search for: teleco

...s 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 different teleco's message is different, and sometimes our faked message cause the customers confused. So we wonder whether there is anyway to hear the teleco side message directly.

...,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 card TE110P get crazy? is the TELECO are crazy? any idea

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 external. This is the setup: Teleco <-> Kamailio <-> Asterisk SIP --> 1.2.3.4 10.0.0.1 --> 10.0.0.2...

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110811/25a1b3c7/attachmen...

...ndpstn group=1 channel => 1-15 callgroup=1 pickupgroup=1 Whenever I try to connect this up to the ISDN line I get a series of Red Alerts and any attempt at outgoing calls results in a no channels available message (essentially all the lines are shown in use and cannot be cleared). I have had my teleco reset the line which just results in further red alerts. NTL, bless them, came out with a test rig and plugged this in the back of my * box and we made a series of test calls which all worked fine, although the NTL chap said the attenuation was out as there was a lot of buzz on the line. He suggest...

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/

...\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 CVS-05/22/03-11:14:50 built by root@teleco on a i686 running Linux Does somebody knows the problem? It's a version problem of *??? I would be very pleasured if somebody can help me or told me if I'm mistaked about this functionality of MeetMe. Thanks a lot.

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 we restart * with restart now cmd. Which taks approx 10-20 seconds to co...

...example using a mechanism such as MLPPP LFI.(RFC1990) >> 2. The packet size threshold before fragmenting AF and BE packets MUST be configurable. >> Thank you very much! > >What I mean is that if you want to mess around with packets below ip >level, the other end - your ISP/teleco , will need to be running >software that knows what you are doing so that it can reconstruct the >packets before routing. > >If you have a specific need for your upstream not to be delayed more >than X ms you could adjust your MTUs/MSS clamp - the size will depend on >your bitrat...

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 is a X100P card and this is my extensions.conf for incoming PSTN calls: ; Parameters for calls from PSTN PSTN_RNG_EXTEN=SIP/251&SIP/261&SI...

...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 i...

...es. No Problem I thought, I'm well on the way to this anyhow. So I'm thinking (although not tried it) that if I got my Asterisk box running for my company (E1 card for outside link) I would AIX the hosted PBX for the customers to my PBX to allow them to make outgoing calls. I would get my teleco to provide phone numbers for them and also get my PBX to redirect that number to the hosted PBX. Is this correct so far? Or should I keep their system separate on another E1? Or should I forget my PBX and push their incoming / outing calls out to a SIP / AIX provider on the net and wash my hands o...

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

...balance Asterisk server,when it is a SIP client. I am little confused on load balancing, when asterisk server is also a sip client. Based on these, XO Communications one of the largest US DID Provider, now offer SIP Orignation Services for wholesale. Verizon Communications One of the largest US Teleco, now offer SIP Orignation Services. That means no need for PRI card. So if I take service from them, then my asterisk server will be SIP client. Right? How can I set up my asterisk servers so that the calls originated by XO/Verizon goes to different asterisk servers based on load. Has any one doe...

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