similar to: Incoming calls picked-up then simply hanged-up

Never mind. The Asterisk@Home documentation is incorrect the echotest is in *43 and it works fine. -----Original Message----- From: fhunter [mailto:fhunter@survivorsoft.com] Sent: Thursday, May 12, 2005 4:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up Something else I have noticed when

I am having problems getting incoming calls from the PSTN to route to extensions, digital receptionist and even voicemail. When I call a DID number for one of the lines, it rings twice then says: "Goodbye" and hangs up. (logs to follow below configuration info). When I dial 7777 it goes to the digital receptionist without any problems. The system setup is simple; I have 8 PSTN

I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the new box, I've installed a generic ebay X100P. I don't have my livevoip or voicepulse accounts set up yet on the new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP.

All, I upgraded my asterisk setup from CVS on or about 12/15. Suddenly, *all* of my incoming calls are coming up as FAXes. I had to disable my fax extension because every call to my POTS line was getting redirected to my FAX machine. After removing the FAX extension, if I call my POTS line from my cell phone, I get the following: *CLI> -- Starting simple switch on 'Zap/1-1'

Hello: I'm newbie in asterisk, please help me. My context is as follows: 192.168.4.2 --> Asterisk 11.13.1 complied from source 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension 7777 (configured as a hotline on TG100) to asterisk server, but asterisk server sends me "SIP/2.0 401

2014-11-12 2:45 GMT-02:00 Luis Eduardo Cortes <luedcortes at gmail.com>: > Hello: > > I'm newbie in asterisk, please help me. > > My context is as follows: > > 192.168.4.2 --> Asterisk 11.13.1 complied from source > > 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway > > When I call from a GSM cell phone, my TG100 GSM gateway answers and > dials

http://defect.opensolaris.org/bz/show_bug.cgi?id=759 Summary: ''zpool create -o keysource=,'' hanged Classification: Development Product: zfs-crypto Version: unspecified Platform: i86pc/i386 OS/Version: Solaris Status: NEW Severity: minor Priority: P3 Component: other

Greetings ocfs2 folks, A client is experiencing some random deadlock issues within a cluster, wondering if anyone can point us in the right direction. The iSCSI connection seemed to have dropped on one node briefly, ultimately several hours later landing us in a complete deadlock scenario where multiple nodes (Node 7 and Node 8) had to be panic'd (by hand - they didn't ever panic on

In a sol10 box which in ZFS filesystem,I took a snapshot of whole sol box (root dir) and then made some changes in /opt dir(30 - 40 MB).After this ,When I tried to rollback the snapshot,the sol box got hanged.Does any one faced similar issues? Is it depends on the size of changes we make?Please comment on this. -- This message posted from opensolaris.org

Hi all, I've got a problem with my FXO cards. I've configured them to give a service to people on PSTN network, to call the lines connected to my Asterisk by a digium fxo card, and dial my VoIP network numbers. PSTN -> Asterisk -> SIP Client The problem is when a call is made by a user from PSTN network and, after talking or not, hanging up, the line is not hanged up by asterisk.

Dear all. I have next question. I am using SIP protocol to connect to VoIPGW. Now I need in my extensions.conf in script to operate with phone number that is passed to asterisk and insert it into database. Which parameter is holding A number and can be used in extension script? Thak you in advance CELL PHONE -> GSM OP ->VoIP GW

Hi, I use asterisk with sip3000 device with "sip-aho" connected to PSTN and "sip-ahi" connected to a phone. When call arrives from PSTN, the *phone continues ringing even after caller hanged up*. The dialplan contains the following lines: [from-pstn] ... exten => 99,n,Dial(SIP/sip-ahi,30,g) exten => 99,n,Hangup() The asterisk properly detects hangup of the caller as I

I wish to set up two PSTN calls and then connect them similar to Jajah (is this called 3pcc?). The PSTN interconnect is handled by a third party SIP provider. I can do this using the manager or call files. An example (using php) would be: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n");

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

Hi List, I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The problem is that if the receptionist is on the phone and looking up something on the PC she some times

Hi, I have a PHP code with AMI to using in click2call system. here is my code: $user = "usernamr"; $secret = "secret"; $channel = 'SIP/' . $sip; $context = "from-internal"; $waitTime = "20"; $timeout = 20000; $priority = "1"; $maxRetry = "2"; $pos = strpos($number,

I have 2 issues which I need to resolve on our production Asterisk server: We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I

Okay, I'm going to preface this by saying I'm sure I've overlooked something really basic here. I just need someone to hit me with a clue stick and point out what I'm missing. I've got a TDM card with four FXO modules. I've plugged one of them into a PSTN line. I'm working through the examples in the Asterisk Documentation Project guide, but I can't get Asterisk

Hello All, I have this strange problem, I can dial out with my sip phone and it seems to work relatively well, but when I call in, the line just rings and rings, I get no indication in asterisk that it's detecting an incoming call. The strange thing is that in ztmonitor 1 -vv the rx volume goes from around 99% when the line is idle to 0% as soon as the line starts ringing. I'm located