Asterisk Manager Interface can bypass a security check and execute shell commands when they lack permission to do so. Under normal conditions, a user should only be able to run shell commands if that user has System class authorization. Users could bypass this restriction by using the MixMonitor application with the originate action or by using either the GetVar or Status manager actions in combination with the SHELL and EVAL functions. The patch adds checks in each affected action to verify if a user has System class authorization. If the user does not have those authorizations, Asterisk rejects the action if it detects the use of any…
Hello, the following is an email from Daniel, of Kamailio project: "ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry: * http://www.itspaawards.org.uk/ Open Source VoIP Projects won a special category this year, Members' Pick, for providing a real value to VoIP Industry. I had the chance to attend the event in London and I have been selected to pick up the award. I made a news on the website of the project I am mainly involved in (Kamailio) with more details:
I've tried upgrading one of my servers with yum update to the latest dahdi/asterisk, and found out that my 4th gen TE410P is failing the dahdi init with
Running dahdi_cfg: DAHDI startup failed: Input/output error
Rolling back to 2.5 restores the normal operation, and reading the dahdi 2.6 change log I think I'm hitting this bug fix with my mobo/card combo?
2011-12-14 19:02 +0000 [r10379-10380] Shaun Ruffell
With dahdi 2.6 I'm getting this: #cat /proc/interrupts 209: 1 0 IO-APIC-level wct4xxp No interrupts?! #dmesg kernel: ACPI: PCI Interrupt 0000:02:01.0[A] -> GSI 24 (level, low) -> IRQ…
Use jabberd and qmail.
On 2/3/12, bilal ghayyad
> Hi All;
> Any advise for a good collaboration solution (open source)? Chat + Email
> call center.
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