similar to: Ast<->Ast: IAX2 error w/no audio

Hello! I try to maintain a couple off old Openwrt based routers. The routers run openwrt whiterussian, so it quite outdated. But it is a little dangerous to update a remote machine from whiterussian to kamikaze or backfire, this may be usefull for others. I used the openwrt whiterussian 0.9 sdk and the attached Makefile to compile all tinc versions till 1.0.11. Since 1.0.11 the compile process

Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other

Asterisk 1.2.16 I have an extension "102" with a Polycom 430 I am trying to protect against forwarding loops If I set the phone to forward the line to itself, extension 102 I get the following -- Got SIP response 302 "Moved Temporarily" back from 206.83.240.18 -- Now forwarding Local/102@mycontext-b2ee,2 to 'Local/102@mycontext' (thanks to

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

i apologize if this has been asked before, but things are too busy to preclude a full search of the list. i have both cable and dsl from the local providers here. due to wiring issues here, i''ve been forced to put the cable modem in one end of the house and the dsl modem in the other. the cable modem is firewalled off by a cisco pix 501 (192.168.2.12). the dsl modem is firewalled off

Hello, I have a very strange problem that I can''t seem to be able to figure out. First of all: my network scheme: http://heavyg.safehex.be/network.png I have an agreement with my neighbour that I can use his wireless network to experiment with extra internet uplinks in exchange for some filesharing etc. Now I have set up a wireless router at my place, that connects to his network.

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

Using CVS from 7/12/04 and trying to get the voicemail broadcast feature to work. Voicemail.conf has [mycontext] 3722 => 1234,BroadCast Test,,,cc=*@mycontext . then many other voicemail boxes. ----- whenever I leave voicemail at box 3722, only box 3722 gets the voicemail. It is not expanding it to other voicemail boxes in the [mycontext] context. Even if I replace the cc= line with

On machine A I have something like the following in extensions.conf: [iax-extensions] exten => _9.,1,Dial(IAX2/machineB/${EXTEN:1}@mycontext) exten => _9.,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => _9.,3,Hangup On machineB I have something like this: [mycontext] exten => 2002,1,Dial(SIP/2002,60) exten => 2002,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => 2002,3,Hangup If I use a

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

Hi folks Is it possible to turn off the native bridging on Asterisk? I've been hacking about app_disa.c to support account & pin numbers, that tag the calls depending on who logs in..... It all works fine, then dials the destination number they requested. My setup is as follows [ENDPOINT] <IAX1> [MYASTERISKBOX] < IAX1 > [TELCOBOX]<>(PSTN) If i dial

I have asterisk running in a wrt54gs attached is a pap2 with 2 extensions working on it, the problem now is that there is lots of echo, some rythm in the background, and the voice is delayed by about 4 or 5 sec's between the 2 extensions. memory usage is about 15 to 20 megs so I think I can solve the problem with correct settings, anyone know where I might start to correct these issues

Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND

Hello all, I would like to set a different caller-id for the second leg of a call when doing an originate. For example: Action: Originate Channel: sip/1234 Context: mycontext Exten: 1 Priority: 1 Callerid: "123 <123>" Async: true This sets the caller-id correctly when dialing sip/1234, but I would like to set the caller-id for the second leg of the call (the one that goes to 1 at

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

My SIP-provider sends my a SIP-invite like this : INVITE sip:329298yyy6 at 80.XX.XX.69:5060 SIP/2.0 Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c Max-Forwards: 70 From: <sip:321445xxx6 at 80.XX.XX.69>;tag=f395877e02bf8eb2fd8f5a0e To: <sip:329298yyy6 at 80.XX.XX.69> Call-ID: f395877e02187250fd8f5a0f at 80.XX.XX.68 CSeq: 1 INVITE User-Agent: SysMaster VoIP

Hello folks: This may have been discussed many times before but I have not been able to find it. I have also not been able to resolve it myself so I am asking here with hope that someone can straighten me out. I am using ip to do multisource policy routing or two connections to the internet. I have a linksys wrt54gs route which connects two machines by wire through the switch and three

Hi, I am attempting to set up some simple outbound shaping following the LARTC HOWTO. The HTB qdisc seems to work as the documentation says, but my filters don''t seem to be working. All of the packets go to the default queue regardless of what filters I set, it seems. (according to tc -s qdisc show) I am trying to get this working on my openwrt box (whiterussian rc6), but when

I just set up a WRT54GS and now I can't dial out or in. sip show registry shows: CODE: SELECT ALL Host dnsmgr Username Refresh State Reg.Time draytel.org:5060 N xxxxx 120 Request Sent I seemed to recall that running in cli always showed a response back, but there's nothing now. Using

Sorry to resurrect an old thread, but I finally got around to testing this approach (round tripping through bitcode in memory) and it works beautifully - and isn't that much slower than cloning. I have noticed however that the copy process isn't thread-safe. The problem is that in Function, there is lazy initialization code for arguments: void CheckLazyArguments() const { if