similar to: Call bridged, but no sound

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to

Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Added the appropriate entries in sip.con and on the PAP2. I then tried to call from one line to the

I'm sure I'm doing something wrong here, but I would appreciate a pointer. I have tried to rtfm, but find the docs a little sparse wrt pxechain.comand gpxelinux.0. I am trying to setup a pxe chain server (aka chainloading?) where one of the entries on one pxe server forwards to another (cobbler, in this case). I know this works fine with traditional pxelinux.0 images, but it's

Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the

Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic "chan_sip.c set_destination: can't find address for host" I have set the "canreinvite=no" in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

Hi, I have two physical machines connected through a switch in a 192.168.1.* subnet. Dom0''s are assigned static ip addresses 192.168.1.100, 192.168.1.101 respectively. Dom0''s can ping each other. Each physical machine has a domu guest and static address assigned to domu''s are 192.168.1.200, 192.168.1.201 respectively. However domu''s can neither ping each other

On Wed, Aug 7, 2013 at 3:58 PM, Hans Lellelid <hans at velum.net> wrote: > I'm sure I'm doing something wrong here, but I would appreciate a pointer. > I have tried to rtfm, but find the docs a little sparse wrt > pxechain.comand gpxelinux.0. > > I am trying to setup a pxe chain server (aka chainloading?) where one of > the entries on one pxe server forwards to

IVR Number :17275691533 When I try it from xlite configuring my provider directly, it works perfectly. When I try to dial out from dialer , it doesnt work. [sip8] type=peer username=user fromuser=user authuser=user secret=password host=8.14.146.111 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 What cld be the reason ? --------------

hi i got iconnect to work, works pretty well now except calls sometimes (more often than not) hang up after a couple of minutes.. heres a bit of the debuging Record-Route: <sip:61892142222@213.137.73.178:5060;maddr=213.137.73.176> From: sip:61892142222@natrelay.deltathree.com;tag=3281050172-73809 To: "JUSTIN XLITE" <sip:2001@61.95.68.84>;tag=as09766a78 Call-ID:

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

Hello, I notice that Asterisk always sends Subscription-State: active, even when the SIP-peer is offline (IP-phone cut from power) : /[Jul 31 11:56:58] NOTICE[32273]: chan_sip.c:26194 sip_poke_noanswer: Peer 'testacc77000' is now UNREACHABLE! Last qualify: 49// //[Jul 31 11:56:58] Really destroying SIP dialog '78b0d1701d3694b1494a0c4b55344d57 at ip-sip-server:5060' Method:

Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

When someone calls in on a zap channel with FXO and presses an extension, and another user picks up using (*8) I changed it to 888, after a few minutes ( I think 2), the call gets dissconected. The users all use Cisco 7960. I didn't yet have a chance to test it when not using Call Pickup (*8)888. Please help. Here is the screen shot in asterisk: +++++++++++++++++++++++++++++++++++++++

Hi We've been running asterisk 1.4.17 (deb package) in a production environment for some while now and are finally taken the plunge to update to 1.4.38 (Ubuntu servers). All of this is using the RealTime Architecture I have upgraded the asterisk version in one of our test environments and blind transferring seems to have suddenly stopped working. It was working fine under 1.4.17 So, call

Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So,

Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}

Greetings, I'm trying to continue to do some processing after a TIMEOUT (absolute). In my dialplan below, when a call comes in to [default], I call macro-phonenum and pass it a timeout of 20 seconds. macro- phonenum sets TIMEOUT(absolute), then loops saying the phone number that was called (in MACRO_EXTEN). When the timeout expires I want to call my macro-hangup (so it can say