similar to: Tone Detection Problem

I am having a problem with softphone registration, having read the list and watched it for a while for similar problems I just cant seem to figure out the problem. Using SJPHONE or DIAX , I can make outgoing calls but I can't get them to register with asterisk, I have other sip devices registering OK-7940's. From the list and the digium web site this seems to be a straight forward set up

Hi All, I hope this is the right list for this sort of request. I'm wondering if you all could recommend (or are) an asterisk integrator. I've been following the lists, etc, and have played with the software, but just don't have the time to really figure it out, nor to deliver a solution in a fixed time. I need someone who can help me spec the hardware and configure * for a small

Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ;

I hope this question is adequate for this list I use the nnet code from V&R p. 348: The very nice and general function CVnn2() to choose the number of hidden units and the amount of weight decay by an inner cross-validation- with a slight modification to use it for classification (see below). My data has 2 classes with unequal size: 45 observations for classI and 116 obs. for classII With

I have heard it mentioned several times by different people but can anyone explain to me how you can set up a single dialplan for 2 or more than asterisk boxes located on the same local network? MATT---

I''m having some trouble with the instructions at the link below: http://instantrails.rubyforge.org/tutorial/restarting_the_dev_env.html I had started at the tutorial at http://instantrails.rubyforge.org/ tutorial/index.html and had gotten as far as creating the cookbook2 exercise and had gotten to the "Welcome aboard you''re riding ruby rails" screen in explorer, but

Hi, I'm having a problem with DTMF tones from my SIP client apparently crashing the chan_capi driver. However I'm not sure whether this is a bug or misconfiguration on my part: if I set "softdtmf=1" in /etc/asterisk/capi.conf the problem goes away. Does the AVM B1 not support DTMF detection? The set up I have is using latest CVS (3 days old) running RH8 on a 933MHz P3. SIP

Hello, I am new to Asterisk, as of today. Installed on a RH9 box, with no problems. Built with 'make samples' as to get an understanding of how things work. Currently, I am utilizing SJPhone as a SIP client - not interested in shelling out cash for the IP phones, until I know I have a hold on things. I have run into a few problems I was hoping you could help me with : 1. When

I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address:

Hi. I have sjphone installed on windows and working except for dtmf. I read the docs for sjphone and it uses inband dtmf. I configired dtmfmode=inband but it still does not recognize it. Someone on the lists said that inband only works using alaw or ulaw but i tried only allowing that too but still no go. Hmm.. any other ideas? I can't get any other client to work on windows :-/ I

Dear all expert, (i posted this question one time, but i couldn't reach the answer -so allow me to post here) 1)I download asterisk realse version 1.2 beta1. After that i compile it successfully and run it with: asterisk -vvvc 2)I follow the instruction in http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html in sip.conf: i add two account: [ivan] type=friend username=ivan

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

> -----Original Message----- > From: H. Peter Anvin [mailto:hpa at zytor.com] > Sent: 06 June 2014 16:14 > To: Andrew Rae; 'syslinux at zytor.com' > Subject: Re: [syslinux] EFI booting over network - can't then load anything > > On 06/06/2014 07:27 AM, Andrew Rae wrote: > > Hi all.. > > > > Quick one if someone could show me a line out of their

> > From: Matt Fleming [matt at console-pimps.org] > > Sent: 25 June 2014 07:39 > > To: Andrew Rae > > Cc: Gene Cumm; syslinux at zytor.com > > Subject: Re: [syslinux] testing out 6.03 network booting... > > > > Andrew, could you try out syslinux-6.03-pre18? Peter pushed the release > > button yesterday and -pre18 contains my change. It would be

I have Asterisk running on my LAN with softphone clients (SJPhone) and Cisco 7940/60s, all using SIP. I also have a few remote sites connecting to my Asterisk server. I am getting an echo back of my voice when talking with one particular site. The caller does not hear an echo on their end. All calls on the LAN or to other sites do not produce an echo. When the caller places his SJPhone on

Hi there, I tried to get a few "Optipoint 400 SIP" working with *, but it refused to work properly. In my testing-network i have two Sjphones (they are working really fine) and three optipoints. I?m able to dial the number of a Sjphone on all of the optipoints. The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established. But when I

Hi, I notice that SJPhone is registering to asterisk with an expires of 120 secs. However, when I invoke the command "sip show peer [sip id]". I notice that the output indicates the expires 427 and the expiry is 900. Can someone explain these numbers to me? I also notice that just before SJPhone re-register, when I try to make a call to the SJPhone, asterisk will complain that

configuration ISDN BRI card : ISDN Olitec PCI 128 (hisax gazel) internet connection by ISDN 64kb/s dynamic IP nom de domaine registered to : dyndns.org avec ddclient to register IP par ddclient asterisk (on internet gateway) configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf) logical telephone SIP "SJPHONE" on 2 local stations "windows" (i don't succeed

Today's "frustrated programmer" award goes to Linphone, which has the following debug output: > (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the