[asterisk-users] No Audio with SIP to only one
provider whenswitching servers
Hadar Pedhazur
hadar at unorthodox.com
Fri Apr 27 07:10:35 MST 2007
Brad Sumrall wrote:
> I would not rule your firewall out as the problem!
> Port 5060 is only the authentication port, the rtp stream is normally 10,000
> thru 20,000.
> Some of your phone may have STUN modules on them.
>
> Open 10,000 thru 20,000 and 5060 on the firewall.
> Stick some holes in it for testing purposes.
> Verify ports are open with telnet:port number "both ways", telnet is your
> friend.
> If it works, close the holes up and consult your firewall docs
>
> Brad
Thanks for the response Brad (and Brian Capouch as well in a
separate note!).
I was offline all day yesterday, but I can do more testing today.
Of course, it's quite possible that it's the firewall. That said,
all other providers (including SIP) work, so it would have to be a
reasonably tight number of ports that are open to the other
providers, and a different set of ports that are closed that
StanaPhone is trying to communicate on.
Anyway, more testing on the way ;-)
BTW, I run Shorewall (which is a cover for IPTABLES), and it
usually logs every dropped packet, and I see _no_ rejections in
the log file for source IP from StanaPhone and destination UDP
ports on my machine. I'm running the same Shorewall "rules"
(different version of Shorewall and different OS on the two linux
boxes) on the box that works with StanaPhone...
Thanks again to both of you for the responses!
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Hadar Pedhazur
> Sent: Wednesday, April 25, 2007 6:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] No Audio with SIP to only one provider
> whenswitching servers
>
> I have been running Asterisk for years on a machine with a public
> IP. Most recently, I have been running 1.2.17, from the day it
> came out, with no (noticeable) problems.
>
> Yesterday, I switched over to a new server that is on the same
> public subnet, one higher than the original server.
>
> I built 1.2.17 from source on that machine (as I did on the old
> server). My firewall on the new machine is configured identically
> to the old one as well.
>
> All of my IAX connections just worked. All but one of my SIP
> connections just worked as well (which is why I can't believe it's
> a firewall issue).
>
> StanaPhone, which I use for 2 incoming DIDs, registers correctly,
> and rings my phones correctly when a call comes in. However, once
> answered, there is dead silence in both directions, on 100% of the
> calls.
>
> There isn't any problem on StanaPhone's side (which has provided a
> _fantastic_ service ever since I signed up!), because I can
> connect to them with X-Lite and receive calls with audio. More
> importantly, if I fire up Asterisk on the old server, it still
> works!!! I can connect with X-Lite to the new server, so the new
> server definitely accepts SIP connections, and audio works.
>
> It's _not_ a codec problem. I verified that on both the working
> and non-working servers the connection is established with ulaw on
> both sides.
>
> I have dumped the "peer" and the "channel" on both, while the call
> was active, and they look identical to me, except for the random
> bits associated with a particular connection. Here are the ones
> from the machine that fails:
>
> *CLI> sip show peer XXXXXXXXXX
>
>
> * Name : XXXXXXXXXX
> Secret : <Set>
> MD5Secret : <Not set>
> Context : default
> Subscr.Cont. : <Not set>
> Language :
> AMA flags : Unknown
> CallingPres : Presentation Allowed, Not Screened
> Callgroup :
> Pickupgroup :
> Mailbox :
> VM Extension : asterisk
> LastMsgsSent : 32767/65535
> Call limit : 0
> Dynamic : No
> Callerid : "" <>
> Expire : -1
> Insecure : port,invite
> Nat : RFC3581
> ACL : No
> CanReinvite : No
> PromiscRedir : No
> User=Phone : No
> Trust RPID : No
> Send RPID : No
> DTMFmode : rfc2833
> LastMsg : 0
> ToHost : sip.stanaphone.com
> Addr->IP : 204.147.183.18 Port 5060
> Defaddr->IP : 0.0.0.0 Port 0
> Def. Username: 12345678
> SIP Options : (none)
> Codecs : 0x4 (ulaw)
> Codec Order : (ulaw)
> Status : OK (20 ms)
> Useragent :
> Reg. Contact :
>
> new*CLI> sip show channel
> 14cdca3b6c900b1a54dcad2547234596 at sip.stanaphone.com
>
> * SIP Call
> Direction: Outgoing
> Call-ID: 14cdca3b6c900b1a54dcad2547234596 at sip.stanaphone.com
> Our Codec Capability: 4
> Non-Codec Capability: 1
> Their Codec Capability: 4
> Joint Codec Capability: 4
> Format ulaw
> Theoretical Address: 204.147.183.18:5060
> Received Address: 204.147.183.18:5060
> NAT Support: RFC3581
> Audio IP: AAA.BBB.CCC.DDD (local)
> Our Tag: as360c7ca5
> Their Tag: 0bd46ffd48e4fbffb3a68f13f8ad2599
> SIP User agent:
> Username: 87654321
> Peername: 12345678
> Original uri: sip:204.147.183.55:1024
> Need Destroy: 0
> Last Message: Tx: ACK
> Promiscuous Redir: No
> Route: sip:204.147.183.18;ftag=as360c7ca5;lr=on
> DTMF Mode: rfc2833
> SIP Options: (none)
>
> Finally, I built 1.2.18 from source today, and everything is
> working perfectly _except_ for StanaPhone, which continued to
> connect with no problems, but deliver no audio in either direction.
>
> I have no idea what else to try, and would appreciate _any_ guidance.
>
> Thanks in advance!
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