similar to: Outgoing calls but no incoming calls with X100P

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

Hello list, New to asterisk and to the list (although experienced in Unix/Linux administration). Short problem description: -------------------------- I cannot get the Echo() application to run on any 32bit platform I can get my hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have runs just fine. In all cases asterisk log shows the same -- that Echo() is executed Details:

Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the "s" extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general]

I'm curious as to what versions of everything you are using. Reason being this line " -- DAHDI/i1/00256312261627-1 is proceeding passing it to SIP/5000-00000000". It states "DAHDI/i1/00256312261627-1..." and I don't recall seeing that before (my 2.4.0 says " -- DAHDI/1-1 is proceeding passing it to SIP/801-0000000c" [1-1 being the span and channel

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten =>

I don't use digitalvoice, but based on a similar provider you may need to have your username inserted in your extensions.conf context.... [incoming_calls] exten => username,1,Answer( ) exten => username,2,Playback(demo-echotest) exten => username,3,Hangup( ) Just an idea....

Hi. I want to handle incoming chan_misdn traffic by asterisk, but I've got message - 'Extension can never match, so disconnecting'. What I'm doing wrong ? How I can pass incoming dialed number (dad) to misdn context (in my case 'dss1_incoming') ? Works unrouted calls (s extension) if I set immediate=yes in misdn.conf, but I want to route calls by dialed number. log

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

Hello list, i have installed elastix 2.4.0 with call center model and i have created an Outgoing Calls <https://192.168.1.251/index.php?menu=outgoing_calls> my question i want to know the name of the tbale where the csv file is uploaded in order to do some works. NB: i found the cdr table in asteriskcdrdb database but the is no information related to my csv file any help please

I have a TMD400 card installed in a PC with one fxs (installed in slot 2) and two fxos (installed in slots 3 & 4).? fxos work fine but I am unable to get a dial tone for any devices connected to the fxs.? I have correctly connected the power supply to the card and I have even tried moving the card from slot 1 to 2 on the board. Below is from the console when I try to route a call from FXO on

Relax your PRI is fine. What Xo is sending you is 4 digits os DID. If for example you have 1130-1153 as the last 4 digits of your Number you can use this to rout your calls. exten => 1130,1,Goto(ivr,s,1) Exten => 1140,1,Goto(extensionss,100,1) Exten => _X.,1,Zapateller Is the above config. XXX-XXX-1130 with go to the s extensions' first priority in the ivr context. XXX-XXX-1140

Hello again dear list. Could you please help with this? Thank you for all support, you are great, and i am now at a late stage in the setup and tweaking this server, So I hope you can help me again. I Can't make include the context nighttime. Just to demonstrate if it works, I have a playback function there. But CLI reports: CLI [Jun 18 14:20:22] WARNING[2287]: pbx.c:9542

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

Hi, Im getting mad with this error, I have a x100p installed with wcfxo module loaded perfectly, I can receive incoming calls and detect very good the hangup for incoming calls. But for outgoing calls its a mess. When I place a call for outgoing, i heard the ringing, my cell or phone rings and when I pick up the phone it hangs: -- Called g1/91xxxxxxx -- Hungup

Hi folks I have a X100P in my home asterisk box and I have it setup as a default context of 'incoming-pstn' in my extensions.conf i have [incoming-pstn] exten => s,1,Goto(incoming,01225<myofficenumber>,1) then: [incoming] exten => 01225<myofficenumber>,1,Answer exten => 01225<myofficenumber>,2,Dial(SIP/data|m) etc etc Anywho back to the plot. It all works

Hi All, I wasn't too sure how to word the subject, so I apologise for that. Anyway, I've got two X100P cards here accepting calls. Basically in Australia our ring cadence is 400ms on, 200ms off, 400ms on, 2000ms off, repeat. What I've noticed is that it takes about 8 seconds after the caller has hungup for the internal phones to stop ringing, whereas in Australia the maximum time

Hi Gurus How do I make x100P does not answer incoming calls ? I want * play dead for incoming calls. I do not have any context for incoming calls from x100p, in zapata.conf. Call also get logged into the CDR, that too I do not want. I am using x100p for outgoing calls only. Any help appreciate. Cheers SW