similar to: Follow-me challenge

Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or

I am trying to launch Asterisk on a different directory with the parameter 'C asterisk -vvvvvvvvvvvvvvvvvvgc -C /etc/asterisk1/asterisk.conf Parsing '/etc/asterisk1/extconfig.conf': Found Resetting translation matrix UUID system initiated Parsing /etc/asterisk1/asterisk.conf == Parsing '/etc/asterisk1/asterisk.conf': Found Not changing threadpool size since new size 0 is

Hi Please help me understand about the below issue ? [root at asterisk1 ~]# /etc/init.d/asterisk restart Stopping safe_asterisk: [ OK ] Shutting down asterisk: [ OK ] Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted

Hi everybody, I'm testing asterisk@home 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from syslogd@asterisk1 at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user xxxx@yahoo.com could

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

Hi guys I am currently getting the following in my log asterisk1*CLI> Parsing '/etc/asterisk/manager.conf': Found asterisk1*CLI> == Parsing '/etc/asterisk/manager_custom.conf': Found asterisk1*CLI> == Connect attempt from '127.0.0.1' unable to authenticate Can anyone tell me why asterisk would not be able to authenticate it's self?

Has anyone else installed the official 1.4.0 release? I have, and it installed very easily. However, I don't have any of my usual command line tools for monitoring and debugging zap channels and PRI lines: asterisk1*CLI> pri show span 1 No such command 'pri show' (type 'help' for help) asterisk1*CLI> Ditto with zap stuff: asterisk1*CLI> zap show

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > fax channel bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1

Hi, I need some information on DID and Callback. Please read-on: Question on DID (User1 Calling User2 via normal Telephone line and sending its CLI: Connectivity is as below: User1 ==PSTN==> DigiumE1/Asterisk1 ==INTERNET==> DigiumE1/Asterisk2 ==PSTN==> User2 1. Can User1 make a single stage call to User2 via Asterisk1? Currently User1 is able call User2 on Two Stage basis (Asterisk

Hi, Could anyone knows what went wrong with the error below result of installation of libsupertone. [root@asterisk1 latest]# tar xvf libsupertone-0.0.2.tar libsupertone-0.0.2/ libsupertone-0.0.2/AUTHORS libsupertone-0.0.2/Makefile.am libsupertone-0.0.2/COPYING libsupertone-0.0.2/config/ libsupertone-0.0.2/config/ltmain.sh libsupertone-0.0.2/config/missing libsupertone-0.0.2/config/install-sh

Hello, I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk as followed: [SIP_BD1] type=peer qualify=yes host=192.168.0.254 disallow=all context=from-pstn allow=h723 and inside the quantum I change the config sip useragent to 5060. Up to this part if I run sip show peers, I got: asterisk1*CLI> sip show peers Name/username????????????? Host??????????? Dyn Nat ACL

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

Hello, I've got the following situation: --------- Asterisk1 ------------- SER ---------- other world | | ----------Asterisk2 ----------------- In addition i'm doing a sort of "vhost" on the asterisk machines, so there could be 3 seperate companies using 1 asterisk box. If an asterisk1 user calls

Hi!I've a question: I've 2 asterisk, I want pull the ethernet wire and then reconnect it after 5 second, using the VRRP protocol, where must I set the IP for the connection goes on the second asterisk? I want this: I call to asterisk1, then I pull the ethernet wire down, vrrp makes up the other asterisk but not the audio streaming...the callers are always pointed to asterisk1, but for the

I have installed Asterisk 1.2.18 am am trying to install chan_capi. The current RPM ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm installs but Asterisk dies on startup. The following appears in the log: May 27 03:28:18 asterisk1 kernel: divas: Diva Server V-4BRI-8 IRQ:7 SerNo:25290 May 27 03:28:18 asterisk1 kernel: divas: started with major 252 May 27 03:54:17 asterisk1 init:

Hi I am trying to make this setup work phone1---g729---asterisk1---sip---asterisk2---g729---phone2 I have tried several configurations but none worked I keep getting transcoding errors I have installed one g729 licence on each asterisk, but I can't verifiy because the show g729 command is not available, I use 1.2.17 Do I need 2 g729 licences per asterisk ? Do I need to register

I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax (.tiff) from the first asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an INVITE with audio G.711. Asterisk2 accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to Asterisk1. But, Asterisk1 responds with "488 not acceptable here". I double check t38pt_udptl = yes in

Hello. I have just installed Asterisk on my HP DL140 Fedora Core 1 Server. The server has two interface cards active. I have been unable to fix this error although all DNS lookup works fine. my /etc/host.conf contains order bind,hosts /etc/hosts contains # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1

This is not really a problem just something I noticed in my testing.. When two or more Asterisk servers are connected by IAX2 trunks it does not make use of any "shortest path" type system.. (maybe this is still planned somwhere down the line, but may come in handy to those who have multi asterisk installations) Here is the setup.. UA1--- Asterisk1----[IAX2 Trunk]---Asterisk2---UA2

Hi all, I need to test the following scenario: +-----------+ +-----------+ | asterisk 1| | asterisk 2| +-----------+ +-----------+ | | | | _______|__________________|___________ | | | | | | +-------+ +-------+ | ATA 1 |