similar to: Problems with Analog - SIP phone conversations

Hi, I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS, where an incoming line is plugged and also analog phone plugged to the FXS port. Am faced with the problems below. - For conversations between analog phone and sip phone, Analog phone can't here the SIP user but Sip user hears. - Calling the PSTN

Leg/Transaction Does Not Exist" and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:7531 at 192.168.100.197:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: "asterisk" <sip:asterisk at asterisk>;tag=as4ea953db To: <sip:sip:7531 at

I am doing a little application to originate a call through Asterisk via AMI (Perl Asterisk::Manager). It logs in successfully, does an originate command with Exten: 0020 (which is set up to answer and wait for 60 then hang up) Channel: SIP/5101234567 at test-host (which comes to my desktop machine also running Asterisk). At the target machine I see only a CANCEL to which it immediately responds

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

On 11/05/15 18:56, bar?? tombul wrote: > What is your opinion about the cause and solution I thought I gave you the cause, your clients are trying to update their own dns records, they are denied and then the system you have set up to update dns carries out the update. The solution, stop the clients trying to update their own records, your clients are probably windows and if so, it is a

this is my named.log file. Have the dark line problem? best regards. May 10 08:11:08 samba named[752]: queries: info: client 127.0.0.1#54056 ( WPAD.test.com): query: WPAD.test.com IN A + (127.0.0.1) May 10 08:11:08 samba named[752]: queries: info: client 127.0.0.1#54056 ( WPAD.test.com): query: WPAD.test.com IN AAAA + (127.0.0.1) May 10 08:11:08 samba named[752]: queries: info: client

Hello community, I've been running Asterisk on an embedded device for about six months, and my operation has been largely trouble-free. I'm hoping I could get some help with a minor problem: Every week or three, my PBX gets stuck in a state where it can receive calls, but it becomes completely unable to originate outgoing calls until I do a "sip reload". After doing the SIP

Hey there! We had a great session last time with Ragavendra Talur, and I'd like to continue this again! If you're interested in a 5 minute lightning talk in this meeting, reply back and I'll put you on the schedule. Here's the call details: To join the meeting on a computer or mobile phone: https://bluejeans.com/6203770120?src=calendarLink Just want to dial in on your phone? 1.)

Interested in coming and giving a 5 minute lightning talk? We'll do our next meeting on the 16th so you have plenty of time to prepare (or forget about it over the end of the year, your choice!) Respond on this thread with what you'd like to present, I'll post another reminder as we get closer. Here's the call details: To join the meeting on a computer or mobile phone:

Hey there, As not everyone's come back from the holidays (and other things!), I'll move this to later in the month. Watch this space for more, I'll reannounce with enough time for people to sign up. - amye On Tue, Dec 12, 2017 at 7:07 AM, Amye Scavarda <amye at redhat.com> wrote: > Interested in coming and giving a 5 minute lightning talk? > We'll do our next meeting

http://bugzilla.mindrot.org/show_bug.cgi?id=1028 dleonard at vintela.com changed: What |Removed |Added ---------------------------------------------------------------------------- Summary|sshd does not forward non- |sshd does not forward final |query conversations to |non-query conversations to |client

http://bugzilla.mindrot.org/show_bug.cgi?id=1028 ------- Additional Comments From dtucker at zip.com.au 2005-07-14 13:57 ------- Does the attached patch fix the issue you're seeing? ------- You are receiving this mail because: ------- You are the assignee for the bug, or are watching the assignee.

Hello list, I've been looking around to solve this problem both in the IRC, the wiki and Google without any luck. I have a box running * 1.07 (with zaptel 1.07 and libpri 1.07) and I'm having massive problems with random and increasing conversations cuts both in Zap and SIP channels. When one starts a call or attend any, sometimes it cuts-off at 20-25 seconds without any apparent

Hello I assume it's possible to do this with Asterisk: To train a new worker who works remotely, I'd like to have him listen in on support calls so that he gets to learn the kind of problems that come in, and how they're solved. When a call comes in and the support person thinks it's worthy to have the trainee be part of it, he will ring the trainee so he can join the call.

Hello list, is it possible with the monitor-command to record conversations and place the soundfile in a pre-defined directory per user ?! So when extension 200 presses '*#' to record the conversation, the resulting sound file is written to his home directory on the Samba-server. This way each user has his own directory with its recordings that no one else can access (as default rights

Hello, I wanna record the conversations in a conference call, anyone know how can I do it? I've already configurated a room on meetme.conf but I don't know as I can record the conversations. I'm using SUSE 11 and Asterisk 1.6.2. Thank you so much for help me. Bye ____________________________________________________________________________________ Veja quais s?o os

Thought this might interest a few people on the asterisk list as well Cheers, Dean ________________________________ From: Dean Collins Sent: Tuesday, March 15, 2011 1:49 PM To: 'newtech-1 at meetup.com' Subject: RE: [newtech-1] Uncovering Spoken Phrases in Encrypted Voice over IP Conversations Wow, of course how stupid, when someone says hello at the start of the call

Hello, I am wondering if there is any way to support Gmail style conversation when using Dovecot? Someway of associating all inbound and sent emails..? I know that some clients let you display the emails in "conversation" which kinda does what Gmail does, however, it does not associate the Sent emails with it. Thanks for any information.

Hi, My Asterisk systems runs like a dream with mISDN, SIP and even and old Digium board. But have almost in every conversation some irritating DTMF being generated. The seems to be just as often from all trunks but are worse if noise load speaker in other end. Any good advices? Where to look for forgotten DTMF detection settings? Thank you! HB

I propose a talk "Life of a gluster client process" We will have a look at one complete life cycle of a client process which includes: * mount script and parsing of args * contacting glusterd and fetching volfile * loading and initializing the xlators * how glusterd sends updates of volume options * brick disconnection/reconnection * glusterd disconnection/reconnection * termination of