similar to: Clustering

Hi everyone, Ive just installed a bunch of Snom 360s, and now having a NIGHTMARE of problems! Ive got a receponist phone with a extra sidecar on it. And when she gets 2+ calls and puts them on hold, when she goes to transfer them out the calls on hold get merged together. Somehow the calls on hold get merged and not to the extension needed!! Any help on this would be great guys, that would be

Hey list, Im in this HUGE crisis. Im trying to get a Polycom 601 with two expansion modules to work. I need the XML config files I guess. Does anyone have these I can have? Im trying to get this phone up and running, and haveing MUCHO problems, can someone help me out!! Im sure if I see the configs I can see how it works, just need those XML files!! The ones from the 501 that I have dont seem to

Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug.

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

Hey, You know, the Digium guys said both are good. They said the the DNS method is better because you dont have the extra point of failure (SER) but said the SER method is better in that it gives you more exact control in the handling of the calls and registration. They did acknowledge there would be a possible downtime only for incoming calls to users with dynamic IPs if the

Hi List, I cant seem to find the setting that changes this! You put a person on hold, they are on hold like normal, but after a few seconds the phone will then start having dialtone coming from the speakerphone, really weird!! Anyone know how to fix this? I see where it could be nice, but in this case, we just want them on hold is all, no dialtone! Any help would be great! Thanks! Ron

I know someone who's at VON this week. Apparently Mark Spencer was up there talking about how Asterisk supports SRV. Sounds like vaporware to me. > -----Original Message----- > From: David Thomas [mailto:punknow@gmail.com] > Sent: Thursday, March 16, 2006 11:54 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Feedback from VON

Hi List, Ive got a bunch of Avaya IP9640 that we want to convert to SIP and then hook up to Asterisk so we can dump this overpriced Avaya system. Ive got ahold of the SIP firmware, but I cannot find anything on how to convert the phone itself to SIP, when I go into setup mode it wants a "command" which im guessing is the program code like the rest of the Avaya systems works. Has anyone

Hi list, Ive got a server with 6 ports on it (4+2 port card) we have a DS3 delivering all voice DS1's to us. Carrier has a trunkgroup for the first 8 span (we only have the first 6 plugged in right now). Everything works fine until we fail the primary D channel (D's are on 24,48) the secondary then picks up and outbound calls do not work, if we reboot Asterisk the D on 48 comes up and it

Hi List, Im planning on setting up asterisk for a large scale enviorment, with multiple sites. We will be doing quite a bit of inner office calling at each site, and want to place a smaller scale * box at each site with no PRI's, and have that connect to our main * servers at our data center that will have the PRI connections. Can this be done? I havent seen to much of this on the mailing

> /usr/bin/ld: /usr/lib/mysql/libmysqlclient.a(libmysql.o): relocation > R_X86_64_32 can not be used when making a shared object; recompile > with -fPIC > /usr/lib/mysql/libmysqlclient.a: could not read symbols: Bad value > collect2: ld returned 1 exit status Never seen that before. What OS are you using? What version of MySQL? > Dec 29 12:58:22 WARNING[22260]: No such

Hi Everyone, I have a routing situation where Tinc looks like it could come in extremely useful, but I have a query I hope someone can cast an eye on, as I'm unsure whether Tinc can help me here. I currently have a Quagga OSPF linux router which connects LAN A to LAN B over the quickest available of two routes (both routes at both ends connect to Quagga boxes to prevent collisions). One of

Can someone show me how to set up DUNDi, I will be using it to connect 14 asterisk servers internally. I don't want to use it on the external world. If anyone has any examples of connecting 2 or 3 (if their is a difference) machines in a DUNDi co-operation that would be helpful. Johnathan Falk Network Administrator Clinton Community Schools

Hello, does anybody know if quagga for CentOS 5.3 is compiled with -enable-multipath? I want to implement ECMP over 2 ISP. Thanks, Cristi Carstea

Hello, I use both ser/asterisk . In fact i wish asterisk to forward all the sip requests which are not handled by domain=domain.tld in sip.conf Here is a diagram: The sip agents use the Sip proxy as an outbound sip proxy and domain=domain.tld . When the sip agents dial sip:user@otherdomain.tld so the request is sent to sip proxy and so to Asterisk. I wish Asterisk to Look up the

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

Hi, I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, freetds-bin, but, when I run configure and then make menuconfig in section "Call Detail Recording" -> "cdr_tds" it's "disabled". It only writes that "Depends on: freetds(E)". On another server (same

I've been trying to use a set-up whereby I have several TA's connected to an Asterisk server (1.2.4) and they act like they're in a hunt-group i.e. try the first, if busy jump to the next etc. in my extensions.conf I had something like [inbound-trunk] exten => 441234123456,1,Dial(SIP/s1a,20,r) exten => 441234123456,102,Dial(SIP/s2a,20,r) exten =>

Hi list, I have a problem I thought was simple first, but now I''m stuck. In a nutshell, it''s about redundant uplinks at an outside location. Crude ASCII-Art follows: Internet | | +------------+ | cisco with | | uplinks | +------------+ | | ATM interface +----------+ ... | alvarion | | | wireless

I''m getting an error when shorewall is trying to add the default routes for my multi-isp configuration. I''ve attached a shorewall dump... If anyone can give some input I''d appreciate it. RTNETLINK answers: Invalid argument ERROR: Command "ip -4 route replace default scope global table 254 nexthop via 67.110.119.245 dev eth3 weight 1 nexthop via 66.29.181.113