I have been banging my head against a wall trying to get this figured out. I have been using csipsimple, bria, sipdroid and they all drop calls at exactly 32 seconds. My understanding is that “the app never received the confirmation of the fact the call is established”. Has anyone else experienced this?
This is an examle of CLI:
] WARNING chan_sip.c: Hanging up call [email protected] - no reply to our critical packet (see doc/sip-retransmit.txt).
After trying everything I am almost convinced this is verizons fault? Any ideas?
10.233.242.106 is in a private network unless you have a route there you probably need to set up sip to use its NAT settings to rewrite all headers and make sure your firewall/router forwards those SIP headers to your box.
Im sorry i pasted that from a forum by mistake. this is what i am getting,
chan_sip.c:4198 retrans_pkt: Hanging up call -DkCxlta9soUUULDHJ9YLRDselkJJar2-S - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Still a network problem still probably still a NAT/router one also you need to keep the SIP conversation open and well translated.
I have a asus ac66u router. I have run multiple firmwares including dd-wrt. Is there anything more specific i should look for? I am going to try another router now. ill report how it goes.
So far you have been completely unspecific e can’t read your mind, please explain you network a little better.
Im sorry. My network is simply my router (AC66u) connected directly to my pbx. All of the necessary ports are forwarded. To rule out the router i have just tried everything with a different router and am still having the problem. Everything is working great within my network. Nat is enabled in the PBX advance sip settings. I have tired multiple sip providers and all drop at exactly 32 seconds using a number of soft phones for android.
Just registerd a obi202 remotely using the same settings. To my suprse everything worked great. So my conclusion is that verizon is to blam here not freepbx. thank you for your time. Anyone else using verizon and trying to make calls on a sip client softphone please let me know what your expieriences are.
Thank You Again
I have learned that verizon uses SIP ALG on there networks. My understanding is this is something that just does not work with NAT. Most people here would advise you to turn this off in your router if you are having problems with 1 sided conversations.
This is the reason people are having problems with sip and verizon. A workaround is to use TCP. Here is a guide for anyone having similiar problems.