Voice, outgoing calls PAP2T problem
Hi all
I have a pap2t voip adapter and a lot of problems.
Here's the situation.
In our office, we have a linux box that acts as a router, iptables configured by shorewall. Connected to the linux box is the lan internet.
After a difficult time, I finally got to do the adapter to connect and register with our sip (sipgate.de) provider.
But, it's almost everything.
What works, is that I can call from extern and get a dial tone on the phone (connected to the adapter). But that's all.
When I try to make a call, I get the following error in the system log:
[0:5060]-> 217.10.68.147:5060
[0:5060]-> 217.10.68.147:5060
Sep 24 12:30:52 192.168.15.9 [0:0] call of Rel AUD When I look into the adapter information page, I see the message that the call is not valid. What I find suspicious, it's that I don't send or receive packets rtp or bytes depending on the info page while I get and send a lot of sip packets and bytes. I tried transfer ports 16000-17000 la carte (I gave him a fixed ip address), but this custom work too. I googled a lot and I tried many things, which lead me to this situation, and here I stand without knowing where to go. Greetings and thanks in advance Sven I solved the problem. What I had to do was to unload the modules nf_nat_sip, nf_conntrack_sip and add them to the DONT_LOAD line in shorewall.conf. She's. Tags: VoIP Adapters No voice with incoming calls PAP2T I bought a LinkSys PAP2T unlocked last year that worked beautifully until a few days ago. Outgoing calls work fine, but I now have problems with incoming calls only. When I receive an incoming call, the phone rings, but when I pick it up, I can't hear anything and cannot the caller on the other end. After a few seconds, my phone disconnects and begins to beep as if the line is busy. This occurs regardless of who calls me and whther they use a VoIP device or a normal PSTN phone. I have not changed the settings on the router (a Belkin) or the PAP2T, he simply stopped working a few days ago! I have a connection 10MB Virgin Media broadband service for my VoIP provider is Draytel that have verified the account and settings and says it works very well on their test kit. Since then, I tried just using line 2 and the deactivation of line 1 to do a hard reset of the PAP2T and updated the firmware to the latest version. Even tried to change the analog phone connected, but I still have the problem. Does anyone have any ideas as to why it suddenly stopped working with anything has been changed and what I might be able to do to fix? The PAP2T is still the title of the 12 month warranty period then maybe she developed a fault? Thank you very much in advance for all those who can help you. Cheers, Vic For me it looks like the problem of RTP streams - probably, it does not pass the firewall / nat on the path between SIP provider and you. This may have is that your Belkin router/NAT, or on internet ISP router/NAT enabled SIP ALG (Application Layer Gateway). If this is the case, you will need to disable SIP ALG on your router or ask your ISP internet to disable on the firewall router. Another question may have been if you (accidentally) changed the port forwarding on your Belkin router, or the PAP2T local LAN IP address changed and port forwarding is no longer works. So my proposals to check: -check and DISABLE SIP ALG on your Belkin (if the router has this feature) -check the port forwarding on your router Belkin... IF you set up port forwarding for 5060/61 you MUST set the same for RTP ports (16384-16483) -If you don't have port forwarding on your Belkin, try before installation of 5060-5061 AND 16384-16483 *. * (If this is too many ports, set the RTP on PAP2T portrange to 16384-16394 for example and then forward only go) -If neither of above of aid, change your SIP Line1 of 5060 for port we will tell 6070 and RTP ports to the beach we'll tell 17300-17310 -Finally, you can do it on your side is to remove all the port forwarding and put the PAP2T to your router Belkin DMZ settings. Problems with the outgoing calls and constant reboot I got my Droid on 12/23. It's a company phone, so I hesitate to consider the Verizon store. It worked great until the 12/29 and then he stopped outgoing calls and began planting time in time, and then more often otherwise, now it hangs every two or three minutes. It became useless as both a phone and for the most part. I took the card SD and that did not help. 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Sep 24 12:30:24 192.168.15.9 [0:5060]-> 217.10.68.147:5060
Sep 24 12:30:24 192.168.15.9 sip:[email protected] GUEST SIP / 2.0 ^ M Via: SIP/2.0/UDP 77.20.237.113:5060; branch = fc32b89a z9hG4bK ^ M from: Richter
Sep 24 12:30:44 192.168.15.9 [0:5060]-> 217.10.68.147:5060
Sep 24 12:30:44 192.168.15.9 NOTIFY sip: 217.10.68.147 SIP / 2.0 ^ M Via: SIP/2.0/UDP 77.20.237.113:5060; branch = a39224c8 z9hG4bK ^ M from: Richter
Sep 24 12:30:52 CC 192.168.15.9: failure w / call
Sep 24 12:30:52 192.168.15.9 Sess terminatedSimilar Questions
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[0:0]CC:STUN OK:192.168.1.8->83.247.45.219, 5060->5060 16388->16388
STUN has detected that your external address is 83.247.45.219, the same port numbers by NAT.
[0]->212.45.38.48:5060(979)INVITE sip:064.....[email protected]/* */ SIP/2.0Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK-f6bed01From: "3134.....99"
Call setup. Audio stream Incommimg asked to be sent to 83.247.45.219:16338. It is in the transmitted range (16000-17000).
[0]<<212.45.38.48:5060(817)
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.8:5060;received=83.247.45.219;branch=z9hG4bK-da26c2e5
From: "3134.....99"
Of the other answer for call setup. Coming out of the audio stream asked to be sent to 212.45.38.48:63808. He is outgoing don't stream so no special arrangements on recalcitrant NAT.
It doesn't look good in the direction of the cabin, is not so surprising that it worked.
Most important messages for incomming call (for outgoing see above). Commented by me.
[0]<<212.45.38.48:5060(789)
INVITE sip:3134.....[email protected]/* */:5060 SIP/2.0
Via: SIP/2.0/UDP 212.45.38.48:5060;branch=z9hG4bK8jld5h207grg5v4up6m0.1
From:
Configuration of incoming call. Coming out of the audio stream asked to be sent to 212.45.38.48:63316. He is outgoing don't stream so no special arrangements on recalcitrant NAT.
[0]->212.45.38.48:5060(918)
SIP/2.0 200 OK
To:
Our response to the incoming call configuration. Audio stream Incommimg asked to be sent to 83.247.45.219:16392. It is in the transmitted range (16000-17000).
It seems not good in the direction of the cabin. According to statistics BYE...
P-RTP-Stat: PS=990,OS=166160,PR=594,OR=142560,PL=0,JI=0,LA=0,DU=17,EN=G711a,DE=G711a
... SPA112 sent 166160 B/990 packets of audio data at the other end. I see no outgoing audio problems. Call person to hear.
Well, despite the packages are OK at the time, they may not be the same at the receiving end.
You have active SIP ALG feature on EdgeRouter? SIP - LG is known for having questions about so many routers, which one of the Uniquity. In addition, you should not combine STUN on SPA112 with SIP - ALG on router. Please turn off ALG SIP on the router.
If it does not solve your problem, then ask provider upstream assistance. We need SIP & RTP packets captured on the side of the trunk.
A note unrelated to your problem. It seems that you SIP-> settings-> value 0.030 RTP toRTP packet size . It's a suboptimal value for your upstream provider (in fact it is bad everywhere in Europe). Your provider uses 0.020 (according to the settings specified in the INVITATION). Consider the same value. It can increase the sound quality (for the end-user of the other).
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