similar to: instant fallback to zap in case of sip/iax/xyz-failure

Rafa³ Dutko has just discovered a potentially serious bug in version 1.3.0 and 1.3.1. In both versions, where an interface option appears on multiple interfaces, the option may only be applied to the first interface on which it appears. A corrected firewall script for 1.3.1 is available at: http://www.shorewall.net/pub/shorewall/errata/1.3.1/firewall and

I have a dial cmd buried amongst a series of others in a macro like so: exten => s,n,Dial(SIP/1${ARG1}@sip_peer,60,T) Reason for adding a "1" is all the others in the macro don't want the "1" so this was easiest at the time. Now I need to send NA long distance through this macro. All the other dial cmds will just work, but this one is going to try to dial 11NXXNXXXXXX

Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret: $strSecret\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket,

I'm trying to to to "dial1" if caller id match: but dial plan execute 220,n(dial1) regardless exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?dial1) exten => 220,n(dial1),Dial(${sales_support}&${accounting}&${family},25,m(penguin)w) exten => 220,n, I was under impression that if condition is met it will jump to

I'm learning asterisk, and decided to make myself an answering machine out of it. Seems pretty straightforward to use an agi (perl) to do what I want. What I want is: Answer the phone. check for time of the day If TOD is during the time I sleep I announce i'm sleeping & prompt caller to dial1 (or whatever) to connect to my extension & then go to voicemail if busy/una,

Hello ppl, I want to make a special dial plan for routing calls to a peer which has an pin protection. Normally if you want to call through that peer you must first enter pin for example 1234# and after that you hear the tone from line and after that you can dial desired numbers. I tried something like that, but doesn't worked. Did somebody have some clues? exten =>

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial

Hi everybody. We have a samba4 domain deployed across serveral countries. Some of them (overseas) have a poor VPN connection with mainland. Since Samba4 does not support (yet) subtrees, we have deployed a DC in each location for domain validation. However users in mainland logon randomly at overseas location and sometimes this is a problem due to low bandwidth available. Is there any way to

hello, my webrtc calls ends after ~60seconds with "res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert unexpected message', terminating". any ideas where can be problem? or howto debug this problem? asterisk13.4.0-rc1 + sipml5 latest (chrome,firefox) -- --------------------------------------- Marek Cervenka

I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them. Here is what I have in my sip.conf: [stanaphone] type=friend secret=pAsSwOrD ; skewed for this message. username=3475341914 host=sip.stanaphone.com

Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip

I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble.... thanks Todd

For some reason I now seem unable to control which codec is chosen. The problem happens with outgoing calls to Stanaphone. Even if I chose disallow=all and allow=ulaw as the only codecs it connects with GSM. Has anyone else got problems with these settings? Any suggestions? As I recalled it, such a setup would not establish a call if the ulaw-codec was not offered by the provider. Stanaphone has

I have an odd problem. I got controllers in a namespace and controllers outside of the namespace. For example, I have a PagesController and a Admin::PagesController. When I run rspec from the top, tests pass and I get the following warning: spec/controllers/admin/pages_controller_spec.rb:4: warning: toplevel constant PagesController referenced by Admin::PagesController This makes no sense. I do

Hello list. When using SIP DNS SRV to define a production Asterisk server with high priority and a backup Asterisk server with a lower priority on this DNS-server, will this work as follow : - production server is reachable, so registration of the IP-phone goes to this server - production server is unreachable, so registration goes to the backup Asterisk server - production server is

Hi, I am pretty new to all of this but was able to set up an asterisk server and have been able to succesfully connect to asterisk with x-lite as sip client. I have also connected asterisk to FWD (using iax2) and to voipjet (also using iax2). Now I am trying to connect asterisk to Stanaphone. It has to register as a SIP client but I am not being succesful at all. My asterisk server sits behind a

Anyone having problems with Stanaphone registration today? I'm getting the following.. Nov 25 11:35:58 NOTICE[229390]: chan_sip.c:4053 sip_reg_timeout: Registration for 'xxxxxxxxxx@216.128.82.18' timed out, trying again -- Got SIP response 500 "Internal Server Error" back from 216.128.82.18

I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is

We are having lots of problems with stanaphone. It used to work ok, but now it's terrible. As of this moment, can't make outbound or inbound calls. Anyone has it working? Please provide sip.conf example commands.. Thank you -- Leandro Rzezak leandror@gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am attempting to do selective fixture loading against my test database. When I run rake db:fixtures:load FIXTURES=DataSetInfo I get no output. (A full trace is below). I''m trying to figure out why nothing happens. 1) Does the command above attempt to operate against my test database? 2) DataSetInfo is the name of my table and the name of my yml file I generated these YAML