similar to: getting asterisk to reliably answer a voip line

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

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'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

I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware

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 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",

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

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

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

Hello all, I'm having a problem with getting incoming callerid to a lan-connected phone. The Asterisk server is connected to the Internet, and a Grandstream BT101 phone on a lan interface: INTERNET ----(eth0) Asterisk (eth1) ---- Grandstream (192.168.1.51) The phone registers with the Asterisk server as ext 20. I can initiate and receive calls from the Grandstream phone fine. The

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"

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

I have written this configuration script which uses OpenSky to make Skype calls directly from Asterisk devices using my companies SIP to Skype gateway. Users can dial skype_anyskypeusername or manually add names or extensions which can get mapped to the correct dialing sequence. The right sequence is username at opensky.gizmo5.com but that gets mapped to sipphone address so I set that up to map

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

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

Hello, I am new user of Asterisk. Yesterday I was trying to call from softphone to Asterisk, and that Asterisk routes this call to sipphone.com provider. I have found information on internet about how to register to sipphone and it seems that I have done. "sip show status" (or similar command) in CLI was showing me that I was registered. To call was not working, and on Asterisk's

I've been wondering for a long time why my reinvite option is not working with Broadvoice anymore. It happend during the time Broadvoice was having a lot of issues, so I decided to wait. Recently I decided to test the same sip.conf with another VSP (SIPphone) and it worked fine! No issues on the reinvite. Note: clients and server using ULAW (only), no NAT or firewalls, public ip address and

I'm attempting to do this in an AGI program: char line[80]; ... printf("EXEC PARKANDANNOUNCE pbx-transfer:PARKED|300|Console/dsp\n"); printf("GET VARIABLE ${PARKEDAT}\n"); fgets(line,80,stdin); The script hangs because asterisk doesn't provide any response to stdin. Elsewhere in my program, I have e.g. printf("EXEC READ foo|enter-password|4\n");

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.