similar to: Question about TFTPD server

does anyone get a hold of the SPA-941 Provisioning Guide? i tried call Sipura's tech support, seems like none of them heard of the term "remote provisioning". they kept refering me to their web site which i've check thoroughly, and could not find any documentations on the SPA-941. finally they gave me a phone number to call, which appears to be a fax machine. that's when i

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

hi. i connected Asterisk to an NEC NEAX system with a crossover T1 cable and the Digium TE405P using E&M wink signaling. the connection's ok. however when dialing from the NEC to the Asterisk. most of the time the Asterisk only sees the first digit of the dialed number(which is 4 digits). some time if i dialed the 4 digits very fast it might get through. seems like there's a timming

Hi all, I'm trying to setup a TFTP server to serve as repository for the config of all my Cisco network devices. As per the the tftpd man, I've added the "-c" option into the /etc/ xinetd.d/tftp (as follows) but I still cannot get write access (unless the file is already present). [root at chl1 ~]# cd /etc/xinetd.d [root at chl1 xinetd.d]# cat tftp # default: off #

Greetings to all. I am hoping someone can help me out with a problem I am having getting my Cisco phones, 7960s and 7940s, to download the appropriate files from our TFTP server. The TFTP server is running on Fedora Core 4. The TFTP server appears to be setup properly: service tftp { socket_type = dgram protocol = udp wait =

hi folks. i have the following puzzle: when i call certain cell phone# using a regular phone & POTS. the called cell phone co. usually return a message such as phone travel out of range or phone is busy etc. if the phone is unreachable. now when i have the following setup: sip phone -> asterisk -> PRI -> phone co. i call the same cell# and if it's unavailable. the PRI return

hi folks. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]:

i recently bought an SP4000 conference phone but having problem provisioning it using ftp, every time it just hangs at "Updating initial configuration..." screen. when i switch it to tftp, it'll work fine. i though it was bootrom/firmware issue so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the

hi folks. we've experienced some weird problems lately. we have about 600 SIP phone on a single system running *1.4.26.2 for about a month. recently there was massive UNREACHABLE messages like this one showed up: chan_sip.c: Peer '2699' is now UNREACHABLE! Last qualify: 1252 then they all became reachable again in a few seconds. sometimes it last for couple minutes. but sometimes

Hi All; Any one can suggest a TFTP server to be installed in Fedora (same machine that Asterisk is installed) to be used for Cisco IP Phones to download the required firmware and configuration files. Thanks for the help in advance. Regards Bilal

hello, i'm having a wierd problem with 0.31 tftpd-hpa. i'm using xinetd, with this config: service tftp { disable = no socket_type = dgram wait = yes user = root log_on_failure += USERID bind = 10.13.0.254 server = /usr/sbin/in.tftpd

hi folks. i'm experimenting with iaxmodem + hylafax using DID to determine where to send the fax to it's final destination. however i have difficulties passing the DID information from iaxmodem to hylafax. in extensions.conf: exten => _XXXX,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r) exten => _XXXX,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r) exten => _XXXX,n,Busy exten => _XXXX,n,Hangup

hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305]

Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:.

I'm running Asterisk 1.4.18 and having a problem with the clearglobalvars option. I have a NIGHT_SERVICE variable which I initially set equal to off. I then have an extension they can dial which will toggle that variable. My problem is when you enter the CLI and type "reload", it resets to "off" again. I've tried setting the clearglobalvars=no as well as just

Greetings all- During some recent testing and debugging, I wanted to use the 'fxstest' application. However, I found it hasn't been built when doing the standard 'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0... Can anyone tell me how to build fxstest? Thanks! --Tim

In the following example exten => _1NXXNXXXXXX,1,Set(GROUP(outbound)=myprovider) exten => _1NXXNXXXXXX,n,Set(COUNT=${GROUP_COUNT(myprovider at outbound)}) exten => _1NXXNXXXXXX,n,NoOp(There are ${COUNT} calls for myprovider) exten => _1NXXNXXXXXX,n,GotoIf($[ ${COUNT} > 2 ]?denied : continue) exten => _1NXXNXXXXXX,n(denied),NoOp(There are too many calls up) exten =>

Has anyone had any experience using a modem through the Asterisk system? I have some technical support personnel that need to use a computer modem to connect to a remote system for troubleshooting. Is there a SIP compliant gateway that will support a modem connection at decent speeds (minimum of 28.8) that anyone knows of? If not, has anyone used a Digium FXS card for this? Thanks

Froogle found me one supplier for the SPA-841, not sure I trust them though. Does this phone even exist yet? Does anyone have any experience with it? Does anyone know a vendor other than Atacomm/voipsupply?

I am using a context to change values in a DB. Currently in my context, I am passing it to exten => s,1,WaitExten(7) ; 7 seconds to input exten => s,n,Set(NEW_VAR=${EXTEN}) ;Here is my problem. This is the only way I know how to 'grab' user input, which was normally from ${EXTEN} but I realize this won't work for extension 's'...... The short google search I did