Hi all,
is it possible that asterisk uses two proxies with SRV?
The enddevices are registered on one of the two Proxies (Kamailio).
The two proxies communicate with each other.
And asterisk can choose one of this proxies with SRV.
asterisk
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Proxy1Proxy2
I have tries to
Date: Thu, 5 Sep 2013 12:11:36 -0700
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] high cpu average load
On Thu, 5 Sep 2013, Kamlesh Kumar wrote:
Running one asterisk server with below details.
Only SIP to SIP calls. No real time
On 06/09/13 09:42, Dominique Haeber wrote:
Hi all,
is it possible that asterisk uses two proxies with SRV?
The enddevices are registered on one of the two Proxies (Kamailio).
The two proxies communicate with each other.
And asterisk can choose one of this proxies with SRV.
asterisk
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|
Gareth Blades mailinglist+aster...@dns99.co.uk schrieb am Fre, 06. Sep 13:21:
No asterisk will always use the first SRV record and wont load
balance or switch to a backup if its not reachable.
hmm okay :O
What we do is have each endpoint defined in sip.conf with
qualify=yes and then in the
Consider the following scenario:
1) One or more Digium DPMA phones are plugged into the network. I know their IP
addresses and MACs.
2) The Asterisk I want to use as the telephony server starts without the DPMA
module. Therefore there are no DPMA sessions between the phones and the server.
3)
Looks like res_digium_phone will need some work for Asterisk 12...
WARNING[9372]: loader.c:561 load_dynamic_module: Error loading module
'res_digium_phone.so':
/usr/lib64/asterisk/modules/res_digium_phone.so: undefined symbol:
__ao2_container_alloc
--
All,
I'm trying to setup asterisk to run off of a raspberry pi and use the
alsa module to access the sound card (a usb logitech 270 camera/mic).
Everything appears to be working except the audio quality is very (very)
poor and there is an error from the asterisk conosle:
Loading chan_alsa.so.
I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get
iax to work.
I've opened 4569 in the EC2 Security Group.
I'm using the zoiper client. Using tcpdump I can see the zoiper packets
coming in on 4569, but nothing shows on the asterisk cli.
Frame 33: 79 bytes on wire (632
On Fri, Sep 6, 2013 at 10:41 AM, Alex Villacís Lasso
a_villa...@palosanto.com wrote:
Consider the following scenario:
1) One or more Digium DPMA phones are plugged into the network. I know their
IP addresses and MACs.
2) The Asterisk I want to use as the telephony server starts without the
Trying to figure out the best way to pull an active call out of a queue by
unique id and put it on hold. I don't want to put it on hold on the agent's
phone but I want it to be pulled away from the agent's phone and into Asterisk
limbo somewhere.
Shortly after I want to pull the same call out
On Fri, Sep 6, 2013 at 12:43 PM, George Joseph
george.jos...@fairview5.comwrote:
Looks like res_digium_phone will need some work for Asterisk 12...
WARNING[9372]: loader.c:561 load_dynamic_module: Error loading module
'res_digium_phone.so':
/usr/lib64/asterisk/modules/res_digium_phone.so:
On Fri, 6 Sep 2013, Sean Darcy wrote:
I'm not sure asterisk is even listening for the packets:
[root@asterisk ~]# netstat -apnt | grep 4569
[root@asterisk ~]#
'-t' meand TCP. IAX is UDP.
--
Thanks in advance,
-
Steve
El 06/09/13 14:44, Malcolm Davenport escribió:
Howdy,
Please forgive the off-list e-mail. I'm not subscribed to the list, I only
peruse the archives.
The follow up from George is correct. For phones that have already been
attached to DPMA, DPMA disables the enable_check_sync phone setting.
On 09/06/2013 07:08 PM, Steve Edwards wrote:
On Fri, 6 Sep 2013, Sean Darcy wrote:
I'm not sure asterisk is even listening for the packets:
[root@asterisk ~]# netstat -apnt | grep 4569
[root@asterisk ~]#
'-t' meand TCP. IAX is UDP.
My bad:
netstat -apnu | grep 4569
udp0 0
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