similar to: Cannot make calls

Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the "s" extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general]

Hi, I installed asterisk server in my linux box. I configured a user 1000 using xlite and registered with asterisk server in the same linux box. I configured one more user 1001 in other box and this user also got registered with asterisk. But i am facing two issues here. 1. When a call is made from 1001 to 1000 i could see an incoming call blinking but no audio flow is observed. 2. When i made a

Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go

This is probably an easy one. I have just setup Samba on my Debian 2.0 system with kernel 2.0.34. It was extremely easy set up, great job Samba and Debian people. My problem is that I can see both of my Debian computers in Network Neighborhood but can not log in to both of them. The one I can log in to is Debian 1.3 with 2.0.33 on it. When I double-click on the Debian 2.0 machine it prompts me for

all, I am having a weird situation . I downloaded the latest samba version and followed the installation instructions. ./configure make etc... Then I tried to run samba but if use /etc/init.d/smb restart I get bash: /etc/init.d/smb: No such file or directory because there is no such file at all. If I try the following: [root@localhost RPMS]# ls sam* samba-client-2.0.7-36.i386.rpm

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

Beginner to Asterisk, but not beginner to VoIP FreePBX front end running on a dell 1550 and XLite running on a different Woindows XP box Both boxes connected via switch on same subnet. No NAT involved On FreePBX I created a new extension 1001 with a SIP password of 1001 On Xlite, username is 1001, password is 1001, authorization user name is 1001, and domain is IP of Free PBX XLite tries to

Hello guys, I need your help related to a mediatrix 1204 configuration. I read some of the messages that you posted in the asterisk users mailing list about the mediatrix 1204 and decided to contact you. I know that the community is not related to Mediatrix devices, but so far I have not found any other group that has work as much as you with them. I bought the mediatrix in Mexico and my provider

Any expert suggestions most welcome , I am really stuck .. thnx -----Ursprungligt meddelande----- Fr?n: Ashish Garg Skickat: to 2003-05-08 09:04 Till: Jair; John H Terpstra Kopia: samba@lists.samba.org ?mne: Samba Installation help for a domain HI Guys, I need some help regarding Samba Installation for a domain. My Windows 2000 clients connect to a paricular doamin. My Samba

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Hey guys... one last thing. I have set up agents in my Asterisk... and one agent refuses to log out. I have tried to log out from Xlite. I have tried from the console... AGENT LOGOFF 1001. It still gets the call. If I shut down Xlite, it still tries to contact agent 1001, but gets a congested message... if I bring Xlite back up, it gets the call. If I kill Asterisk and restart... its _STILL_

Hello, I'm trying to use presence with PJSIP and I have a "issue". I created correctly hint priorities like: exten => 1000,hint,PJSIP/1000 exten => 1001,hint,PJSIP/1001 Extension 1000 can subscribe extension 1001 y vice-versa. The problem is when the extension 1000 make or receive a call. In the softphone where the extension is present on buddy list, the extension appear

Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the help desk support on the Suse

Hi All, I'm getting a strange warning message when I perform a MYSQL data lookup. The operation performs fine, I retrive the data I'm looking for and continue on through the dial sequence without an issue. I'm wondering if this warning message is something to be concerned about, can't find any info about it. warning message: Mar 28 15:55:40 WARNING[27481]:

Hello, We have a strange situation (asterisk 1.6.2.14), where we get a result for DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL. This is the (relevant) test dialplan: -------------------------------- [incoming-private] exten => _X., n, Dial(SIP/1001,30) exten => _X., n, NoOp(${DIALSTATUS}) exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1) [incoming-status] exten

Hi gang, In 500 words or less (if possible), please explain what is a legal music-on-hold file? My boss hates the stuff provided with the distribution and I figure that I'm asking for trouble if I take my Les Mis tracks and run them through Audacity and SOX to make new files. Thanks in advance Danny Nicholas -------------- next part -------------- An HTML attachment was

I would like to add the ability for our users to be able to press "0" whenever reaching someone's voicemail box to re-reroute them to the auto-attendant. Here's a sample extensions.conf: [incoming] include => ciscophones exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,BackGround(auto-greeting)

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

I have a T100P card connected to an Adtran and then a T1. I have added the following configurations to Asterisk...but, when I dial 9 and then a local phone number, it bounces between the dial tone and silence and the *error* light on the Adtran blinks. zaptel.conf span=1,0,0,esf,b8zs fxsks=1-8 loadzone=us defaultzone=us zapata.conf [channels] context=from-sip signalling=fxs_ks

Hi all. I'm trying to get a simple configuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on