similar to: Asterisk with Realtime registration problem

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

I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all

All, we're running realtime and ast 1.2.7.1 stable. The problem I'm having is when you register a SIP device on ServerA, works fine, sip show peer works fine. When you dial the SIP device from ServerB however, it tries to dial, even does a MySQL query like it should but comes back saying no route to dest. When I do a 'sip show peer ..' on ServerB, the peer comes back fine, it even

Hi guys, im using realtime and I want to show registered users or online users on a webpage and offline users. Im taking regseconds field to make this happend If regseconds value is 0 then user appers offline, it regseconds is something else then its online, but sometimes this works and sometimes it does not. Im using the following options rtcachefriends=yes rtnoupdate=yes

If your phones are setup to connect to the asterisk box by name, then a smart DNS server can just point phones to the backup box after failure. However, since asterisk running on the backup box doesn't know about the phones, this is only half the solution ________________________________ From: Mohamed A. Gombolaty [mailto:mgombolaty@noorgroup.net] Sent: Thursday, June 30, 2005 8:30 AM To:

I haven't been able to find this on the wiki: If rtcachefriends=yes. When will a change to a realtime user/peer take effect? Next registration? Never? It's also not clear to me what the purposes of rtautoclear and ignoreregexpire are. The only info I have found is the comments in the sample config file. Sounds like rtautoclear will save memory if I have lots of peers. Is there any

Dear All, I have downloaded the xlite version 2.0 for windows and I made the following conf in the xlite itself as the document suggested in order to make it work with Asterisk but still it doesn't work as a matter of fact when I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server at all, here is the my conf on the xlite itself: in the

Hi, Is there something wrong with REALTIME (ARA) when used with rtcachefriends parameter? In my sip.conf (Asterisk 1.2.0): rtcachefriends=yes rtupdate=yes rtautoclear=yes Desired configuration is realtime configuration (via odbc) for SIP phones + MWI. Realtime means the following: when I make changes to db they should apply with no extra commands executed in CLI. In order to use MWI with

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've

Questions: Does anyone have a really STABLE asterisk system running about one year without need to restart the service or the SERVER ? Does anyone have a production Call Centre saled that don't lockup and is stable for 6 months ? I'm asking this questions because we have choose Asterisk for our call centre solution but, since the bugtracker only grows and people still want to stuck more

Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is

Hi, I'm looking for a reliable provider to use mainly for outgoing calls in the UK, incoming isn't so much of a worry as I think I'm going to accept them over ISDN. Cheers! Steve -- Steve Foy steve@narnian.org

Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and

I love TrixBox, with the custom config files you can tweak pretty much with TrixBox too, I have at least done some. Plan to do a plain Asterisk install later, but for now I learn a lot about the config files just with TrixBox. Some things might be a bit harder with TrixBox due to some of the premade dial plans, but can get it to work :-) _____ Fra: asterisk-users-bounces@lists.digium.com

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br> <br> Using Asterisk 1.4.25.1<br> Using realtime sip_buddies<br> <br> I notice

[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'gerrie' Am I correct that when I turn on rtcachefriends in sip.conf, database-changes in my MySQL-DB will not be reflected untill a reload ?? Am I correct that when I turn off qualify in my realtime

Hello, Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1 with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2. Libpri and dahdi is only for dahdi dummy cause of the meetme function. After the upgrade we had the problem that some Linksys spa941 phone at one location could not dial out. incoming calls to the phones works without any problem, but outbound the

Anyone know the command that tells * to load a sipfriend from the realtime db rather than saying no such host? I've tried various combinations of the rt commands: rtcachefriends=yes; ;rtcache=yes ;rtAutoClear=yes ;rtautoreg=yes ;rtIgnoreRegExpire=yes ;rtupdate=yes rtfromcontact=yes Basically I have a group of 4 * servers all routing calls, but only two are hearing the phones

We have several Asterisk servers (1.6.2.15) all configured for Realtime, all backed by the same database. The Asterisk servers are all listed under DNS SRV records, and SIP ATAs find us this way. Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as "regseconds", "lastms", "ipadr", etc. However, with

Dear All, I was trying to enable call forwarding, following the steps of the link on voip.org regarding this issue it doesn't work and the phone I am trying to implement on is still ringing. below is my conf in extensions.conf and the CLI output during the process. My configuration is : exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2}) exten => _*5X.,2,Hangup exten =>