similar to: Grandstream 2010

[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcement problem, because if all members

_____ From: Cary Fitch [mailto:caryf at usawide.net] Sent: Tuesday, April 14, 2009 1:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk-beginner : cannot make phone calls using Asterisk May I suggest divide and conquer? I haven't followed every detail, but it seems that your phones are not registering. Put them on

We are an asterisk user... small time system 50-100 users or so. But, we have an opportunity to get into a big time telecom activity. It would have 2000 to 30,000 user lines per city, and we would like to have those brought back to a central location for control and because transport can be more economical than remote site rentals, maintenance and personnel. We could take the local lines into

I am trying to get a queue to do more than just play music and hold calls. Specifically, making some "comforting" voice announcements would be nice. Below is the queues.conf file relevant portions. Member phone number is munged to protect the guilty. We shouldn't need the announcement source info, but I have been trying everything. The problem is with the member busy, we get no

I am not sure what the problems are and the reasons for the basic 64K modems used in VOIP are. I understand the compressed codecs that get the bandwidth down to 20-30 K. And perhaps the 64K units give much better potential audio than you would get on a normal POTS line. But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old phones. Multiple transcodings cause issues.

Is there a way to program an FXO device to totally ignore incoming calls? I want to put an FXO on a Fax line so that 911 calls can be sent via that line, but all other activity on the line is between the Fax machine and the phone company. Perhaps munge the ring tone detect if nothing else? Cary

I don't want to start the "How many calls can Asterisk handle?" discussion or "How many angels can stand on the point of a pin?" discussion either. But can anyone contribute some practical knowledge of systems that take in channel bank T1s or DS3s from "far away", and process the calls? I am looking for real world, been there, done that, or "check the

We get this error message [Mar 23 10:10:09] WARNING[4325]: chan_iax2.c:1056 __send_lagrq: I was supposed to send a LAGRQ with callno 14034, but no such call exists (and I cannot remove lagid, either). -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'IAX2/brandx-14819' == Spawn extension (usawide, 3, 2) exited non-zero on 'DAHDI/1-1' -- Hungup

Hi, I have a customer running a 120 second long WAV file on their MoH. The problem is that it's always starting from the beginning, so people being put on hold, talked to, put on hold again, etc always hear the first 10-15 seconds. Is there a way to have Asterisk MoH remember where it left off? Or at the very least just play the same stream to all people using the same MoH class, so

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

Has anyone here used T1s with RBS with asterisk? Cary Fitch

After passing authentication, Then with this line, extent => 3616739999,5,DISA(no-password calls-outbound) As soon as the first digit of the intended number to be called is entered, the system does a Hungup 'DAHDI/1-1' It has done that no matter what I have tried. I am missing the boat somewhere. Anyone have tips? Cary Fitch

Can any one suggest a little code to either ring a cell phone when a new VM message is recorded, or send a text message? Basically outside sales people want to know they have a new message, but don't want to be interrupted to take a forwarded call. While a message by message notice would be nice, even just a single notice on the first message would be an alert to call for messages.

When Asterisk sends a call to "a phone company" via a PRI/Dahdi, does it actually send CALLERID(ANI), or only CALLERID(NUM)? Cary Fitch

Does anyone have any light to shed on: c_avpair_new: unknown attribute sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597 We are getting congestion errors on a Pri to telco, and not sure what is going on. Thanks Cary Fitch

In a 1.4.24 system, out of several lines, one of ours gets 1 or more random calls a day with "Asterisk" as the caller ID. I have just seen this described in the last couple of weeks, but at the time it wasn't happening to us, and I the explanation didn't stick with me. Can anyone give me a pointer to this "feature"? Searching the message base for "Asterisk"

I need a country, state, city database for a web application. Anyone have a free version they can email (or drop.io) for me? Looking for something like this at $197 but may as well ask in case you know of a free source. http://www.globixdata.com/pop.cfm?db=world&v1=l&v2=s&v3=a&pricing=99 Regards, Dean Collins Cognation Inc dean at cognation.net

[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer '3516533812' is now UNREACHABLE! Last qualify: 86 [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: Peer '3516533812' is now Reachable. (98ms / 2000ms) [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Only 100? We had a single server over 300. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve

Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as this will be some kind of a "hosted pbx". Given he finds a provider wich has this much SIP