similar to: Connection Problem

I'm trying to put together a package of asterisk-head, spandsp, and app_rx,tx fax. I can get everything to compile: spandsp-0.0.2pre20 asterisk-head (cvs co -r HEAD asterisk) the app_rx/tx from soft-switch.org in the 1.1 folder However, asterisk complains that there is unused symbols when running /usr/sbin/asterisk -vvvvvvvgc ARGH.. Does someone have a package with files that I could try?

All, I have downloaded and installed openh323 as per the documentation. When the machine now reboots safe_asterisk just keeps restarting. If I start another session and just load asterisk -vvvvvvvgc asterisk loads. If I enter noload chan_h323.so in the modules.conf then safe_asterisk will kick in. Not 100% on Linux yet but I have added the environment variables info into /etc/profile so they

Thanks for the info so far. I am still trying to asterisk'ize my ML9.2 firewall box and can't get the external SIP registration to work. If I hook up my Sipura directly to the WAN it registers OK. This is the message I get from asterisk: Mar 8 21:03:07 NOTICE[196621]: chan_sip.c:3140 sip_reg_timeout: Registration for '263872@192.246.69.223' timed out, trying again If tried

I appear to be seeing the performance of a local ZFS file system degrading over a short period of time. My system configuration: 32 bit Athlon 1800+ CPU 1 Gbyte of RAM Solaris 10 U6 SunOS filer 5.10 Generic_137138-09 i86pc i386 i86pc 2x250 GByte Western Digital WD2500JB IDE hard drives 1 zfs pool (striped with the two drives, 449 GBytes total) 1 hard drive has

My BT100 won't register with my Asterisk server, it always comes back with a 403. I've included my sip_additional (only one to to have the username 2201) and a portion of the sniffer trace (packets 27 & 28). This has me puzzled as I have my SPA-3K working (incoming and outgoing). On my BT100 I get no dial tone, I can't call it (asterisk says the extension is busy) but I can call

I've searched and tried several ideas (na.action. and other things), but I can't see to figure this out. I'm guessing this is so simple I'll feel foolish for asking, but here goes. Thanks, L.A. Dataset$Rcil=with(Dataset, ifelse(Rpr >= .95, Dataset[,"percentchgn"], NA)) Dataset$LLCI<-with(Dataset, ave(Rcil, LEAID, Property, FUN=function(x)max(x))) LEAID

Hi, When I dial from my sip device to the extension 1234 which is linked to the ALSA console driver the call fails with the message "No channel type registered for 'ALSA'" (see below). I would like to have the console autoanswer for paging. However when I call from the console to the sip device the call completes fine. I alias alsa device hw:1,0 to card1 in /etc/asound.conf

I have Centro 5.4 it named starts at bootup but fails to restart when I give command service named restart and it there is no any error in syslog messages. With reload command I get the following message in syslog :- Nov 25 14:03:30 unitedinfotechs named[2201]: using default UDP/IPv4 port range: [1024, 65535] Nov 25 14:03:30 unitedinfotechs named[2201]: using default UDP/IPv6 port range: [1024,

I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten => 2201,1,Dial(sip/2201@gs1.uucp,20,) exten => 2201,2,Voicemail(u2201) exten => 2201,3,Hangup exten => 2201,102,voicemail(b2201) exten => 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

https://bugzilla.mindrot.org/show_bug.cgi?id=2201 Bug ID: 2201 Summary: -R tunnel disappears Product: Portable OpenSSH Version: 6.1p1 Hardware: Other OS: Linux Status: NEW Severity: normal Priority: P5 Component: ssh Assignee: unassigned-bugs at mindrot.org Reporter:

Sorry if this comes in twice. Wasn't subscribed first time :-( Anyone help me here...... It worked once :-( I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server.

Using CVS update from 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': Found Sheriff*CLI> Disconnected from Asterisk server -- Dave Cotton <dcotton@linuxautrement.com>

Dear CentOS Doc, We are planning to organize CentOS Dojo at Bangalore, India - http://wiki.centos.org/Events/Dojo/Bangalore2014 - in the month of Nov. I request you to grant `edit` permission to update my homepage - http://wiki.centos.org/DominicGeevarghese - for further followup with speakers/presenters. Thanks in advance. Cheers, -- Dominic Geevarghese 2C7C 2AFC B327 6D12 FB55 29D8 046D A4F6

Good Day again list, Encountered another problem in the ACD queue... If I use the ADDQueueMember to dynamically add members as foolows, exten => 403,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM}) lets assume I called extension 403 from my extension 2204. then a caller (extension 2203) enters into the techsupport queue I am able to receive the support call on my phone (extension 2204

Hello. Im trying to get Fax-to-email working. I've installed Rx and txfax, spanDSP and every package needed. I've done everything on this page (altough, some bash-scripting problems): http://www.voip-info.org/tiki-index.php?page=Asterisk+Fax+to+email anyway, when i try to send an fax, i get theese messages in asterisk: -- Executing Goto("SIP/5060-08148520",

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:

I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for "buzzing" on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and