similar to: Help with Gizmo from outside firewall

I am still trying to figure out why samba wont let me create in subdirectories I've tripple checked everything and a few things leapt out at me. *) I can modify a file inside a directory that I cannot create a file in. I did not know it was possible under linux to do that. *) If I set all perms on 777 I can create. But neither 775 or 755 will allow it. *) When I go into a mount

Got an email this morning with the subject "Welcome to Gizmo Project". I didn't sign up with those yokels. Anyone else got spammed by them?

I have been successfully connected (incoming and outgoing) to FWD for a very long time. A few months ago, I changed from SIP-based FWD service to IAX2-based, and that went fine as well, both incoming and outgoing. At the time, I was running Asterisk 1.0.3 Stable. I rarely use the service, so other than noticing that I was always successfully registered to FWD, I didn't make or receive calls

for inbound calls, i can register context = from-sipphone register => 1747xxxxxxx:passwd@proxy01.sipphone.com but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^

Hi, I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a user via SIP and playing a reminder file when the user picks the phone. I use Gizmo service for SIP and I'm able to call through it. However, when asterisk dials a number, Gizmo first answers then tries bridging 2 channels. Right after answer Asterisk starts playing the reminder. It obviously results in

Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer

Hi, I've set up a Gizmo Project account for access on my Nokia E61 because they work through NAT. Trouble is If I include my gizmo account in an asterisk hunt group and I'm not connected (phone is off / outside wireless coverage) the gizmo project always answers. Either the call goes to voice mail or if I turn voicemail off the call gets answered by a recording saying I'm not

I have a sipphone.com account, with asterisk set to answer incoming calls, using the following settings (phone number and password omitted) in the Peer Details for the SIP Trunk: allow=ulaw context=from-pstn dtmfmode=rfc2833 fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx host=proxy01.sipphone.com insecure=very secret=xxxxx type=peer username=1747xxxxxxx The Asterisk machine is

I've battled this one for a few days now and its costing my sanity so I am hoping that someone here has an answer I am running the samba server 3.0.4 on a linux 2.4.25 server. I can connect to my share point and authenticate fine I can create files or directories in the root of the sharepoint fine I can go into subdirectories and look at all the files... But I cannot create in any

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

With 1.6.1 svn: [2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout: -- Registration for '17470121145 at proxy01.sipphone.com' timed out, trying again (Attempt #30) [2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable to lookup 'proxy01.sipphone.com' [2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register: Probably a DNS

I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places the call on hold. During hold, the Sipphone user cannot hear music, only silence. The silence continues after the hold, though

Hi all, I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a while and it works fine .... I just added CALL OUT ... I have no problem with call setup ... the called party hears me ... but I can't hear them .... again if the call comes INTO the server both sides work fine. Just looking for some tips at where I should be looking .... firewall port forwarding ....

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Hi there everyone! I use asterisk as a home pbx. My internet connection is a DSL one, and I have a Linksys WRT54G that nat things for me in a 192.168.X.X style network. I've installed asterisk on my mac, and tried several examples I've found on the net (voip-info, gizmo, etc.) about how to create a Gizmo and a FWD trunk. However, all my attempts failed. The FWD thing kinda

Hey guys + gals, Would you know where to point me to for an open source or commercial implementation of Theora & Vorbis for your mobile phone? Maybe as a Symbian app? Maybe a Java midlet? I'd like to be able to download & play video files to the mobile as well as maybe stream them too? Am I asking too much? :-) Thanks in advance for your help! Callum.

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

Hi, Anyone know how to set up tftp server for grandstream. I gues it should be somethink like <tftpserver-dir> <mac-address> firmware.bin config.txt Is this correct ? And how should the config-file look like. ? I had search sipphone.com but did'nt find anything. /HHA _________________________________________________________________ Rethink your

my problem is this. I have google forward the call to gizmo5. I have this line in my sip file : register => user:password at proxy01.sipphone.com I believe this lines connects asterisk with gizmo5 so when it gets a call from Google, asterisk will answer it? At the end of my sip file i have this [Calls-From-Gizmo-Network] type=user context=demo disallow=all allow=ulaw allow=ilbc allow=gsm

Hi All, At the usual time, 12 Noon ET on Friday April 3rd, we expect Michael Robertson to join the discussion to filed questions about OpenSky and Gizmo5. I have been testing all of these Skype to X methods except SIP for Skype since I have no word from them. I can tell you that we've had good results with bith Skype for Asterisk and OpenSky. In fact, I am currently accepting calls to my