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Or: "An IAXy by any other name is still an IAXy" http://nextalarm.com/abn.jsp These folks are selling alarm supervision for us phone-line impaired folks. What's scary about this is that we probably need to install a fire/smoke detector with the IAXy ;-) IAXy users on this list -- who of you would trust their home/business security to an IAXy? Is there a s...

We have a script which executes "asterisk -n -r -x ....." periodically against the running server, to check the status of a few things, and pipe the output to a file. With prior versions of Asterisk this worked fine, but having just upgraded to 1.2.9, we are finding that if the output is lengthy, then Asterisk seems to terminate before fully flushing stdout. Is this a known bug, is

If anybody here is a sixTel customer, can you share any tips & tricks for getting in touch with anybody there? They are absurdly hard to get a hold of, particularly when you have a technical issue needing to be resolved. If anyone has any phone numbers other than their main 800 line, I'd sure appreciate it. Thank you, Bryan -------------- next part -------------- An HTML attachment was

If I issue a "Goto" command to an invalid extension within a valid context, then the "i" extension is invoked properly. But, if I issue a "Goto" command to an invalid context, then no "i" extension appears to be invoked. Is this intended behavior? How can I get it to go to "i" if the context being Goto'd is invalid? Thank you, Bryan

We are using Asterisk 1.0.7. We have this scenario: IAX2 user comes in to Asterisk, dials an extension, and transfers to a SIP user. The dial command is simple, looks like this: exten => 300,1,Dial(SIP/300) Extension 300 is a legacy PBX device operated by touchtones. The user (coming in over IAX2) is trying to drive this PBX using touchtones. But the trouble is, by the time the touchtones

We are tracking the following situation: SIP client connects to our Asterisk server, and then connects to another SIP user. Re-invite is OFF, so Asterisk is in the middle of the whole conversation. When one SIP client sends DTMF tones, the SIP client uses RFC2833 to send the tones to the server. (This is correct). The server then sends RFC2833 tones out to the other SIP client. The problem is,

I'm trying to gauge the amount of overhead for idle users (NOT in the middle of a phone call) per user, per server. These are a combination of SIP and IAX2 clients, with "qualify=yes". On, for example, a dual 2.4 Ghz Pentium server (with plenty of RAM), how many hundreds, or thousands (rough ballpark) of clients can be supported? Again, these are mostly idle, and I'm interested

We've been seeing a pattern over the last couple of weeks with our Asterisk servers (1.0.9). The number of ports in use (RTP) seems to grow by two every minute or so. Eventually the server will run out of allowable files open and crash. We are resetting the server once per day to prevent this from occurring. Running "lsof" shows the end of the list like this: asterisk 26733 astx

We are using TDM40B's to connect some devices to Asterisk which depend on caller information arriving as ANI, rather than as Caller ID. I am unsure if the TDM40B supports this in the first place, and if so, I am unsure how to configure it so. I've searched the wiki but couldn't find anything. Can someone please confirm whether or not this is possible? As a fallback I could reconfigure

We have several Asterisk servers (1.6.2.15) all configured for Realtime, all backed by the same database. The Asterisk servers are all listed under DNS SRV records, and SIP ATAs find us this way. Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as "regseconds", "lastms", "ipadr", etc. However, with

We are using PostgreSQL real-time connector (res_config_pgsql) with Asterisk 1.6.2.15. From time to time, we need to reset our PostgreSQL server, causing all active DB connections to close. While other packages in our system re-connect gracefully when this happens, Asterisk appears to not bother trying. It instead goes into an endless loop complaining that the connection has closed. Question --

We have a customer trying to dial through our server, and our server is throwing tons of these log messages: Jul 27 14:21:02 NOTICE[29210]: rtp.c:431 ast_rtp_read: RTP: Received packet with bad UDP checksum Is it pretty certain, that these are caused by a bad or misconfigured router along the path, or something else network-related? As opposed to the SIP hardware itself? The SIP ATA is the same