similar to: Realtime question

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

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

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]

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

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

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

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

Hello there! I was testing Asterisk for the last two weeks using the Realtime driver for MySQL, and leaving "rtcachefriends=yes" configured to enable MWI. Today I started making additional tests with "rtcachefriends=no" because we will probably need to use Asterisk without this cache. For some strange reason, calls stop to get routed between the SIP clients. I've

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

I've been using realtime for sip users information. I noticed that when you are doing this, if you do a 'reload' or restart asterisk, the information in a 'sip show peers' goes away. When I do this, MWI stops working. I always though MWI used the astdb file ('database show') to determine where to send MWI but it must be using 'sip show peers' because when this

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

Can anyone shed some light on realtime caching? My desired behavior is that MWI works with realtime voicemail/sip/extensions AND updates to the database take place on the next call to the extensions. Right now I have rtcachefriends=yes, and MWI works, but updates to the database for a cached user seem to still require a reload. It is my understating that removing rtcachefriends will

hi folks, I am using asterisk 1.2.13 (debian etch). My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old

Hi, I use SIP-Realtime to store my SIP-users and I keep the informations about the SIP-Providers my Asterisk registers to in sip.conf. I'm running into the following problem. If I set rtcachefriends="yes" because I want to use MWI and run a "sip reload" because I changed something in sip.conf, Asterisk forgets about all registrations of the users which are all unavailable

Arrgh. I just made a call with Asterisk to extension 2944093. That extension exists in astdb and I have rtcachefriends=yes in sip.conf. Asterisk did a database query... SELECT * FROM ast_sip_users WHERE name = '2944093' Uhm... Why? Doug