All,
I upgraded to asterisk 13.10. I have minimal load on the box. 20-30 calls a day.
Right now, there are no calls on the box at all.
top shows me this:
PR 20
NI 0
VIRT 1570540
RES 84620
SHR 26296
S S
%CPU 99.7
%MEM 8.4
TIME+ 3468:39
COMMAND asterisk
When I run this command
while
Hi all,
I was using 13.5 but upgraded today to 13.9 (13.10 came out a few hours
after I upgraded).
On both 13.5 and 13.9 asterisk seems to use 100% of the CPU. This usually
happens a few hours after starting asterisk. A restart of asterisk gets the
CPU back down, but only for a little while.
I hope so! Snom just added opus support in their latest firmware if that
counts for anything.
Hope digium figure it out.
Tzafrir, does your update support pass through only or transcoding too?
Thanks all,
Chirag
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Hi all,
Sorry if this has been asked before. I searched a lot, but found
conflicting answers, so hoping for some clarification.
My question is does Asterisk 13 support OPUS? If so which version exactly?
If asterisk 13 requires a patch, which is the correct one and where do I
get it?
Kind
You were right. I had non-default rtp ports open in iptables. Edited
rtp.conf et voila. Everything seems to be working.
Thanks so much for your patience and guidance!
Have a lovely eening.
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-- Bandwidth and Colocation
So I see:
EXTERNAL_SNOM_IP -> EXTERNAL_ASTERISK_IP (UDP, length 218, src: 60798,
dst 11128)
EXTERNAL_ASTERISK_IP -> INTERNAL_SNOM_IP (UDP, length 218, src: 11128 dst
60478
So i see udp from the phone, but there's no audio.
I do also see some packets ::
EXTERNAL_ASTERISK_IP ->
In the PCAP I can see asterisk sending UDP packets to my local IP
192.168.0.5
It's funny, when I switch to TCP on 5060 audio seems to work fine. The
moment I go to 5063 on TLS everything goes a bit awry. Any further input is
greatly appreciated.
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I'm dialling from the snom and every few calls asterisk sends media to the
phones external IP and it works!
And then now and again it sends the media to the phones internal IP and I
hear nothing. I'm really at a loss.
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> Joshua Colp wrote:
>>
>> Have you done a packet capture to see if the RTP from the remote device
>> is hitting the machine to narrow things down?
>>
>>
>>
Nope. When I run with RTP encryption on it seems that rewrite_contact does
not work in PJSIP.
When I turn off RTP some calls get media, some
> Joshua Colp wrote:
>
> There should be nothing different, except for how you configure things.
> What is the full PJSIP configuration? What is the environment where
> Asterisk is running? Is ICE actually in use on the other side? What is
> the full SIP trace?
>
The full configuration is here:
I have two accounts on Asterisk 13. One uses chan_sip and the other pjsip.
In my snom 760 the setup for these two accounts is identical.
When I call echo test from the account using chan_sip audio comes through
fine.
When I call echo test from the account using pjsip there is no audio.
With
Hi George,
I tried the nightly build and also Bria. I can replicate the same issue on both.
This morning I made many successful calls in succession. This evening
it was intermittent again.
Could it be the mobile network is blocking the RTP but it seems odd it
works sometimes and not others.
Hi,
I have a PJSIP account configured as below. I am testing with the Echo Test
application on Asterisk 13 and using CSipSimple.
I can create a call with TLS and SRTP, however for some reason only 1 in
every 5 calls has audio.
When I connect over WiFi, I have audio every single time. When I
Hi all,
I'm using Kamailio + Asterisk 13 (PJSIP), where Kamailio (using rtpengine)
acts as the registrar and forwards all calls to Asterisk.
This works fine when using udp / tcp and RTP. When switching to TLS/SRTP,
the call is set up correctly, however, I get no audio.
When I skip kamailio and
From: Matthew Jordan mjor...@digium.com
If the INVITE request is not shown in the CLI with 'pjsip set logger
on', then Asterisk is not actually receiving the request.
Does a pcap show the message being sent to the correct IP/port? If you
change the transports to bind to port 5060, does
From: Matthew Jordan mjor...@digium.com
If the INVITE request is not shown in the CLI with 'pjsip set logger
on', then Asterisk is not actually receiving the request.
Does a pcap show the message being sent to the correct IP/port? If you
change the transports to bind to port 5060,
OK, it stopped working.
It turns out the transport and endpoints in PJSIP are ok. I can send an
invite from my unregistered snom phone and I can see some activity in the
CLI.
However, when I dial from my snom to Kamailio and have it pass the message
to asterisk, PJSIP seems to ignore the sip
Joshua Colp wrote:
Have you configured any transports? PJSIP does not create any by
default, you have to explicitly configure them. Without them no traffic
can come in or go out. You can also remove the explicit transport from
the endpoint.
Yes I have two transports
[transport-udp]
Hi,
I want to have Kamailio in front of one or more Asterisk boxes.
I don't think it is necessary for Kamailio and Asterisk to register with
one another. I'd like for PJSIP to recognise Kamailio by its IP address.
I have two boxes, both have public IP addresses, they also have private IP
Chirag Desai wrote:
I've tried explicitly setting the IP in bind and leaving it as above.
Nothing seems to come into asterisk. Although, as mentioned I can see the
SIP messages when I ngrep 5061.
I got it working, I can see the sip traffic in the CLI now.
I was trying to match on the IP
Hi all,
I have Asterisk 13 running and I'm currently trying to get PJSIP working on
TCP.
My transport looks like this. My box is not behind NAT.
[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5061
My endpoint looks like this:
[user1]
type=endpoint
transport=transport-tcp
Joshua Colp wrote:
Chirag Desai wrote:
* Joshua Colp wrote:
* * snip
* * Remove transport=transport-tcp from your endpoints.
* * Joshua...I did that but now my endpoints won't register.
*
That should have no impact on things. Can you clarify what you mean by
it doesn't register? What
Joshua Colp wrote:
snip
Remove transport=transport-tcp from your endpoints.
Joshua...I did that but now my endpoints won't register.
Kind Regards,
Chirag
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-- Bandwidth and Colocation Provided by
Hi all,
I saw Matt Jordan's recent Kamailio world talk and was interested in the
idea he proposed of stripping out authentication and registration from
asterisk and letting Kamailio handle it.
All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding
registrations to asterisk.
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