Hadar Pedhazur
2007-Apr-25 15:13 UTC
[asterisk-users] No Audio with SIP to only one provider when switching 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@sip.stanaphone.com * SIP Call Direction: Outgoing Call-ID: 14cdca3b6c900b1a54dcad2547234596@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!
Brad Sumrall
2007-Apr-25 18:32 UTC
[asterisk-users] No Audio with SIP to only one provider whenswitching servers
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 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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@sip.stanaphone.com * SIP Call Direction: Outgoing Call-ID: 14cdca3b6c900b1a54dcad2547234596@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! _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hadar Pedhazur
2007-Apr-28 11:26 UTC
[asterisk-users] No Audio with SIP to only one provider when switching servers
I snipped all of the previous data, as I'm trying to "boil down" this problem to its essence... I turned off the firewall for a few seconds, and still got no audio. For those that will be suspicious, the commands were: shorewall stop shorewall clear tested connection, no audio shorewall start I also have a SIPPhone number, which (obviously), connects via SIP. I called that number from the outside, using one of their "Access Numbers", and my phone rang and I heard audio in both directions (this with the firewall back on), so SIP definitely works, just not with StanaPhone. Then I connected from another server that I run, which is behind a NAT router. That server is running 1.2.18 (as is the one that isn't working, but is on a public IP). Audio works perfectly with this one. To my knowledge the only difference between them is that the two servers that work are both Red Hat 9, with Asterisk 1.2.18 built from source. The one that fails is CentOS 5.0, with Asterisk 1.2.18 built from source. Here is a dump of the active channel from the NAT'ed server, which _works_: * SIP Call Direction: Incoming Call-ID: 342ed93a5d0cda7866f5b7122696e040@66.114.240.26 Our Codec Capability: 1822 Non-Codec Capability: 1 Their Codec Capability: 262 Joint Codec Capability: 262 Format ulaw Theoretical Address: 204.147.183.18:5060 Received Address: 204.147.183.18:5060 NAT Support: RFC3581 Audio IP: XX.XX.XX.XX (local) Our Tag: as78cfb201 Their Tag: da6aae9eb017f29b6c9de270fb85c352 SIP User agent: Sippy Original uri: sip:204.147.183.55:1024 Caller-ID: XXXXXXXXXX Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:204.147.183.18;ftag=da6aae9eb017f29b6c9de270fb85c352;lr=on DTMF Mode: rfc2833 SIP Options: (none) The only things edited above are the Audio IP, which is my correct "local" (before NAT) server address, and my Caller-ID. Everything else is unchanged. Here is the channel with dead audio: * SIP Call Direction: Incoming Call-ID: 3d0ccaf3482538f637278d3d2fd5272f@66.114.240.26 Our Codec Capability: 1542 Non-Codec Capability: 1 Their Codec Capability: 262 Joint Codec Capability: 6 Format ulaw Theoretical Address: 204.147.183.18:5060 Received Address: 204.147.183.18:5060 NAT Support: RFC3581 Audio IP: XX.XX.XX.XX (local) Our Tag: as45dbcfef Their Tag: 420bab62c5da9eae42686897ae65a385 SIP User agent: Sippy Original uri: sip:204.147.183.55:1024 Caller-ID: XXXXXXXXXX Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:204.147.183.18;ftag=420bab62c5da9eae42686897ae65a385;lr=on DTMF Mode: rfc2833 SIP Options: (none) The same two fields are edited above, and both were correct. To my eye, these are identical. Both are selecting ulaw, correctly. I'm stumped. I guess that I didn't do any packet tracing, but I'm not sure what the value of that would be given that it's not a firewall problem... Suggestions welcome!