similar to: Echo with two IP phones through Asterisk using SIP

I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I did a modprobe on ztdummy I was able to enter into a conference room using my softphone clients. I'm using SJphone and Firefly. I have noticed a significant delay (1 to 3 seconds) while talking within the conference room. I have tried both clients, SIP and IAX protocols and various codecs. I have also tried it from different

I have searched for how to locate echo cancelation on SIP clients, but cant find anything and echocancel=y doesnt seem to have any effect. Configuration: CVS-HEAD from last month iPAQ h5500 with SJPhone (gsm/ulaw/alaw) Problem description: When I place or receive a call I hear a faint delayed echo of myself. The other party hears a really bad nonmuted echo that makes the call unusable. Aside

Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_ W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewIte m Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid =27 But looking for real world feedback. Cheers,

Hi, I'd like to try using a good quality microphone and a set of PC speakers (in the first instance) to create a powerful speakerphone; if I get that working, I'll probably try more elaborate audio equipment. For this to work, I'll need software acoustic echo cancellation, or the caller at the other end will constantly hear his/her voice echoing back. I gather Asterisk can do

Hi ALL i implemented asterisk as my home PBX system my * machine recieves a call and transfer this to my computer The problem is this that i get my voice back mean there is too much echo(there is no complain from the caller). I have set following values in zapata.conf echocancellation=yes (also tried different powers of 2) echocancelwhenbridged=yes is there any other setting or not ??or this

I have the following situation: Asterisk 1.2.1 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 Some 501's local to my network, some across the great INTERNET divide. PRI connected to Sangoma card. Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker phone when calling from extension to extension. echo not present when calling outbound

Hello all, One of our clients reported that they are experiencing echo in SIP calls (even on internal ones). What do you think could be causing echo in internal SIP calls? We're using Polycom telephones, do you think they could be causing it? Thanks, Alex

Hi, You need to have completely replaced the Microsoft driver, because it doesn't support the headset or ctp Bluetooth profiles. This gave me fits! I followed the instructions at http://www.windowsdevcenter.com/pub/a/windows/2005/07/05/bluetooth.html and it works with both a Plantronics and a Motorola Headset, and I can answer calls with idefisk, eyebeam, x-lite, and kapanga. If you end

Now here is strange problem i experience. Setup is easy, IAX line out with SIP softphone registered to Asterisk. All work fine except for one client. When using Sjphone the other end can not hear a thing. When using X-pro the opposite happens, local user can not hear a thing. These softphones work fine on other clients on same network. I've also tested several headsets but same outcome. Also

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

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

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",

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

Hi, I have * working with X-Lite and Sipura adapters, but I have one person who is linux based, and is trying to use Linphone and Kphone. His end works, but I get very bad echo on my end. Have any of you folks been able to get linux based soft phones working well with *? I'd appreciate links to howtos/docs if you have them, and/or samples of working configs for * and the linux

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