From what I can determine while troubleshooting a voice-dropping
issue, the Asterisk server in my organization has been dropping RTP
packets between the asterisk server process and the network interface.
I determined this from an RTP debug that showed packets sent to the
phone and packets
Hi All,
Here's a funny bit of a problem. I've got an asterisk server which appears
not to be sending any RTP out of the system. Any ideas why such a weird issue
would arise?
I've tested this scenario via several termination gateways with SIP, and
always
there was no RTP in either
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Hash: SHA1
Nir Simionovich wrote:
Hi All,
Here's a funny bit of a problem. I've got an asterisk server which appears
not to be sending any RTP out of the system. Any ideas why such a weird issue
would arise?
I've tested this scenario via several
fairly baffled.
Nir s
- Original Message -
From: Matt Riddell (IT) [EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, September 3, 2006 1:02:18 PM GMT-0800
Subject: Re: [asterisk-users] Asterisk
Title: RE: [asterisk-users] Asterisk not sending RTP
I've now also enabled RTP debugging, and noticed that Asterisk doesn't send out RTP at all.
All the lines appear as the following:
Got RTP packet from 62.219.61.73:59436 (type 0, seq 1243, ts -1997588432, len 80)
Got RTP packet from
Nir Simionovich wrote:
Any ideas anyone ?
Do you have a compatible codec?
What does the SDP show?
Is sip.conf binding to a valid IP address?
Jeremy McNamara
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Nir Simionovich wrote:
I've now also enabled RTP debugging, and noticed that Asterisk doesn't
send out RTP at all.
All the lines appear as the following:
Got RTP packet from 62.219.61.73:59436 (type 0, seq 1243, ts -1997588432,
len 80)
What
Title: RE: [asterisk-users] Asterisk not sending RTP
well, here is the full SIP debug:
Sep 3 10:05:59 DEBUG[6139] manager.c: Manager received command 'Originate'
Sep 3 10:05:59 DEBUG[6139] chan_sip.c: Setting NAT on RTP to 0
Sep 3 10:05:59 DEBUG[6139] chan_sip.c: Outgoing Call
Nir Simionovich wrote:
Sep 3 10:06:00 VERBOSE[6124] logger.c: Capabilities: us - 0xc
(ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), c
Very simple, there is no codec being sent from the peer, thus the near
end wouldn't be sending RTP.
Jeremy McNamara
]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, September 3, 2006 4:45:08 PM GMT+0200
Subject: Re: [asterisk-users] Asterisk not sending RTP
Nir Simionovich wrote:
Sep 3 10:06:00 VERBOSE[6124] logger.c: Capabilities: us
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