search for: rschall

Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any

Hey all, I had a user complaining of calls which were dropping mid-conversation. I looked into the time of one of the calls, and saw the following: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660]

So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone

I'm trying to duplicate a behavior we had with our old avaya system, and I've come across Auto Answer (Ring Answer). However, its not quite the same yet. Right now, when I dial **5053, it will add the SIP header for Ring Answer and it will call 5053. The phone auto pickups just fine. However, we need that call to be muted. If you were to call into a meeting, we wouldn't want them to

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

...hich I believe is a "Normal Hangup", but could be wrong. Any thoughts as to what could cause this? -- Accepting call from '3126290600' to '3125727758' on channel 0/1, span 2 -- Executing [3125727758 at from-pstn:1] Macro("Zap/25-1", "faxreceive|7758|rschall at callone.net") in new stack -- Executing [s at macro-faxreceive:1] Answer("Zap/25-1", "") in new stack -- Executing [s at macro-faxreceive:2] Set("Zap/25-1", "FAXFILE=/var/spool/asterisk/fax/7758_3126290600_1207329420.10.tif") in new stack...

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

Hello All, Hoping all of you might have an additional option for me to try at this point. :) My Goal: To have a paging option that does the following.... When I press **_XXXX it will send a ring-answer page to that person. The person on the other end should be muted, so if they are in a conference, you can't hear what is going on in the meeting. If that person hears me and decides they want

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching

Hoping someone out there has run into this or has some ideas for us. We currently have asterisk set up with Realtime (using mysql) for its extensions,sip and voicemail files. The problem we are trying to solve, is one of a failover mechanism. What if our mysql server went down. Can Realtime be set up with a secondary mysql server to get its data from. We can set up mysql to sync with its fellow

We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars

We're trying to get a SIP peer going between our asterisk box and our provider. It should then ring our phone. The call does come in and it does execute the extension in the dial plan. But the provider says they never get a 200 OK back and therefore they send another INVITE and then after a few seconds drop the call. Here's our setup: sip.conf [ngt-trunk] type=peer qualify=yes port=5060

<Correction in my zapata.conf file I used> Hey Everyone, So this is a problem I've been having for sometime now. I sent a few messages to the list with no luck. The problem is that when people dial into the Asterisk system using DID numbers, it works the first time or 2, then I get busy signals. A friend recommended I clear out the zapata and zaptel, start over, and recreate my

This might sound like an odd question.... but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But... our old phone system set the caller id on the senders phone to show who they called. Example... If Sally calls Jim, then

A question about Queues and Dial Plans.... We are trying to set up a customer service queue. I've set up the queue and agents who will participate. However, there's still one area I'm not sure how to make it work. After 60 seconds, I need it to decide that no one is available, and forward it to an email box of my choosing. Is this possible? Rob

We just had the oddest thing happen which worries us as new users. We had 3 calls running on asterisk (one from sip to sip and the other to sip to zap). It seemed for no reason, the connections just dropped and the lines went dead. You couldn't call a phone (not even yourself). Once I restarted asterisk (it gracefully shutdown and started back up), then everything worked again. This concerns

Hey all, This should be an easy one. I have a few different queues and wanted to set up a standard macro to handle them, so I can shrink the dial plan down and stop having so much redundancy. But when I try to use it, i get a "no answer". Here's what does work (non macro): exten => 5054,1,Answer() exten => 5054,n,Ringing() exten => 5054,n,Wait(2) exten =>

I wanted to try and see if I could get my Hawkings Net-Talk USB phone to work with our asterisk setup via yakaphone. Has anyone ever tried this? It sees the mic and speakers, but if we could get the keypad to talk with yaka and in turn with asterisk, that would be really nice. If there are any other recommendations for a VOIP USB phone that you could plug into a windows and/or linux machine and

This should be easy, but I can't find the right wildcard. Right now I have.... exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW) for international and for local exten => _9NXXXXXX,n,Dial(${TRUNK}/${EXTEN:1},,wW) The problem is if the call isn't typed in, then you press dial, we have problems... Example: I pick up the handset and get a dialtone. I press 9011331234567 or