search for: sipphon

.../05-09:07:50 Still didn't make or receive calls to FWD since the upgrade, but everything else has worked flawlessly (including sixTel, NuFone, etc.). All my softphones (SIP and IAX2) and Sipura-2000's work perfectly too. On to the problem... A few days ago, I signed up for an account with SIPPhone. When I did a "sip reload", which had the register statement, I immediately got a call "welcoming" me, so I thought everything was fine. It wasn't. I have been unable to make any calls to sipphone, and even though the registration appears to work (and my.sipphone.com shows...

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}@??????)...

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920", "SIP/att/xxxxxxx,80") in new stack Can you...

Hello All, one of the extensions fall into a loop, I don't know how to hangup that channel -- Executing [i at autoatten:2] Goto("Local/100 at sipphones-000001b2;2", "s,2") in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200 at sipphones-000001b2;2 -- Executing [i at autoatten:1] Playback("Local/2000 at sipphones-000001b2;2", "pbx-inv...

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 channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A Before somebody tells me "...

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 host=proxy01.sipphone.com username=17472442457 secret=mypassword fromuser=17472442457 fromdomain=prox...

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 informati...

...ER,30,rA(announce)) which should play file "announce" to the called party once they answer. I also tried exten => s,1,Dial(SIP/NUMBER,30,rG(default^play^1)) which separates caller and callee,for the same purpose. Here's the asterisk console: -- Executing SetCallerID("SIP/sipphone-cbfb", "NAME <NUMBER>") in new stack -- Executing NoOp("SIP/sipphone-cbfb", "Dialing 011XXXXXXXXXXXX to deliver file /usr/vt/result/200611135/test") in new stack -- Executing SetVar("SIP/sipphone-cbfb", "__MSG=/usr/vt/result/200611135...

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" below, I receive a "403 Forbidden&...

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

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 error for registration to 1747yyyxxxx at prox...

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 inse...

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 rang and the audio was fine....

...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", "Zap/g1/[phonenumber]|60") in new stack -- Called g1/[phonenumber] -- Zap/1-1 answered SIP/sipphone-9eb0 And then I get silence. The phone doesn't ring on the other end. I have attached my configuration files. Any hel...

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&qu...

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

Michael Robertson, founder of both MP3.com and Lindows, has launched a new company to supply inexpensive SIP phones ($129 for two) and related services. See today's press release at http://www.sipphone.com/tiki-index.php?page=SIPphone%20Inc My question for the list is who will be the first to report on the compatibility and usability of the SIPphone with Asterisk? The functionality described in the press release suggests the phones are pre-configured to work with Robertson's service so o...

I am getting the following on dial-out via Sipphone to a 1-800 number (numbers obscured): ------------------------------------------------- == Spawn extension (macro-sipphone, s, 3) exited non-zero on 'SIP/eric-9546' in macro 'sipphone' == Spawn extension (default, 1747xxxxxxx, 1) exited non-zero on 'SIP/eric-9546' -...

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 to work with IAXTEL wasn't a problem, bu...

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