similar to: x-lite with wireless connection

Im experiencing a problem with a current setup and I've run out of ways to debug it and come to a resolution. I have two E100P's in a machine which is routing traffic over the internet to a machine that has 1 E400P connected to the PSTN. Clients are able to make calls successfully but when the call is connected experience one-way audio. They cannot hear anything said by the person

How do you set the expiration date in ASTCC? DO you have to customize the CGI script? A maintenance fee field would be nice as well. Anybody? -- Kelly D Griffin Network Engineer Tantella Wireless http://tantella.com 479.273.9992 Voice 479.464.8998 Fax

I have an asterisk server up and running, using Firefly in IAX mode works great, even with Firefly behind a NAT (as expected, since IAX works really well with NAT). Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from behind the NAT, and I can't seem to get there. At this point, the phone will successfully register with Asterisk, and the Asterisk qualify messages get

Hi All, I am getting this error when i try to compile the "Linphone" package by typing----- make. please help me i am feeling very frustrated with this error pasdt from 7 days i am getting this error. please help me. speexec.c: In function `speex_ec_process': speexec.c:112: `spx_int32_t' undeclared (first use in this function) speexec.c:112: (Each undeclared identifier is

Has anyone had any experience with wireless LANs and Asterisk? We have and here are my impressions. We configured an Asterisk in the office as a precaution to see how it would work for our own retail customers. Our office is open space, about 800 sq ft. (20x40 area). We use Snom200 and Grandstream SIP phones. Using the latest Linksys wireless access point (WAP54g) and 3 wireless bridges

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

Hi all and greetings from Australia, I am looking at a suitable VPN to secure a wireless notebook hanging off a private, wired network. Is tinc suitable for this? specifics: "notebook" 192.168.205.63 "AProuter" (access point/computer that the notebook will connect to): 192.168.205.31 What numbers should I give to the various nodes? At this stage, only one notebook VPN:ed

Dear All, I bought one IP PHONE from Integrated Networks which was showed to wiki too: http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks I have problems with the Asterisk authentication. It does't want to LOG IN to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and all paramters seems to be correct. Does somebody use the same IP PHONE with

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

How do I set asterisk not to answer incoming PSTN POTS calls? I want to be able to use the line for outgoing calls only. -Thanks Tim

Dear all, My office have an old asterisk PBX system (asterisk 11.4), and it encrypt all the SIP User's secret. But the voip engineer before me didn't save / documented those password. Now the server's hardware is begin to broke, it hangs a lot, and have a lot of call problem. We already have a new asterisk PBX to replace it, but we have difficulty to retrieve the encrypted password.

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

I have just setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default => quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a standard 1.0.3

I don't know if this is already common knowledge, and it's not specificly for Asterisk, but if you are using Cisco phones and want to roll XML applications, make sure you have "Connection: close" in your HTTP header. Without it screen loads are very sluggish. In PHP, do: header("Connection: close"); I whipped up quick-and-dirty PHP/MySQL/Cisco XML directory and

I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get:

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hi, I am running Redhat RHEL 4, authentification is via kerberos against and AD server, usernames are supplied via ldap service running on another redhat box - winbind has been seg faulting repeating when accessing samba - always the same error message... see logs below - can anyone tell me whats going on? Mar 14 16:12:45 firefly winbindd[14752]: [2008/03/14 16:12:45, 0]

How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44

Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. bkw

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL: