similar to: Extensions Reload | Asterisk Freezes ? 1.4

Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in

Hello, I am attempting to setup Asterisk to work with Gtalk. I am using the following versions: Slackware Linux 12.0 Asterisk 1.6.2.9 GNU TLS 2.8.6 Iksemel (svn v25) OpenSSL 0.9.8o It all compiles however about 10 seconds after starting Asterisk it crashes. To mitigate this issue I have moved jabber.conf to another directory and then Asterisk starts up. So I assume the issue is with this

Hi all, I think I've run out of options in terms of what I know about this. I have created a user called asteriskuser and granted all privileges to the asteriskcdrdb database. Then I created the table via the cdr_mysql.txt file. I have edited the cdr_mysql.conf file to reflect this, and added load => cdr_addon_mysql.so after compiling it from the latest CVS. If I check the

We are using vpn routers to connect home users back to our office network. Basically, shipping a mikrotik router that 'calls home' and establishes a vpn connection for the pc and phone that are connected to the mikrotik... user plugs router in, plugs phone and computer into router, and that traffic is encapsulated back to our office... simple and straighforward. We would like to remove

Forgive me if this is a FAQ question but I didnt see anything on the website of forum spelling out the difference between 1.4.x and 1.6.x Obviously 1.6.x is in development. Is it stable enough for production use? What are the new features being implemented in 1.6.x? Will Cepstral work with 1.6.x? Thanks, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All Can you please let us know how we can modify our Asterisk "extensions.conf" file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one

Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a "show uptime" I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 voip2*CLI> show uptime System uptime: 15 hours, 55 seconds

Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed

Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? Regards Bilal

Hi all, Every time I reload my asterisk it fall down and the following message appear on log: parse error: No category context for line 7 of /etc/asterisk/chan_dahdi.conf If I comment that line, it change to other line. There are some thing wrong with my dahdi? Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda --------------

Hi All; First of all, I am looking to have the H323 Gatekeeper service available at Asterisk, and really I do not know if 1.4 or 1.6 or 1.8 started implementing H323 gatekeeper functionality or not? Until 1.4.26.2 version, there is no h323 gatekeeper functionality. So, any implementation for this feature has been done in the other versions?

Hi, I think ive found a bug but need someone to double check. Whenever I issue a "reload" in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed...

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Leif, Did You faced that with the exact same Chrome version? Since this version is the only one having this issue. I have had this kind of arrangement (intro + live stream) for decades. Technology changes but the idea is the same. I have tested a lot of hardware and combinations. I do have a fail over stream (with different specs) and that hasn’t been an issue at all. I do not know but I

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header

> I'm pretty sure that is incorrect. The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.com

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

https://bugs.freedesktop.org/show_bug.cgi?id=54437 Bug #: 54437 Summary: linux-nouveau2.6 (3.6.0-rc4) : GTX580 : Xorg freezes when using accel Classification: Unclassified Product: xorg Version: git Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: critical

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss