similar to: Need good explanation on contexts and extensions

Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have

I'm trying to clarify contexts and their uses. I do have a good general understanding of them. My question is about "undeclared" and "non-existant" contexts. If I have a block somewhere (in sip.conf, for example), and it has no "context=thiscontext" field, does it just automatically use the "default" context? Or is this settable? (I see there is an

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

Has anyone started work on a ''top''-like tool that would show the current CPU and memory usage of domains? I''m thinking of writing something really simple in python, starting from listdoms.py, but want to make sure that someone else isn''t already there. Steve -- Stephen G. Traugott (KG6HDQ) UNIX/Linux Infrastructure Architect, TerraLuna LLC

Hi All, I''ve been looking through the code and list archives but haven''t found this yet... From dom0, how do you cause a virtual domain to gracefully shutdown? It seems like the machinery is there somewhere, because the hypervisor can do it to dom0... For reference, in UML you do this by putting this in /etc/inittab: ca:12345:ctrlaltdel:/sbin/shutdown -h now ...and then

Hi Folks I have had a look through the Features list, and I see that the system does support an auto attendant, however is it possible to have say 5 telephone numbers that a person would dial and have 5 different messages I.e Dial 555-1121 and you get a message for companyA call centre Dial 555-1131 and you get a message for companyB call centre Dial 555-1141 and you get a message for companyC

Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include <filename>) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. Regards, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am looking for examples of the extensions.conf that puts all incoming calls into a context where extensions can be dials, and all phones in a context where extensions and outside calls can be dialed. i.e. I have seen: [incoming] include => sip-extensions [sip-extensions] include => longdistance [longdistance] .... Doesn't this allow any internal callers to make external

Can realtime extensions be used within AEL contexts? I've tried without success. Also did some Googling and looking around on the Wiki, but couldn't find much. Thanks, Mike Clark

Hi, I'm taking a straw-poll to see who out there is planning on going to AstriCon. I would like to hear from both new members of the community and gurus. What kinds of things would you like to see at an Asterisk Conference? What topics are good BOF (Birds Of a Feather - informal discussion group) fodder? What parts of Asterisk require the most attention? FYI - AstriCon is October 12 - 14

The system is Mandrake-9.1 Linux wih gcc-3.2.2. I am trying to compile xenolinux-2.4.24 (with vanilla sources from ftp.kernel.org for linux-2.4.24). The steps of mkbuildtree ARCH=xeno make menuconfig ARCH=xeno make dep produce no errors, but ARCH=xeno make bzImage results in following error messages. Any pointers will be appreciated. -ishwar --- gcc -D__KERNEL__

Hello, We have finished our tests of the new Digium firmware on the quad T1 cards(TE405P/TE410P). Overall it is a big improvement over the version 1 firmware. Here's the review: http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html MATT--- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? Angus

Hi For a customer, I am trying to setup 3 different companies on one asterisk box, and I need to assign extension 200 in three different companies. I was using different contexts, but was unable to get it to work. So, my basic question is - In Asterisk, Can we have same extension number in different contexts? For example: [Context_company_1] exten => 200,1,,,

I was wondering if the following was possible: 2 separate incoming contexts. The first will be used when there is a secretary present. The second will be used when there is no secretary. I know that this can be done using includes and specifying the time in which each separate context would be included. However, I would like to be able to switch them from the reception telephone. For

Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client "subscribes" to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Any ideas are welcome.

I'm slightly confused about how SIP security and authorization works. I've looked at the Wiki (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) , but it's, um, flawed: > As of Asterisk 1.2, there is no reason to actually use 'user' entries > any more at all; you can use 'type=peer' for everything and the behavior > will be much more

Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, "default" is always being used. The output of "sip show peers" shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the "default" context.

I was wondering if anyone knew of some resources I could look at to learn about how rails works internally. I''ve been looking through the code but I was wondering if there were any high level explanations of how it works overall. Thanks in advance, Ruben --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups

Hello! I'm mounting Windows 2000/XP disk drives from Linux RedHat 7.3 using smbmount. No problems, the disks are mounted. However, it disregards the "uid" and "gid" options and leaves the smbfs owned by "root" giving the "uid" user read-only access and not read-write as expected. Adding the "rw" doesn't help either.