similar to: SIP Notify contents showing 0/0 on VoiceMail

I am running Asterisk CVS-HEAD-07/14/04-16:28:29 and noticed that when I send a subscribe I get back a 403. This used to work in an old version which I forgot to record before upgrading to the above version. Any suggestion? I can register fine with the * server. Sip read: SUBSCRIBE sip:2486@192.168.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK46F2668 From:

Hi, asterisk Users, sorry for my bad English. im really newbie with this excellent pbx. But I 've a problem with callerid num when I recive a call from PSTN. PSTN-> SipGateWay(Welltech3504)-> Asterisk-> BT100 How can I configure my asterisk to receive the callerid from callers and not the callerid from the extension of the SipGAteway. Extension of Gateway (sip.conf) [115]

I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer.

Hello, I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec in my ata the the GW return a media not acceptable error. but If i add the g729 codec the all is ok. I see in the config of the cisco where to define codec for imcoming call but not for outgoing *Jul 17 15:57:02.604: Received: INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP

Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan

I am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an AT&T conf using regular analog phones. I don't have a great grasp as to what Asterick is capable of, but my

Hi, Am I hitting some limit in match? Consider the following example: > tst<-seq(100,125,by=.2)%in%seq(0,800,by=.1) > sum(tst) [1] 76 > seq(100,125,by=.2) [1] 100.0 100.2 100.4 100.6 100.8 101.0 101.2 101.4 101.6 101.8 102.0 102.2 [13] 102.4 102.6 102.8 103.0 103.2 103.4 103.6 103.8 104.0 104.2 104.4 104.6 [25] 104.8 105.0 105.2 105.4 105.6 105.8 106.0 106.2 106.4 106.6 106.8

Hi, Am I hitting some limit in match? Consider the following example: > tst<-seq(100,125,by=.2)%in%seq(0,800,by=.1) > sum(tst) [1] 76 > seq(100,125,by=.2) [1] 100.0 100.2 100.4 100.6 100.8 101.0 101.2 101.4 101.6 101.8 102.0 102.2 [13] 102.4 102.6 102.8 103.0 103.2 103.4 103.6 103.8 104.0 104.2 104.4 104.6 [25] 104.8 105.0 105.2 105.4 105.6 105.8 106.0 106.2 106.4 106.6 106.8

Hi I have a called fil consisting of the following strings. > fil [1] " 102.2 639" " 104.2 224" " 105.1 1159" " 107.1 1148" " 108.1 1376" [6] " 109.2 1092" " 111.2 1238" " 112.2 349" " 113.1 1204" " 114.1 537" [11] " 115.0 303" " 116.1 490"

I wrote a nice detailed post before, and then my mail program lost it for me... so here I go again... I've followed the same process with three different versions of asterisk, my local source copy from about 1 week ago CVS, current CVS from about 24 hours ago, and version 1.0.1, all three versions had identical results: I compiled/installed libpri, zaptel, asterisk I copied config file from

Patrick hi. Asterisk can do that, and you don't need VOIP lines. If you connect Asterisk to the net, and all employees have a VOIP phone (either hardware or software) then you're good to go. What do you need? To begin with, install linux on an old pc (well, not too old). Then go to voip-info.org and take a look at the Asterisk wiki. Everything you need is there. And of course, we're

I am having trouble locating webvmail on my * server. Is this a seprate porgram or does it come with *. I am running version asterick*CLI> show version Asterisk CVS-03/26/04-17:08:20 built by root@localhost.localdomain on a i686 running Linux asterick*CLI> Thanks Kurt __________________________________ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢

https://bugzilla.mindrot.org/show_bug.cgi?id=2486 Bug ID: 2486 Summary: allow ForceCommand none or similar Product: Portable OpenSSH Version: -current Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5 Component: sshd Assignee: unassigned-bugs at mindrot.org

Hi All... Can Asterick detect distinctive ringing on a POTS line and answer with different configurations? Thanks...

Hello, I'm trying to get started with asterisk/SIP so I was trying the demo that is provided in the extensions config file, but the call isn't "answered" by my server when I try calling the number that I registered at my SIP provider. I've registered with register => John.Doe:MyPass:MyUser@my-sip-provider/1000 in sip.conf and if I use "sip debug" I can see the

Can anyone please help me with sample IOS commands to get a Cisco gateway working properly with Asterisk. I cannot get my Cisco 2801 with BRI interfaces to call into Asterisk. The Cisco identifies itself as sip:.@datamerge.local. I cannot figure out how to get it to identify as sip:cisco@datamerge.local. The gateway works with other SIP servers that don't require authentication, but

Hi I am a newbie to * and I am having a problem which appears strange as I did not find any mention of it anywhere in my search. Simply speaking, I have an external SIP proxy server which I am trying to configure for incoming and outgoing calls from my asterisk installation. So here is my configuration in sip.conf [general] register =>

Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400