>> check the system and make sure there really is no firewall like I said
> You were right.
Stick around on the list long enough and you'll realise the truth... he always
is ;-)
Pete
--
_
-- Bandwidth and Colocation
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
Chirag Desai wrote:
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.
If "rtp set debug on" shows no packets
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 ->
Chirag Desai wrote:
In the PCAP I can see asterisk sending UDP packets to my local IP
192.168.0.5
If you don't see anything arriving from the remote side and we've told
them the right IP address and ICE is not actually negotiated... then
that leans more towards something remote unless
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.
--
Chirag Desai wrote:
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.
In the non-working case check the IP address in the
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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--
Chirag Desai wrote:
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
> 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:
Chirag Desai wrote:
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
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
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