similar to: Audio Problem...

Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone -> SJPhone, and also SJPhone -> 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

Guys.. this is killing me.. I hava a laptop running sjphone and I have 2 dial cmds to connect to that laptop in different places, first, on the main phones context like this: exten => 202,1,Dial(SIP/laptop,20,m) so each phone can call it, and it works great. Now, I have anyother cmd on a different context, which is the IVR, like this: exten =>

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 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",

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 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 fellows, Pretty new to CentOS. I was trying to find a way to boot CentOS into command prompt instead of GUI (or without loading any services). Tried using 'Crl+Alt+F1' at the boot process, but, that holds the screen at mounting and doing fstab and doesn't proceed further. Is there anyother way to boot CentOS into command prompt without using Rescue option from the installation CD?

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

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

Hi all, I have the following problem, working with a SIP provider, if i setup my SJPhone to register directly to their STUN server and working over a 384/128 ADSL i have a really good quality, but then if i configure Asterisk to register to the same provider over the same 384/128 circuit the quality is REALLY BAD. The obvious difference is that using directly the SJPhone i am using STUN, while

I recently setup an Asterisk Server- I was able to follow a tutorial from http://www.automated.it/guidetoasterisk.htm#_Toc49248752 Until it told me to call another line, let it ring until voice mail picks up. My problem is the tutorial left out how to configure a SJPhone so that it connects to my asterisk server not directly FWD. I've tried everything I can think of, I must be missing

I'm currently using the SJPhone softphone with asterisk for remote SIP. When I dial into the voicemail, and attempt to pass the extension, I "hear" the sounds, but asterisk is not receiving any DTMF signals. If I use the Estera softphone, asterisk does receive the DTMF signals. Normally, I'd just say "Use the Estera" softphone to myself, but that's not an option,