similar to: SIPphone All-in-One: coments anyone?

I have a lot of experience, all of it pretty good, with various Sipura products, Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones connecting into Asterisk as clients. Good sound quality, great reliability. I've tried two of the units named in the subject line, and frankly I'm frustrated. Calls usually start out OK, but within a brief period the sound goes totally to

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

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

When attempting to call one of the example numbers, like 17474745000, I only get "488 Not Acceptable Here". It works fine when I configure the softphone (Xten X-Lite) to use sipphone's server directly. Am I missing something? Here's my relevant config sections: sip.conf: in [general]: register => 17472442457:mypassword@proxy01.sipphone.com [sipphone] type=friend

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 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

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

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

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

Hi all. I found a connection error from sipphone.com. It seems 'realm based authentication' by sipphone.com. any ideas? Regards. mack

I just got an email from SIPphone advising that there have been problems with the above firmware and advising to reload from their server. This does in fact reload 1.0.4.35 into the phone. And now voicemail has gone AWOL again. -- Dave Cotton <dcotton@linuxautrement.com>

Il giorno mar, 06/10/2015 alle 15.14 +0100, Rowland Penny ha scritto: > Would you care to share your very secret smb.conf file ? > > Rowland [global] workgroup = DOM netbios aliases = s-software, s-print server string = %L interfaces = lo, eth0 passdb backend = tdbsam:/etc/samba/account.tdb log file = /var/log/samba/log.%m max log size = 50 smb ports = 139 unix extensions = No

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

Hello All. I started setting up my Asterisk system yesterday and everything was going well, i have registered with sipphone.com and set-up my Asterisk system to register with sipphone per the sip.conf file below. It was registered perfectly but I could not receive calls so I added in the line "insecure-very" and I then used the Washington DC access number to test and the phone

Should SipPhone support ISN routing for their 747 ITAD? Cast a vote: http://forums.gizmo5.com/viewtopic.php?t=10197 Meanwhile if you're interested, you can use the Nerd Vittles 'bandit' ITAD #1089 to call a SipPhone/Gizmo5 subscriber via ISN, which I think is clever (Karl tips his hat to Ward Mundy) and it's also really, really funny.

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

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

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 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",

I resend this message, as it was not posted on the list first time I send it.... Dan ----- Original Message ----- From: "Dan" <dtoma@fx.ro> To: <asterisk-users@lists.digium.com> Sent: Friday, June 27, 2003 10:13 AM Subject: Re: [Asterisk-Users] Basic Asterisk questions - personal coments > > Why is it that most users who don't understand threaded email is on