search for: 1747xxxxxxx

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

...g to suspect that it's Asterisk CVS HEAD that's possibly the problem... Finally, the machine that is connected to both FWD and SIPPHONE is on a public static IP address, so there are no NAT issues involved here, and no STUN services needed either. OK, here is the sip.conf entry: register=1747XXXXXXX:YYYYYYY@proxy01.sipphone.com/4321 [proxy01.sipphone.com] type=peer ;auth=md5 secret=YYYYYYY username=1747XXXXXXX fromuser=1747XXXXXXX fromdomain=proxy01.sipphone.com host=proxy01.sipphone.com nat=no qualify=no canreinvite=no disallow=all allow=ulaw ;context=default ;callerid="Hadar Pedhazur&q...

...ly), 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 host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx username=1747xxxxxxx password=xxxxx context=fromsipphone dtmfmode=rfc2833 canreinvite=no Any ideas? Am I doing something wrong? Thanks! -JD-

...number of my Asterisk server, not the calling party's SIP number. What's wrong? What I really want is that for inbound calls, I see the callerid of the SIP phone initiating the call. Here are the (hopefully) relevant parts in the config files... In sip.conf: ----------- register => 1747xxxxxxx:mypassword@proxy01.sipphone.com/1747xxxxxxx [sipphone] context=from-sip-external type=friend secret=sip_password username=1747xxxxxxx ;host=proxy01.sipphone.com host=198.65.166.131 callerid="My Name <1747xxxxxxx>: qualify=no reinvite=no canreinvite=no insecure=very [20] context=from-si...

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 behind a Linksys router (full cone NAT). About 25% of the time, when I call that number (from another sipphone account), asterisk answers the line, but about 75% of the time, asteris...

I've been looking at how our budgetone's have been failing and have found the following: A quick layout -- Latest CVS as of tonight. Sip phone behind NAT. * server with public IP address. -------from sip.conf for my phone: [1747xxxxxxx] username=xxxxx secret=xxxxx host=dynamic type=friend nat=yes ------- -------from the * log messages Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c, Line 3908 (check_user): Setting NAT on RTP to -1 Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c, Line 559 (__sip_ack): Stopping retransmission on '9...

...ewayed numbers work). To get FWD, IPTel, or SIPPhone to work, I've deduced I need to do something like the following after reading various online email archives (please correct me if I'm wrong): sip.conf: [general] register => XXXXX:password@fwd.pulver.com/1000 register => 1747XXXXXXX:password@proxy01.sipphone.com/1000 register => username:password@iptel.org/1000 extensions.conf: [default] include=sip [sip] include=sip [fwd] exten => _91339.,1,SetCallerID(XXXXX) exten => _91339.,2,Dial,SIP/${EXTEN:1}@fwd.pulver.com,tr exten => _91747.,1,SetCall...

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' -- Executing Macro("SIP/eric-8e80", "sipphone|1800xxxxxxx") in new stack -- Executing SetCallerID("SIP/eric-8e80", "1747xxxxxxx") in new stack -- Executing SetCIDName("SIP/eric-8e80", "...

...The only thing that seems weird is that is only happens when Gizmo originates the call. I can see the prompts and stuff playing on the CLI, but nothing gets sent to the other end. Also, if I answer a call, sound goes neither way. I've tried a bunch of things My SIP.conf has register => 1747xxxxxxx:password@proxy01.sipphone.com [gizmo-inbound] type=peer context=from-gizmo dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm nat=yes host=proxy01.sipphone.com insecure=very canreinvite=no externip=69.10.14.12 localnet=192.168.0.0/255.255.255.0 I have no idea what to check /...

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" response. From what I've read, this might be...