similar to: New SIP Phone

Hi If I have ser sending calls to asterisk, is there a way to get a different block called in sip.conf for each call (based on some variable, NOT username, From:), if not and they all hit one block which has contect=abc, then when that context is called/matched in extensions.conf, how can I have diff features for various groups of users. EG lets say I have a large company with 4 departments

I'm new on here, so I hope this has not been answered too many times already. I searched the knowledge base at the Wine site without finding a solution. I just switched from Mandrake to Lindows, which seems to have a capable version of Wine running. However, with two Winapps I've tried to install, the initial setup window (fullscreen, borderless, no controls) completely hides the

Having spent the better part of an hour searching the archives and voip-info I hesitantly ask what appears to be an obvious question but one I cannot find an answer for. Using Grandstream phones it seems that the only way to support Call Parking is to enable # transfers (i.e. use T in the dial command) since pressing the TRANSFER button on the BT phone is blind and one does not hear the call

http://www.smh.com.au/articles/2004/04/05/1081017104255.html SingTel ready to break into web telephony April 6, 2004 Singapore Telecommunications is teaming up with US internet phone start-up SIPphone to offer low cost, and in some cases free, phone services over the web. The deal, expected to be announced today, will allow SIPphone - started by MP3.com founder

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

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File

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

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}@??????) ^^^^^^

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 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 am trying to place an analog outbound call from a Sipura SPA-841 through a * server with a TDM400P and 4 FXO's. When I call in from an analog line everything works fine, I can talk over the SIP phone. When I call out, * says: == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 'SIP/sipphone-d29d' -- Executing Dial("SIP/sipphone-9eb0",

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

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

Hello, can anyone comment on how one could use SIPphone's $89 All-in-One adapter with Asterisk? Sounds to me like it should work as both a FXO and FXS. It would be a cheap way of getting started with Asterisk and PSTN. Any comments on the SIPphone FX200? Any comments on SIPphone in general? Thank you for your help

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

Hello when i dial a toll free no using sipphone i get this error message. How do i solve this? Any help will be appreciated. console message: Starting simple switch on 'Zap/2-1' -- Executing SetCallerID("Zap/2-1", "17473863282") in new stack -- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack -- Executing

Has anyone else had problems registering with Sipphone over the last few weeks? Previously, this had worked fine. I contacted Sipphone technical support, but they're not much help. register => 17471234567:password@northamerica.sipphone.com/123

I am trying to get asterisk to dial this address : sip:723@216.52.153.207 Using a softphone on my PC (217.168.168.49) it dials immediately and I get a voice prompt .. I have configured an extension, 1303 on asterisk, modifying the demo configuration : exten => 1303,1,Dial(SIP/723@216.52.153.207) When from my softphone I dial sip:1303@217.168.168.51 on the console I get : -- Executing

Hi, This is my first post here, so if I make any mistakes, warnings are always welcome. I'm working in an organization and we are migrating are domain controllers from windows to linux. I'm trying to deploy a working configuration of samba. Using openSuSE 10.0. I was following the Keith Robertson's how-to from IBM-Developerworks but in the end; the finished installation fails to

Hi I installed Asterisk on WhiteBox Enterprise Linux 3.0 respin 1 The process of installation was the following: First I compiled and installed Zaptel, in order to have ztdummy (uncommented in Makefile). I loaded the ztdummy (modprobe ztdummy) and then i installed Asterisk: make make install make configuration make samples I started Asterisk, and created one SIP account, with the following