similar to: newbie with kphone and asterisk

hello all, now, i want to do configuration to make sip client have extension on my asterisk.but i have a problem with registration of sip client. *CLI> May 31 13:58:01 WARNING[4927]: chan_sip.c:886 retrans_pkt: Maximum retries exceeded on call 4b5cbb235a46d6ee0bcd278c1e294105@192.168.8.125 for seqno 115 (Critical Request) May 31 13:58:15 NOTICE[4927]: chan_sip.c:4585 sip_reg_timeout: --

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hello People, I was wondering if you could take a look at this script, exten => 505,1,dial(iax2/6311${EXTEN},t,25) exten => 505,2,playback(pls-wait-connect-call) exten => 505,3,set(NewCaller=${CALLERID(num)}) exten => 505,4,Set(CALLERID(num)=0${CALLERID(num)}) exten => 505,5,dial(Zap/g1/c/0296389675,20,r) exten => 505,6,Set(CALLERID(num)=${NewCaller}) exten

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>

Greetings, I have implemented the following command to allow CNAM to be delivered to my users. exten => 9969,1,Wait(1) This works great! However it has spawned a new problem. When this is implemented into a full dial plan. If a user is set to DND or sends a call to Voicemail by hitting deny the caller gets a busy. Below is a result of the calls. With the Wait(1) statement -- Executing

hello, I am running asterisk on one linux PC and want to talk through this server using Kphone installed on 2 different PC's. These are the extra lines added to sip.conf and extensions.conf respectively. sip.conf [jitha] type=friend host=dynamic secret=jitha context=sip dtmfmode=inband [sudhananda] type=friend host=dynamic secret=sudhananda context=sip extensions.conf [sip]

On 22/07/2019 12:41, Adam Weremczuk via samba wrote: > Hi Rowland, > > > On 18/07/19 15:52, Rowland penny via samba wrote: >> my plan would be to: >> >> TURN OFF DC2 > I did it on Friday afternoon after my numerous attempts to demote DC2 > failed. > This fixed one issue - made the network shares appear again across all > clients. > A new one has been

On 16/07/19 15:38, Rowland penny via samba wrote: > > You (because of your Samba version) can only demote the DC on the DC > itself, so just follow the info at the top of the page. Hello again, I'm trying to follow instructions for demoting: https://wiki.samba.org/index.php/Demoting_a_Samba_AD_DC I don't think I need to transfer FSMO roles since both controllers own them:

Hi, I'm trying to replace function call with call to wrapper(function_name, num_args, ...), where varargs hold args of original call. Function* launch = Function::Create( TypeBuilder<int(const char*, int, ...), false>::get(context), GlobalValue::ExternalLinkage, "kernelgen_launch_", m2); { CallInst* call = dyn_cast<CallInst>(cast<Value>(I)); if

Hi Dimitry, This makes sense if you think about it from the perspective that the string you want passing must be passed at runtime, and so can't use a const char * from compile time. You need to make the string visible in the compiled image, and use that as the argument. A string is an array of 8-bit integers, so you need to create a ConstantArray. Value *v = ConstantArray::get(Context,

Hi Rowland, On 18/07/19 15:52, Rowland penny via samba wrote: > my plan would be to: > > TURN OFF DC2 I did it on Friday afternoon after my numerous attempts to demote DC2 failed. This fixed one issue - made the network shares appear again across all clients. A new one has been discovered though on one of our CentOS 5.11 boxes. Any command (like sudo or ssh) that needs authentication

I'm trying to get two linux machines with kphone-3.11 two communicate with each other over asterisk. I have them configured correctly on asterisk to use sip channels, but when I call from one phone to the other I don't any voice communication between the phones. According to the phones I'm connected, but according to asterisk, I get the following message: -- Executing

Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 -------------------------------- ACK

Hi, I've been messing with getting SIP working for days now, with limited success. I've got Asterisk set up on a remote server with the echo test. Please try it out to verify I've got the server working right: sip:robot@nixon.butchwax.com Running FC1, ThinkPad T22, headset thru the soundcard. Asterisk is asterisk-1.0_RC1. No NAT. The phones I've tried so far are as

Hi, I am new to Asterisk and SIP. I have just installed ser as sip server and asterisk ser is in 192.168.6.244, without authentication kphone as sip client in 192.168.6.254 asterisk is in 192.168.6.100 and did not install hardward on my pc in sip.conf , i add following lines ... register:jimmy@192.168.6.244/1000 ... [192.168.6.244] type=friends host=192.168.6.100 fromuser=jimmy

Hi, i just installed latest asterisk on fedora rc2 and on the same machine i installed a sip soft phone called kphone. Kphone complains about /dev/dsp being used and can't place/answer calls (/dev/dsp is obviously used by asterisk) . how can "share" my sound card with these two programs? or can i disable the sound card in asterisk so i can use kphone to place/answer calls? BTW kphone

I'm having trouble with kphone on our system. It's using ulaw on an internal network. No NAT. I had it working fine with very similar hardware (an old Dell Optiplex GX1) running as an LTSP terminal. But then I put the same sound card in an Optiplex G1. Kphone will answer the line fine when I call it (call coming from the * machine), but when we try to get kphone to dial, each GUI

Dear useRs, Some time ago I queried the list as to an efficient way of building a function which acts as ls() but with a different default for all.names: http://tolstoy.newcastle.edu.au/R/e6/help/09/03/7588.html I have struck upon a solution which so far has performed admirably. In particular, it uses ls() and not its explicit source code, so only has a dependency on its name and the name of

Got no responses to this, but the list seemed to be down for a while, so here it is again. Sorry for the extra bandwidth! John Hi, I've been messing with getting SIP working for days now, with limited success. I've got Asterisk set up on a remote server with the echo test. Please try it out to verify I've got the server working right: sip:robot at nixon.butchwax.com

I just can't get kphone to register with asterisk, i can make calls to the demos and even get into the mailbox but kphone cannot register. Here's my story. Can you help me?? Please I have installed asterisk on debian using apt-get install asterisk. I have configured an extension in extensions.conf as follows exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt) exten =>