similar to: newbe question sip.conf

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

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

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>

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

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its

Hi, I have a queue with one agent added using AddQueueMember (FAO|Local/1001 at from-sip|0||Agent/602). My extensions.conf is [general] static=yes writeprotect=yes autofallthrough=no clearglobalvars=no priorityjumping=no [from-sip] exten => 100001000,1,Dial(SIP/100001000,,t) exten => 1001,1,Dial(SIP/1001,,t) exten => 1002,1,Dial(SIP/1002,,t) exten => 1003,1,Dial(SIP/1003,,t) exten

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

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

I have a new wellgate 3802 unit. I have not gotten it to register with asterisk 1.2.6. My proxy setting is the correct IP in the 3802. My security config is 1001/1001 and 1002/1002 on the wellgate (simple at this time). My sip.conf has: [wellgate3802L1] type=friend dtmfmode=inband username=1001 secret=1001 host=dynamic canreinvite=yes nat=no context=wellgate [wellgate3802L2] type=friend

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

HELP! Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com

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

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

Hi All, I'm running 1.2.9.1 and have a sip user setup with accountcode=4444 in the context. lab1*CLI> sip show peer 1234 * Name : 1234 Secret : <Set> MD5Secret : <Not set> Context : sip1004 Subscr.Cont. : <Not set> Language : Accountcode : 4444 AMA flags : Unknown CallingPres : Presentation Allowed, Not Screened Callgroup

Hi, I have a requirement to limit the calls to our agents via a queue to 5 minutes. I had posted this to a previous thread by name "Maximum talktime in a queue?" One work around that was suggested was to use the S(x) in the dial command to the agents, so that all calls to that extension would be terminated after x seconds. So I modified the dial command to the agent as: exten =>

Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to

Hi I need help This is the error: Really destroying SIP dialog 'NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.' Method: SUBSCRIBE -- Executing [00919885497796 at internal:1] Set("SIP/1001-0000000b", "CALLERID(num)=8790771141") in new stack -- Executing [00919885497796 at internal:2] Dial("SIP/1001-0000000b", "SIP/00919885497796 at sonetel")

I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on

Hi All, I am using asterisk 1.4 branch on server. Here is a my dialplan. i have set the incoming route to incoming context, and then i have set dial with local channel, The call comes to my server and the call is routed to matched case, so my phone 1001 ring for 30 seconds. If i got the NOANSWER then the channel is not passing to next priority. I need to pass that channel to the next priority of