similar to: Asterisk : No Sound Issues

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 everyone, I just installed asterisk on my system with the purpose of rerouting calls on sip channels. I don't think i need any hardware for that. I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both of them and to call one from the other on the same machine, however could not. I 1-) I could connect sjphone in isolation to freeworld dialup howver i got no sounds

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

Hello, I am using chan_capi and would like * to behave differently depending on the MSN the caller dials. Is there a way to route incoming ISDN calls to different contexts based on the MSN dailled? I have tried something like msn=1234 incomingmsn=1234 context=msn1 msn=4567 incomingmsn=4567 context=msn2 in capi.conf but with no results. Thanks for any hints. -Walter -- Walter Doerr

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

Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

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, Is there any way to connect (register, initiate and receive calls) with Asterisk to FWD through NAT? Since I own my router port forwarding is not a problem. I tried with Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwd.pulver.com but since Asterisk still use internal IP in some SIP fields I got "479 We don't accept private IP contacts. Please set your external

Greetings, I'm having a job getting asterisk to register with a Huawei softX3000 softswitch via SIP. I keep getting 401 Unauthorized. Funny thing is I can successfully register SJPhone, a PA1688 IP Phone as well as a WiFi Phone against the switch without *any* problems. I think it's got to be something as simple as perhaps the register string which is currently

I am having trouble with this. I can dial 1800 numbers fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten =>

OK, background/config. running * (show version reports 0.9.0) on Mandrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci & cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency

Hi, I have read quitea bit of the available resources and have this idea of asterisk. Would someone kindly answer these briefly 1-) Asterisk does not need a sound card...but if i am to record voice into an extension or dial from CLI ( basically use asterisk itself as a softphone ) then i need a sound card. : Yes/No a-) If yes creative soundblaster pci 128 is my best bet. Yes/No 2-) Which is

Hello, I want to buy Grundstream sip phones. Could someone tell me if thay are working with Asterisk ? I want to buy also codec G729. How many channels I should buy if I have 24 ext. Let's say if I buy 20 and there is 20 calls the other calls with use another codec then ? Reagds, -- Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

Im trying to get IAX to work between my * and FWD. I activated my iax2 account on iax.fwdnet.net and I get the output: "Registered to '65.39.205.121', who sees us as 68.14.203.254:4569" when I start asterisk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave

We're on track for a release of Asterisk 0.4.0 soon. I'd like to try to see to it that we have squared away our SIP implementation by then, and after that point, try to keep it in tip top shape. In general, I find that SIP is extremely fragile, and every time I try to fix one bug, I end up creating another somewhere. What I need are strategies for verifying that the SIP implementation

Hi ;) I'm a french newbie and i installed asterisk 1 day ago. I've got an ATA186 and a computer with Sjphone installed. If i want to call the sjphone from the ata or call the ata from de sjphone everything is ok. My problem is ,that i can't call the voicemail or any other phone number ..as 600 for exemple from the ata or the jphone. I don't know why but i looked after a long

Hi. There is not much info on the FWD site about this. What codecs do they use? When I try to connect with X-Lite, it works with GSM. When I try to call out with *, it wants G729. I have disallow=all and allow=gsm in the sip.conf. I end up getting errors: Unable to find a path from G729A to GSM Unable to find a path from GSM to G729A What's up with that? I was able to make a call once

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