similar to: Cannot hear teleco side error message

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 are using asterisk 1.2.13. When callbacklogin agent transfer call using SIP phone's transfer feature, he is always in busy status and cannot answer any more incoming call from queue until the transferee hang up the call. -- Regards! Liangliang

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

Hello, I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. My plan is to use this handler to update my CDRs with values such as Asterish and Tech cause (see function HANGUP_CAUSE). I want to have my custom hangup-handler be run only once and when "the second channel" hangs up. At the moment, I'm issuing a couple of "channel originate Local/1 at mycontext1

I have a asterisk server. Two SIP Soft XLites are connected to the server. I am able to make calls from one SIP Phones to the other SIP Phones and landlines successfully. The SIP Soft Phone on th eother side can hear my voice but I cannot hear their voice. They can call my local cell phone as well. Samething, they can hears my voice, I cannot hear their voice. The microphone and speakers are

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

Antony, It is correct. Noone connects to Asterisk box/server from outside.Callcentric SIP trunk configured and Asterisk maintains connection to it itself. No special ports opened, nothing. Connection happens from us to Callcentric and all calls routed in from CallcentricI don't know exactly how it's doing it by it works. Again, keep in mind it is working for many years for most / 90+% of

Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? Regards, Liangliang

Hi, all, Do you guys happen to know that there are any IP phones have such feature, that it can has some indication for the agent status linked to the phone? E.g some LED show the status, backend we can link the phone to one agent id, then the agent login the system, the 'online' indication will be blinking and on, if logout with type of meeting, then 'meeting' LED will be on,

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+

Hello, I am using voipuser.org service, and am trying to make a SIP call. Everything seems to work fine, except I can't hear anything on my end. When I make a SIP call, the other party can hear me, but I can't hear anything. I am using asterisk + Digium TDM board with an FXO port where I connect a regular telephone. Can anyone assist? I believe I have some asterisk

Hello, This is not technical post, just looking for suggestions on what to check.I have asterisk for long time, no updates, just maintain OS updates. I use SPA504G phones Very rarely and randomly when we pickup a phone - other side does not hear us. Call them back and all works. Now I have couple people I'm talking to and it seems like very call like this. Someone can't hear someone.

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

My MusicOnHold sound is very soft, but when I hear it directly from mp3 playe on desktop, the loudness is quite ok. Wonder whether there is any configuration to change the loudness of MusicOnHold. Regards, Liangliang

Hi, I'm looking for some help with a problem in Asterisk (possibly), and I'm confused as heck what is going on. I've updated to the latest Asterisk version and the problem is still occur. My setup is as follows: I've got Asterisk running on a high end Pentium-IV box running Linux serving 5 calls, it is located in Canada. The calls come in via analog lines through TDM400P cards to

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

2013/3/26 Richard Mudgett <rmudgett at digium.com> > > On 03/25/2013 05:17 PM, Olivier wrote: > > > Hello, > > > > > > I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. > > > My plan is to use this handler to update my CDRs with values such > > > as > > > Asterish and Tech cause (see function HANGUP_CAUSE). >

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

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:

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