similar to: SIP make outside call

Hi, I am can make local extension to and from SIP X-Lite softphone, but I can't dial out using X-Lite but local analog works just fine. Here are my conf files any idea? Thanks, David my sip.conf [general] bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) allow=all [3000]

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

I have just installed * from the latest CVS and I can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I will get the following error message: Spawn extension (default, Number-I-Dialed, 1) exited non-zero on 'SIP/200-6a22' -- Got SIP response 404 "Not Found"

Hi People, The Linux Environment I am responsible for is using ssh key pairs to allow access to a number or accounts on a number Linux Servers. I currently have the opportunity to re-design some of this. So I would like to tap into peoples experiences to see what might be some good changes to make. Specifically I have a couple of questions 1. Currently all of the key pairs we are using

I am just starting out with * so bear with me please. I have tdm400p with 4 fxo modules on it. When I call into the asterisk box from my mobile, I can see the asterisk console picks the call up and routes it to my computer with x-lite. There was no sound coming from either - just silence. I then decided to route it directly to voice mail to see if that would narrow the problem down, but it

Hi, I am trying to access the gmail account through net/pop library.However, I have an error such that: c:/ruby/lib/ruby/1.8/net/protocol.rb:133:in `sysread'': end of file reached (EOFError) from c:/ruby/lib/ruby/1.8/net/protocol.rb:133:in `rbuf_fill'' from c:/ruby/lib/ruby/1.8/timeout.rb:56:in `timeout'' from c:/ruby/lib/ruby/1.8/timeout.rb:76:in `timeout'' from

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Hi, I am trying to get quota working properly on dovecot 1.2.3 with postfix admin, amavisd-new, mysql, and postfix. So far I can see the db getting updated when messages are added and deleted from the mailbox. The problem I am having is that I have imported a bunch of messages from the old mail server that was running uw-imap and mbox to the doevcot machine using maildir++. As a result the

Dear fellow NUT-UPS users, I'm writing this message mostly "for the record" - for others who will follow in my footsteps. Apparently, I'm not the only one, trying to find a way to "make my UPS to initiate the graceful shutdown process earlier". The topic is kind of documented in various different places, and there are a number of howto's and forum posts in the

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

Been learning * now for a couple of weeks and have all basic features running including VM, MoH, FX lines, iaxtel, and FWD. However, I seem to be lacking documentation on a lot of technical things and am wondering if I overlooked something that is obvious to others. (I do have the Handbook, have been doing a fair amount of google searches, and read the README.* files.) Examples, Where should I

Can someone offer a hint on what I'm doing wrong with the basic * config? Just implemented * for the first time using yesterday's cvs. The initial configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956, and using two 7960's for initial testing. When one 7960 calls the other, I get the following and the call is dropped: -- Executing

I am blessed with this user forum and able to set my Dev-PCI Digium card working fine with the Asterisk server of mine (i)But today i just wanted to know if someone can help me to set X-Ten Lite to call PSTN line using my FX0 Currently , I am able to use X Lite to call another X lite user locally (LAN) I also has attached my setting together Thanking you all in advance --------------

Hopefully someone here can save my sanity. I have been trying to solve this problem for days now, but just cant put my finger on it. Im new to * so I have probably done something stupid! I have a TDM400P with one FXO module and a FXS module. The main problem I have is not being able to get the extension attached to the FXS module to ring or be able to make calls. It gets a dialtone fine but I

Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are. CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack -- Calling using options

i am working on asterisk. i am using fedora core 2 on my asterisk mechine. when i was working on stable version my voicemailmenu was working well. i can lissten to menu and send dtmf to control menu now i have compiled CVS version of asterisk. now when i configure my voicemail for any extension suppose i declared a voicemail box 9999 for user 3000. when i dial to 3000 i cannot have any menu there

I have a queue defined as [blabla] member = SIP/101 member = SIP/102 and in extensions.conf this: exten => 101,1,Queue(blabla,t) exten => 101,102,Congestion but when both Agents are busy then still the called party does not get a busy signal. What I`d want is when both Agents are busy that the caller gets a busy, not the long tones, like the phone is ringing, but nobody answers it (

Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia

Hello all, I want to setup Asterisk to forward a call if the dialed extension is busy. I do not want to wait on the line until the extension timeout expired. What I want is when I dial am extension currently Busy (Talking with someone), asterisk inmediately forwards my call to an extension I previosly defined. Someone could help me? Any clue will be appreciated. Regards from Spain. Ismael