similar to: Big practical systems

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

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

[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 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.

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

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

I have a customer installation with an Adtran DS3 mux. The DS1's go into my Asterisk servers that run IVR/Call recorders. The DS3 provider is Qwest, and they tell me that they routinely drop the DS3 service to redundant back-up's and that this is a common practice that happens thousands of times to DS3 lines daily across the US without any service interruptions. They say that the

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

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

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

Any know what Digium hasn't released the DS3 card? It was supposed to be out a while ago. -Matt

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,

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, 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:

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