similar to: MoH - always starting from the beginning

[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

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 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 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

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

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.

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 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

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

Hi, We are trying to implement a complex business logic in Asterisk. Executing "Wait_For_Digit" command after playing IVR. We want to stop the IVR once we receive the digit. It is not recognizing the Digit until it completes the IVR. How can we stop the IVR once we receive the digit? Thanks BB -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone here used T1s with RBS with asterisk? 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.

Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural "monopoly". From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them,

I was copying tracks from CD into mp3 files so that I could use it in Asterisk 1.4.21.2 MOH. (BTW, I have already secured proper license to play MOH to callers.) I used MS Media Player version 11 and rip it at 128kbps (smallest) but whenever I listen to MOH, I saw the following message on the Asterisk console. WARNING[20829]: mp3/interface.c:215 decodeMP3: Junk at the beginning of frame 49443303

Hello, I am having a problem with my current SIP over VPN setup. We have a server running asterisk at our office. All the phones in the office are on the same network / local to this server. We also have two employees with home offices using SIP phones over VPN to connect to the asterisk server. These phones have no problem with calls to the phones in the office, however there is no audio

I have two questions for the group. #1 - I'm looking to use some GSM SIM cards with my Asterisk PBX. Can anyone recommend a gateway? I need about 10-15 SIM slots. #2 - I'm also looking to connect Asterisk to an SS7 signaled DS1 (24 channels) for inbound and outbound voice calls. Can anyone offer any suggestions for cards to use there?

Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear

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

Hello All, Does anyone know if you can start an MOH queue on an individual call? What I mean is, for example if you have a script that you want the moh to start with certain phrases, can it be done, or are you limited to the standard looping audio? It's almost like starting a stream for each call, and terminating it when the call comes off of hold. Regards, Greg