similar to: Transfer Caller ID

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

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

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

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

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]

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

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

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

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'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

Hi all, Here's our setup: Asterisk 1.4.18 Agx-ast-addons 1.4.5 Problem: When accepting a fax, the fax itself comes through just fine, and it does successfully create a tiff file. However, the dialplan should be executing a system command right after that completes, but isn't due to hanging up early. I'm getting a cause 16 hangup, which I believe is a "Normal Hangup", but

Hi Everybody I've a requirement for one of my operators for an autodialler for which i plan to deploy asterisk (I already have 3 asterisk servers on PRI running very well ! ). The scene is like : Asterisk will call a customer and play a prompt that prompts him to press 1 if he wishes to talk to an agent , If the customer presses 1 then the call gets connected to one of my proffessional agents

Hi, I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When calls come in on our POTS lines, the caller id shows up like "555-555-1234 at 192.168.1.10" where 555-555-1234 is the correct phone number and 192.168.1.10 is my pbx server IP. This format does not work for redialing on outbound calls. While there may be an outbound dialing change that could be made, it

Hi, I have been looking for an example of accomplishing this, but I've been unable to locate something similar to what I'm trying to do. Here's the scenario: Users caller ID is set to their internal extension (200-250). This is set in sip.conf for each user. Each user has a local DID as well (hosted through Vitelity, for example (555)111-2222). The problem is that this extension was

Hi Guus, hi all, please find attached a proposed feature implementation for tinc. As mentioned in http://www.tinc-vpn.org/pipermail/tinc/2010-May/002344.html , my goal was to connect nodes on the same LAN using their local (LAN) endpoints. I've implemented a multicast sender, which announces its own endpoint on every connected interface regularly. All nodes receiving multicast

A simple question but one I can't seem to find easily... I have 90 or so DIDs. For all outbound calls, I edit the callerid so that it will always read out main line's number. This poses a problem though, because the CDR detail isn't written until after everything is done. So when you look at the detail, the number you are calling is over written with this new caller id for some

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or

I'm sure this has been discussed many times, but I have a question about hoteling. My understanding would be this: A phone sitting on a desk. A user hits 9000 and it asks what extension you'd like to become. You type "1001" and then it asks for your password. You type 1234, and it says you're "logged in". You now are accepting calls at your phone and you're