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

[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 don't show busy.. there will be no announcements. However, I see no references to such an issue in any upgrade documents I have found. Any one have a tip?...

...ystem at the far end. With SIP it is sometimes "chancy". Is there a plain 64K codec that would simply pass through the SIP server and be handed off to a PRI or phone co. trunk on a T1 on the other side of the SIP server? Digital 64K telco sounds very good as a phone conversation. Cary Fitch

...h the member busy, we get no voice announcements. (For test purposes is being on hold "busy"? We have also just laid the phone on the desk.) We will settle for expected hold time, Thank you announcements, Position in queue, or Dow Jones 30 Industrials news. :-) Anyone have a tip? Cary Fitch Here are some relevant errors from console. [Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12406 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '495f59f65dec70c67849cf4f0cd098ec at 72.49.176.4'. Giving up. -- SIP/3617001000-009a3930 is circuit-busy...

...#39;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. Basically, a call from their own number would be the clue that there is a voicemail waiting. Thanks Cary Fitch

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

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

...witch" for subscribers on standard telco service loops. This isn't a "How many lines can I handle using a Belchfire 2600 processor?" type question but a request for pointers to big time systems. There would be no IP path to the end user, "just" copper. Thank you Cary Fitch

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,

...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

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:

...SIP' (cause 20 - Unknown) What is the (likely) cause of the above errors? It happens with little channel usage at the time. I understand that the peers were not reachable, is the dial exec full message Asterisk's message that it couldn't communicate with those peers? Thanks, Cary Fitch

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

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

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

.... 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" seems futile. Thanks! Cary Fitch

_____ 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...

This sort of message is usually preceded by some magic tones that allow direct marketing application to immediately drop a call to a dead phone number. What is the proper terminology for the tones? Where can I find information about how this is implemented? -- Drew Einhorn