similar to: gsm errors

Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan

Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

Hello there everyone, Well I have set up Asteriks 6.0 and almost have Freepbx working too. However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is not found. I confirmed that by going to the directory. How do I get /var/run/asterisk/asterisk.ctl put in correctly? I am using a Ubuntu 8.10 system. Thanks much.

Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc:

Hello List, iam following to this howto: http://www.sysresccd.org/Sysresccd-manual-en_How_to_install_SystemRescueCd_on_an_USB-stick but when iam try the command: # syslinux /dev/sda1 it writes only into the FAT partition the files they need, but mbr still is empty after. iam do this: # dd if=/dev/sda of=/tmp/mbr.dd bs=512 count=1 and open it with ghex2 in my case and see only the partition table.

Hi iam using 1.2.17 does any one have information meetme in realtime and store in mysql i dont see any document could some one help me is this possible ? ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070606/36d236c2/attachment.htm

Hi iam not able to start asterisk give me following error any help STARTING ASTERISK /usr/sbin/safe_asterisk: line 42: 4633 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} Asterisk ended with exit status 132 Asterisk exited on signal 4. Automatically restarting Asterisk. /usr/sbin/safe_asterisk: line 42: 4637 Illegal

Hi i have 20 people calling agents calling when ever they calling i get this below error May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided deadlock for '0x8b2f50', 10 retries! and the voice go choppy, and voice breakages iam using Latest SVN, any suggestion to come over this problem ram -------------- next part -------------- An HTML attachment was scrubbed...

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

Hi all, we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances behind FW. Problem we face is that when we reboot the DC Asterisks, the trunks (SIP or IAX) become alive from DC Asterisks to clients ones but UNAVAILABLE the other way. In clients logs we see Registration for 'XXX at

Hi, Iam using cento-5.3 on x86 machine and iam working to get the console messages redirected to serial port. I have made the necessary changes in the /etc/inittab file. - - - - - id:3:initdefault . . so:12345:respawn:/sbin/agetty ttyS0 9600 vt100 - - - -- - and also in /etc/grub.conf file - - - - serial --speed=9600 --unit=0 --word=8 --parity=no --stop=1 terminal --timeout=10 serial . .

Hi iam keep getting this log in my asterisk log is this harm anything, and how can stop this, any suggestions Jun 4 18:21:47 DEBUG[2093] chan_sip.c: Stopping retransmission on '45629314783bd11604363618632f07b9@201.x.x.x' of Request 102: Match Found Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received

hello list , my set up is like this ip device -->ser ---> asterisk(astcc) --> pstn gatewsy my asterisk version is 1.0.2 iam using the ser as registration and asterisk aa the prepaid one with the help of the astcc. now my problem is the destination people i.e the pstn line s are listening low voice and also the blurr sound quality along with the audio of the ip device at

Hi everybody, I'm trying to develop an R package with Fortran and OpenMP. I wrote a simple hello world but I'm not able to set the number of threads. I found this old email chain <http://r.789695.n4.nabble.com/Set-the-number-of-threads-using-openmp-with-C-td2284685.html> and I tried to set my compile instructions accordingly but i had no luck. *This is my makevars:*

After rebooting my asteriks server, e-mail notifications are no longer being sent after a voice-mail is left. I can see the messages in /var/spool/asterisk/vm. has anybody had the same experience? how was it resolved? Uri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/fc494bb4/attachment.htm

Hi all Its been long time i have sending this message to this news group but i could not get proper responce from the authors or any experts that, is that bug or only the problem with my ssytem iam tryin this config with FC iptable 1.3.0 and latest TC tc utility, iproute2-ss050330 htb 3.17 iam trying past 2 weeks and digging all googles but could not able to come to conclusion whats wrong

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

Hi list, perhaps someone can help me out? fact: - samba 3.6.9 plus CentOS 6.5 - i have user quotas set up on their HOME directories, which resides in the "/home" filesystem - on windows workstation their disk quota is not shown, instead they see whole filesystem free/taken space (which generate much grumbling) After searching lists, googling etc., i decided to give a try the

Hi everyone! First of all: Big thanks to Tom for this great work!! Now on to my qestion: I am using Shorewall among other machines on an new server where we need some kind of accounting. The script we would like to use for this is iam (http://intevation.de/iam/). The docs say: Alternatively you can use your own iptables script and only add the ''dump'' option, which should