search for: telecos

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

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

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

Hi everybody, i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of TE110P and also if you can tell me have to made a cable like that?? Modem Teleco <-----------Self Crosscable------------>Asterisk Rx+ <--------------------------------------------------------------> Tx+ Rx- <--------------------------------------------------------------> Tx- Tx+

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:

Advance apologies for the length of this mail; I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to replace this phone system shortly and I've been trying to get a * system working for some weeks now. I have configured the dial plan (which works) and all my SIP extensions (which all work) along with

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/

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

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

...ps from the longest slot to make space for an incoming packet, so it''s not tail drop as such, but the results show me it does drop from the tail of the slot - which if you are trying to shape inbound, is a PITA as tcp "slow" start grows exponentially and overflows into my ISP/telecos buffer, causing a latency bump. I think it would be alot nicer if It head dropped to make the sender go into congestion control quicker. However this is not the reason for this post. I tested by capturing with tcpdump before and after the queue. I noticed that the packets were being released in...

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

Anyone know if it''s possible to interleave two IP packets when using PPPoA and VC based lines? Can it be done with any PPPoE implementations? The goal is to reduce the delay when you have a high priority packet waiting, but a lower priority (large) packet already started going out ahead of this packet. I don''t want the overhead of much smaller MTU, which is the other way

El 3/9/19 a les 18:15, Laine Stump ha escrit: > On 9/2/19 10:31 AM, Francesc Guasch wrote: >> Hi. First of all thank you for the work you are doing with libvirt. >> I am not sure this is the right place to ask, I'd appreciate >> if you can give me any hint or directions. >> >> I have several similar KVM Linux boxes and one of them has a really >> strange

I thought XO was reselling Level 3s (old Genuity assets) network/voip just like Qwest ? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of jk@bingoconsulting.com Sent: Wednesday, October 11, 2006 3:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Load balance Asterisk server,when it is a SIP

Chris you might know the answer to my HUUUUUUGE problem A few weeks ago you posted this message: "I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to replace this phone system shortly and I've been trying to get a * system working for some weeks now. I have configured the dial plan (which works)

I have read the thread at http://mailman.ds9a.nl/pipermail/lartc/2006q1/018287.html and still don''t know how to fix this problem. It appears alot of work has gone into it but the HOWTO is so out of date it doesn''t even begin to addresses this method. So here are my questions 1. what is the current state of these patches? are they in a specific version? do i have to patch myself?

Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com