Displaying 20 results from an estimated 30000 matches similar to: "Ok... what is 'sip show peers' really used for?"
2006 Mar 21
4
Realtime SIP Persistency
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
2006 Mar 21
3
Realtime / SIP Peers etc
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
2005 Aug 16
6
realtime caching
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
2009 Aug 25
1
Realtime with "rtcachefriends=no" problems...
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
2011 Nov 23
1
MWI for non-subscribed Realtime peers?
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
2006 Mar 22
2
Realtime Query
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
2006 Jun 28
4
Realtime SIP Registrations
Has anyone considered the idea of splitting the sip registration
information in a realtime database from the actual configuration of the
peers?
I mean, instead of having a table full of the configuration information
(i.e. name, regexten, secret, etc) and registration information (i.e.
ipaddr, fullcontact, etc), you have separate tables with their own
information. This way, you can have separate
2006 Mar 21
12
Fw: anybody has SIP realtime working ?
Hello,
I am just asking this because I am note sure if the problem
is on my side or not, I saw some comments on SIP realtime
today so I was wondering, has anybody has SIP realtime working
with a softfone ?
If yes, please confirm, that would give me a light.
My previous message to the list is below.
Thanks.
Frederic
----- Original Message -----
From: Frederic Jean
To:
2015 Sep 16
4
Realtime Voicemail MWI
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
2010 Oct 11
1
MWI Assistance
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
2005 Aug 05
1
Asterisk MWI and Realtime
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
2006 Dec 04
1
mwi for voicemail not showing up for realtime config.
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
2006 Mar 30
5
Reload astdb?
Is there any way to get Asterisk to reload the /var/lib/asterisk/astdb file?
It seems to only read it on startup.
Thanks.
2005 Jun 06
1
NAT & RealTime
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2005 Feb 08
12
SRV lookups
Hi everyone,
I have a question concerning DNS SRV lookups. The situation is like this:
- one central Asterisk server
- many domains with SRV records, let's say we have bar.com and doe.com
Now the question is: if the SRV lookup is done for foo@bar.com the call is
mapped to foo@myasterisk.mydomain.net. Is that correct?
If so, I have a problem: if somebody calls foo@bar.com, Asterisk
2014 Sep 08
1
Asterisk removes ice lines in sdp when calling between webrtc clients
Hello,
I have a problem with a call between 2 webrtc clients. Asterisk removes the
ice-related lines from the sdp when it sends the INVITE out, and the called
webrtc client rejects the INVITE due to the missing ice lines. Both webrtc
clients are defined exactly the same way, same values in all fields except
the number of the peer.
There's probably something I've changed that causes this
2005 Jul 28
1
realtime: sip show users/peers
I don't see anything with sip show users and sip show peers, however it
works!
Is there a trick?
I have installed realtime (sipbuddies) on one machine and see sip show
peers/users and on my new installed system I don't.
Have I forgotten something?
bye
Ronald
2014 Apr 24
1
Realtime integration: Unregistered clients showing as registered?
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.
In Asterisk cli sip show peers shows nothing but for example realtime load
sipusers name 660 shows the
2010 Jul 21
1
asterisk realtime SIP configuration
Hi All,
I am trying to configure asterisk realtime. But i am unable to get the
extensions listed successfully when i type "sip show peers" in the asterisk
CLI . i am unable to see any failure logs when i do a reload
i can able to connect to the data source through "odbc show" in the
CLI, Any hep in this regard is highly appreciated. Following is the
configuration
2007 Dec 05
1
SIP-Realtime and sip reload
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