similar to: Working example of "switch"?

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

I have an Asterisk 0.4.0 install working with two grandstream budgetone 100 phones, gnophone, and kphone. This is a private network segment (172.17.x.x), with the PBX configured on my outbound firewall which has a public address (66.x.x.x). - I can make calls between phones - all extensions are working. - I can make IAX calls to IAXTEL. No problems (apparently gsm only) - I can call SIP phone

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems to be having problems. The Grandstream and sip.conf were set to RFC2833 now with that setting I get extra digits during "Mailbox" and "Password" phases. 222001 instead of 2201 for example. When both are changed to "SIP info" there is no problem. But what is the new setting "DTMF Payload

i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two

On Zed''s suggestion, I caught two new spinning mongrels and sent a SIGUSR2. The code appears to be stuck in read_multipart for both processes: # kill -USR2 6109 ** USR2 signal received. Thu Sep 21 14:55:39 EDT 2006: Reaping 1 threads for slow workers because of ''shutdown'' Thread #<Thread:0x419d7ce0 run> is too old, killing. Waiting for 1 requests to finish, could

First off: Our clusters are LVS balanced Apache 2.2.3 + mod_proxy_balancer + gem mongrel 0.3.13.3 / mongrel_cluster 0.2 + memcached / gem memcache_client + gem rails 1.1.6 on debian boxen, and a pgcluster backend. On 2 of our deployed clusters, we are getting the "spinning mongrel" problem. As the clusters are very low volume right now, it takes days to collect a spinner, making it

On the Granstream 102 box that I have in front of me, there is a "feature list" on the side. One of the features has grabbed my attention: " - optional voice encryption (model 102D)" Now, digging through Grandstream's site, I see that it's not offered quite yet. However, sending mail to their standard "information" email address has resulted in no

Dear Xapian developers, I am happy to announce that the Xapian-Ruby bindings now pass all tests defined in smoketest.rb. (These tests are mostly drawn from smoketest.py; a few tests that are not applicable to the Ruby version have been removed.) I have also added Olly's patch to integrate the Ruby bindings into the automake/configure scripts. The latest version is available by anonymous

Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the "domain" field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk

Dear Xapian community, We've been looking for a search engine to integrate with our Rails applications. Xapian seems to be the most robust and mature of any open source solution out there. But, there were no Ruby bindings. So... I've started work on Ruby SWIG bindings for Xapian. My work is still alpha-quality; expect bugs and expect API changes before we're done. It's

I've noticed some strange results from the stemmer in the Ruby port: irb(main):003:0> @stem.stem_word("anybody") => "anybodi" irb(main):004:0> @stem.stem_word("swimmingly") => "swim" irb(main):005:0> @stem.stem_word("fiercely") => "fierc" irb(main):006:0> @stem.stem_word("fraudulently") =>

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID

Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK,

I''m creating a new domU using ''virsh create <xmlfile>'' without specifying the MAC address within the XML file, under the assumption that the dom0 or hypervisor generates the MAC address for me. Next I''d like to get the MAC address that was assigned to this new domU, and I don''t see a way to get that information from the command line via virsh

I have followed a combination of instructions from this post and have been able to get an Ubuntu paravirtualized instance running under snv_96. Is there a way to "list" the fact that it is paravirtualized? For example, xm list, virsh list, etc. thanks -- This message posted from opensolaris.org

Hi is there any ritual sacrifice a newbie has to perform to be welcome on this list? I am new to this whole PBX thing in general and Asterisk in particular. I had hoped that the community on this list would welcome a newbie like myself and help me with some answers to my stupid questions, but somehow it seems to me that nobody likes to respond to somebody who appears to be a complete

This is important so please read this message very carefully. There is a DoS for Ruby''s cgi.rb that is easily exploitable. The attack involves sending a malformed multipart MIME body in an HTTP request. The full explanation of the attack as well as how to fix it RIGHT NOW is given below. Most of the work was done by Jeremy Kemper and Jamis Buck. They did all the work of building the

Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations?