Problem with remote extension in Netherlands

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rollinsolo said:
and here are the RTF feedback from a call to my remote extension

Got RTP packet from 173.78.144.2:10070 (type 00, seq 006890, ts 674256760, len 000160)
Got RTP packet from 173.78.144.2:10070 (type 00, seq 006891, ts 674256920, len 000160)
Got RTP packet from 173.78.144.2:10070 (type 00, seq 006892, ts 674257080, len 000160)
Got RTP packet from 173.78.144.2:10070 (type 00, seq 006893, ts 674257240, len 000160)
rollinsolo said:
and here are the RTF feedback from a call to my remote extension

Got RTP packet from 173.78.144.2:10070 (type 00, seq 006890, ts 674256760, len 000160)
Got RTP packet from 173.78.144.2:10070 (type 00, seq 006891, ts 674256920, len 000160)
Got RTP packet from 173.78.144.2:10070 (type 00, seq 006892, ts 674257080, len 000160)
Got RTP packet from 173.78.144.2:10070 (type 00, seq 006893, ts 674257240, len 000160)
This indicates one way audio
it should look like

Sent RTP packet to <far end>
Got RTP packet from <far end>
Sent RTP packet to <near end>
Got RTP packet from <near end>
etc.
etc.

I assume 173/64.144.0/29 is your verizon network and .2 is your asterisk box. 5060 and 10000- are both open on this address so the problem is definitely on the far end router/phone.
Yes to the "enable subscribe" (never used an Atcom, but . . .) which should be on and you should then see INVITE's in sip debug being sent and answered, you say they are all registered now? and a close inspection of the call to it from the CLI might help (maybe DND or something?)
 
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Im not sure how to do an extended ping I can ping and get back ping [externIP] (externIP) 56(84) bytes of data.
 
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No Verizon is my service, I have a remote Atcom phone here to test as a remote extension to their PBX, the .2 is my router, the link for that Atcom is http://www.atcom.cn/En_download.html It has many options under advanced sip settings: /// means its the same row different collumn.

Register Expire Time seconds /// Forward Type
NAT Keep Alive Interval seconds /// Forward Phone Number
User Agent /// Server Type
Signal Key /// DTMF Mode
Media Key /// RFC Protocol Edition
Local Port /// Transport Protocol
Enable Subscribe /// Subscribe Expire Time seconds
Enable Conference Num /// Conference Number
Enable Keep Authentication /// Signal Encode
NAT Keep Alive /// Rtp Encode
Enable Via rport /// Enable Session Timer
Enable PRACK /// Answer With Single Codec
Long Contact /// Auto TCP
Click To Talk /// Enable URI Convert
 
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PING externip (externip) 56(84) bytes of data.

--- externip ping statistics ---
1137 packets transmitted, 0 received, 100% packet loss, time 1135926ms
 
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rollinsolo said:
No Verizon is my service, I have a remote Atcom phone here to test as a remote extension to their PBX, the .2 is my router, the link for that Atcom is http://www.atcom.cn/En_download.html It has many options under advanced sip settings: /// means its the same row different collumn.

Register Expire Time seconds /// Forward Type
NAT Keep Alive Interval seconds /// Forward Phone Number
User Agent /// Server Type
Signal Key /// DTMF Mode
Media Key /// RFC Protocol Edition
Local Port /// Transport Protocol
Enable Subscribe /// Subscribe Expire Time seconds
Enable Conference Num /// Conference Number
Enable Keep Authentication /// Signal Encode
NAT Keep Alive /// Rtp Encode
Enable Via rport /// Enable Session Timer
Enable PRACK /// Answer With Single Codec
Long Contact /// Auto TCP
Click To Talk /// Enable URI Convert
Sounds like you've got some reading to do . . .

Did the Atcom phones work nicely while on the same network? (I assume so)
Are the codecs used conversant with eachother? (I assume so if the above was yes)
You seem to have a static subnet, just for the hell of it it might be worth trying to expose the phone itself "naked" to the internet (or a little less disruptive,does your router allow you to do one-to-one routing of en external IP to the phone? (bridge mode)) (just for debugging?)
 
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This is in the debug when I make a call.

v=0
o=root 23420 23420 IN IP4 pbxIP
s=session
c=IN IP4 pbxIP
t=0 0
m=audio 14128 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv
 
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OK just tried something new, in elastix I changed extension nat = no, and removed it from sip_nat.conf as well and shazaam I can hear. Will test more and see if that is the case all around, and will inform, hope my lack of knowledge helped spread some knowledge in the troubleshooting world. Thanks.
 
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And you can change extensions qualify back to yes. YES! I am so excited I get to move on to new problems now. Thanks Again.
 
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just noticed the thread origin!

sorry wiseoldowl for hijacking . . .
 
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yes I can assign a static Ip straight to the phone and see if that works, but the thing is mine is working, maybe because I already have static, and the others do not, I can order static from them if I have to which I think I will but I was hoping to skip that step. I got my phone to work by disabling nat on mine and its perfect in and out, so this is def a nat issue. You were right like 6 posts ago, now its just tinkering. Its hard to just make one little change and run 12 tests on that change. I am trying not to make 5 changes and then try to reverse engineer what I did, but I do get anxious. Tired now will pick this us tomorrow. Thanks.
 
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Well I went to the site you gave for the atcom device, I read the AT510 manual the first one on the list) (actually I tried to read it , but my chinglish is poor and it hurt my head, I'm not surprised your confused!!) I notice that this phone supports IAX(2) as a voip protocol and suggest you try it (if your model supports it,) IAX nat transversal mechanism is a lot cleaner (oh and BTW, that's one hell of an ugly phone). I think I would try a soft-phone as a remote device before I went any further to eliminate the deciphering inscrutable oriental implementations of SIP (Note to self, "Don't be tempted to buy cheap chinese phones without learning mandarin/cantonese so I can call customer support!!) B)
 
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Two things, the phone was more expensive than a Aastra 6731 which is a way better phone (stupid me) Live Learn and Make mistakes along the way, 2nd thing, since my phone works here and I use static IP's I have contacted the ISP to order Static IP's for the other locations as well and I will replicate my settings here to there. I had IAX working, for Inbound, but it would break and mainly I wanted to do Full SIP to fully learn what is happening with SIP calls, good experience for me, bad for the Customer, I did a lot of free work for him so he is ok for now.
 
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Ok so things are still messed up with good ole NAT traversal and after many many manipulations I am basically at the same drawing board as before, luckily I got to take a break from this when I customer Cut Over Phone service and was down for Two days because the NEC Phone system would not pass a long Caller ID Name, rejected all calls.

Anyways I did like you said Dicko and used the IAX on the phones as well as the SIP, I have to use both in that scenario because it will call out on SIP but only come in on IAX2. But just a refresher I have a working model at home and cannot pinpoint what I am doing different here as opposed to at the customers remote locations. My model at home is using just a SIP account and stays connected for days without having to refresh anything, turn Nat on or off, qualify set to no or anything, it just works how its supposed to and cannot figure out why.

I would give a big ugggh but I have reached a certain enlightenment with troubleshooting and also I am very stubborn so I really just want to learn at this point. I am enjoying this forum and can't wait to actually know what I'm typing about so I can contribute to it myself. Thanks.
 
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/SIP/Registry/6900 : xxx.78.xxx.2:49155:60:6900:sip:6900@xxx.78.xxx.2:49156
its seems that my phone is using some higher ports for udp and the others are stuck on 5060. should I forward those remote phones and force them to use 10000-20000? I have set any forwarding on my IP phone it just works.
 
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I meant to say I have not set any kind of forwarding on my remote extension and it works fine. Hmmmm.
 
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given the apparent "quirkiness" of the hardware you are using and your problems with nat traversal, perhaps (if the phones support it) use of a STUN server by the phones might help.
 
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after a quick search I see there are many options as to what to use as a Stun client/server do you have any that you would recommend or have proven to solve a Nat Issue such as this.
 

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