similar to: Rx+,Rx-,Tx+,Tx- of TE110P

Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first port on server B (te_ptp) but the port light cotinues blinking on red on both sides once the

please can someone send to me his files like zaptel & zapta if he si using TE110P thank you

Has anyone tried the new ariavoice $89 VOIP desk phone with Asterisk? ` http://www.voip-info.org/wiki-AriaVoice -- Jim James H. Thompson jht@lj.net

hi eveybody, after many test with your help and the irc channels help, i get the led on TE110P green with this config: span=1,1,0,ccs,ami =====> alarms OK Green Led but the provider say that i have to set my span to this span=1,1,0,ccs,hdb3,crc4 =====> alarms: YEL/RED i can't make call's yet to test because they have to sync the Modulator in the other side so any remark? is my

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the

Copying a file (650Mb) from a W2K pc to Samba doesn't work well. I have a transfer time of 25 minutes while the 2 machine are connected with a crosscable to each other (10Mb ethernet). A ftp between both systems went well. What's the problem...... -- Regards, Roger -- Running Redhat 7

I'm having loads of problems with recordings, as in crappy audio quality and lost pieces of the recordings. I've been searching for a solution and the solutions i find on the interwebs include a ramdisk, for local recording, or another machine, handling the recording. I guess the ramdisk would be the "easy" solution and the external machine would be little harder to set up. I do

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

Hi All, I want packets [request/response] capture for ISUP packets , i have E1 line terminated to my digium card i just want a packets flow between my machine and teleco side, is any tool or utility [command] availabele for observation this packets and data. any help appericiated Thanks Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good afternoon, I'm trying to use MeetMe in an AGI script written in Perl, as follows: print "EXEC MeetMe 2000|p \n"; $res = checkresult(); The problem that I have is that when a user press '#' in order to exit from the conference, everybody goes out. This is randomized because sometimes doesn't happened. My current version of asterisk is: Asterisk

I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (just one!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time

need help trying to figure out why calls hang when using multple ports on Sangoma card. we have 1 quad card with 3 T1 ports configured, Port1 acts as connection to teleco (to our T1 PRI) port 2 connects second system and routes calls to port1 port 3 is Asterisk pbx calls all go in and out properly but sometimes we get a call hang on when both sides hangup. this causes all calls to fail until

Hi,Andy Thank your very much! For the MLPPP LFI,I found that in Cisco configuration,it use "ppp multilink;ppp multilink fragmentation;ppp multilink fragment-delay 20;ppp multilink interleave " command to enable MLPPP LFI.So I think just realizing the same function on my linux router would be fine.But I got no idea how to do this on linux.So is there anything with iproute2?Would you

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware

I want to provide internet to home users with 256 Kbps and I have a 3 Mbps dedicated internet connection. Do you think It''s ok to split the 3 Mbps in 480 users? Thanks, -- Nicolas _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

Dear community, I have a 2-node gluster cluster with one replicated volume shared to a client via NFS. If the replication link (Ethernet crossover cable) between the Gluster nodes breaks, I discovered that my whole storage is not available anymore. I am using Pacemaker/corosync with two virtual IPs (service IPs exposed to the clients), so each node has its corresponding virtual IP, and

I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after

I posted this to asterisk biz but didn't get a reply.. I didn't want to offend anyone being that this is kind of branching into hosting, and maybe outside of the remit of this list. Hi Been lurking on the user list for a while but I have some what of an immediate requirement and I'm wondering if you can suggest the best solution (if mines a rubbish idea) I have been testing Asterisk

Hello! Hardver: Windows XP Compaq Proliant DL360, Linux Compaq Proliant DL380. 2 pieces of processors Intel Xeon 3,2GHz, 2GB RAM, 6 gigabit interface (2 tg3, 4 e1000), Debian Woody, 2.4.31 vanilla kernel. 6 U320 SCSI 15krpm HDD, 2 HDD RAID1 system, 4 HDD RAID1+0 data. Every network cards connect at speed of 1000MB full duplex, with XP crosscable (but we've tried with gigabit switch, too, we

When we use mobile to call certain number, we can hear the message like, " The number you dialed is incorrect", "The customer is currently is unavailible" and etc. But when we use asterisk to call the same number, just busy tone. We found that since version 1.0 it support the standard hangup cause, so we base on the HANGUP_CAUSE to fake the error message, but seems