similar to: Realtime SIP Persistency

I'd love to understand what the function of the peer list returned by 'sip show peers' is for, especially when Realtime is used. If I start Asterisk with realtime enabled, a 'sip show peers' yields none. As each peer (phone) registers, or a call is made to the peer, Asterisk adds them to the list returned by 'sip show peers'. Correct? Apparently Asterisk doesn't

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

Ready to scream here.. 1. After 6 months with Asterisk I'm STILL trying to understand the difference between a SIP user, friend and peer. 2. Exactly what resource does Asterisk use to send MWI to registered phones? I thought it was astdb? 3. It looks like it isn't astdb. It looks like it will only send MWI to a phone if it shows up in 'sip show peers'. 4. WHY then does a reload

Hi, I have an Asterisk behind an OpenSIPS proxy. The proxy handles registrations and also SIP SUBSCRIBE for MWI. The Asterisk are configured to send NOTIFY to the proxy even when the SUBSCRIBE haven't been received. I can configure a user in sip.conf that works: [az5134939706] type=friend host=xxx.xxx.xxx.xxx (IP of proxy) port=5060 nat=no mailbox=1234 at customer subscribemwi=no

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

I'm testing my asterisk system and the realtime backend. My Asterisk build is rather aged, 03/18/2005 CVS. I have successfully moved Sip peers and Voicemail boxes to the realtime database backend and this works very well except for MWI. I don't seem to be able to get MWI to work when I store the voicemail information in a database backend, from a flat file it does work fine. I'm using

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

Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the

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

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2593 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050606/5878c98a/attachment.jpeg

Greetings All, Regarding this archived post. http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html Did anyone ever find an solution to this? I've got a new box running 13.3.0 with the exact same issue. For those that don't read the link. I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, These are loaded into asterisk without

Hello, I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for either. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for OpenSIPS or Kamailio with the following features: (a) Support an arbitrarily large number of

Hi, I'm struggling to get the MWI set up on a few Polycom phones. The setup is like this. I've got a few phones in the context called [company2_phones] and I've got a few mailboxes in the voicemail context [company2]. Therefore, for each entry in sip.conf (i'm actually using sip realtime if that makes a difference), i've entered "mailbox=1 at company2" (1 being

With rtcachefriends = yes in sip.conf, my SIP phone registered to Asterisk Server A cannot dial another SIP phone registered to Asterisk Server B. The error message is: "Cannot create channel of type SIP (Cause 3 - no route to destination)". The two phones _can_ call each other if I set rtcachefriends = no. The common extensions.conf simply uses Dial(SIP/extension) to dial extensions.

Hi folks, i'm trying to configure my extensions.conf as small as posible and for that reason i'm using macros. The problem is that maybe I have a misunderstood the concept for the directive "mailbox" in sip.conf. Under my knowledge configuring the mailbox directive to the mailbox I want would be enought to leave an retreive messages in that voicemail box. Of course it seems to

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 someone else confirm that your phone does not recieve MWIs when using SIP and RealTime? Is this a problem with SIP or with Voicemail? -Matthew

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

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

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