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98 posts

Master Geek
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# 173831 8-Jun-2015 10:39
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Hello,

Due to circumstances beyond my control, I'm stuck getting a SIP phone working outside the office network. The issue I'm experiencing at the moment is when I call out from the remote phone to say, my cellphone, my cellphone rings but on the remote phone there is no audio (not even a ringing.. tone). When I answer, I get no audio in either direction. About 5 seconds later, the call terminates automatically on my cell. On the remote phone, call stays established until hung up normally.

The asterisk log (shown below, [pastebin link of whole log]) shows the NAT'ed Internal LAN ip of the phone -- 10.2.10.113, not the WAN IP. Should it be showing the WAN IP? Is this correct behaviour?

 

Is my diagnosis that this is a NAT problem correct? If yes, how can I resolve this? Unfortunately due to the location of the devices getting a wireshark capture is proving to be a little difficult.


== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/DVX/<CELL NO>
-- SIP/DVX-00000aa9 is making progress passing it to SIP/260-00000aa8
-- SIP/DVX-00000aa9 answered SIP/260-00000aa8
[2015-06-08 10:26:27] WARNING[2270]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 1019324208@10.2.10.113 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2015-06-08 10:26:27] WARNING[2270]: chan_sip.c:4198 retrans_pkt: Hanging up call 1019324208@10.2.10.113 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/260-00000aa8", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/260-00000aa8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/260-00000aa8", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/260-00000aa8", "") in new stack



[1]: http://pastebin.com/tKLRux6Z

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27899 posts

Uber Geek
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  # 1319213 8-Jun-2015 10:43
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Yes the issue is NAT.

Exactly what's causing it would require knowing the exact network settings and PBX configuration.



921 posts

Ultimate Geek
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  # 1319237 8-Jun-2015 11:12
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Sounds like SIP is getting through, but RTP is not.

Do you forward any ports through to the internal Asterisk server?  Last time I did this (er, 10 years ago?) I forwarded a range of 1024 UDP ports through and told Asterisk to use only those ports for RTP.

 
 
 
 




98 posts

Master Geek
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  # 1319251 8-Jun-2015 11:23
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Here's the topology: http://i.snag.gy/awQr4.jpg

No SIP ALGs or anything there. Both ends are pfSense.

Edit. @deadlyllama -- the SIP/RTP ports are forwarded. UDP 5060 and 10000-20000. But I don't think this is the problem. The reason being, the Asterisk PBX logs are referencing the LAN IP Of the phone which will be unreachable from the PBX. The Asterisk PBX should be responding to the WAN IP of the phone.

Under Status > Peers, the correct WAN IP of the phone is shown.

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Ultimate Geek
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  # 1319465 8-Jun-2015 15:15
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I had similair issues with my setup behinf pfsense. I think it came down to an isue with pfsense's random outbound port mapping (NAT)

I think you can disable outbound port remapping but leave NAT enabled.







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Master Geek
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  # 1319475 8-Jun-2015 15:26
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Is that the "static port NAT" option?

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Uber Geek
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  # 1319477 8-Jun-2015 15:28
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bonkas: I had similair issues with my setup behinf pfsense. I think it came down to an isue with pfsense's random outbound port mapping (NAT)

I think you can disable outbound port remapping but leave NAT enabled.

yea I seemt o remember the same also. Static NAT fixed it (slightly different scenario).







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Master Geek
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  # 1319499 8-Jun-2015 15:59
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Static NAT is on already in my test bed, didn't seem to help. Static NAT however did help with my trunk not registering with WxC (got stuck on Invite Sent).

 

Anyway I think I will move the PBX to a public IP rather than waste any more time on this.

 
 
 
 


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  # 1319506 8-Jun-2015 16:09
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You still haven't provided your PBX settings - what are you using for NAT and IP address settings for both extensions and the PBX?







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Master Geek
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  # 1319737 8-Jun-2015 21:55
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Moved PBX to public IP -- works flawlessly now.

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  # 1319743 8-Jun-2015 22:02
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breakaway: Moved PBX to public IP -- works flawlessly now.


It should do. The issue is in your PBX extension or SIP NAT settings.

I hope your box is fully firewalled off with no exposed ports outside your other site or it won't take long to be h8x0red!




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