TIPS Opensips No Audio

sstasterisk

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Exten to Exten dial works, but I receive no audio on both ends of the call
 

billsimon

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Cool story bro ;)

You should troubleshoot that by looking at the opensips logs as well as SIP captures.
 

billsimon

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Auth required is a normal part of the handshake; it is the proxy telling your phone that a password is required to make calls. Your phone replies with the username/password and the call proceeds. In the sngrep, take a look at the INVITE that is going from the proxy to the called phone. What IP address is being offered in the SDP portion of the INVITE (the lower portion; the IP address is on a line that starts with c= ) ? That is the IP address that the called phone will send the audio to. If it's the IP of the calling phone (that is, you are not using rtpproxy/rtpengine on your proxy to relay traffic), you need to make sure the phones can communicate directly with each other without being blocked by a firewall or NAT. If it's the IP of the proxy, you need to make sure rtpproxy is running so that your audio traffic is relayed.
 

sstasterisk

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Auth required is a normal part of the handshake; it is the proxy telling your phone that a password is required to make calls. Your phone replies with the username/password and the call proceeds. In the sngrep, take a look at the INVITE that is going from the proxy to the called phone. What IP address is being offered in the SDP portion of the INVITE (the lower portion; the IP address is on a line that starts with c= ) ? That is the IP address that the called phone will send the audio to. If it's the IP of the calling phone (that is, you are not using rtpproxy/rtpengine on your proxy to relay traffic), you need to make sure the phones can communicate directly with each other without being blocked by a firewall or NAT. If it's the IP of the proxy, you need to make sure rtpproxy is running so that your audio traffic is relayed.
Right this moment traveling, no reliable internet connection, but here is my /etc/default/rtpproxy https://pastebin.com/ybc20Q8D

Ip address matches public ip address
 

sstasterisk

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@billsimon ok, i think i've found the area to which your referring, i've now been able to get audio one way from the initial caller, but not from the receiver's end.
 

billsimon

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If you are using rtpproxy then opensips needs to know about it.

In your /etc/default/rtpproxy it shows that rtpproxy is listening on "udp:127.0.0.1:7890" for the control socket.

In your CP in the rtpproxy section add that as the control socket and reload.
 

sstasterisk

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Thanks i've gotten it working, also had to uncomment the line in opensips.cfg
 

sstasterisk

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have tested a couple calls back and forth, from friends in different areas, and its working, the only issue now is how to call other networks such as linphone or sip2sip, etc?
 

sstasterisk

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@billsimon I think this would be the last concern, I can't figure out how to dial other networks, such as linphone or sip2sip or demo.nerdvittles.com
 

billsimon

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You can set up those domains as gateways in the dynamic routing module. (NV article: http://nerdvittles.com/?p=29316) You can also dial them directly by putting the whole SIP URI into your phone. That's one idea behind using a proxy vs. a PBX which assumes that all SIP INVITEs are targeted to itself.
 

billsimon

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Sorry, don't really want to dig into the weeds on this. You have the right idea at this point. Investigate that 500 error from your provider to figure out why they are rejecting your call & maybe work with them if it's not clear.
 

sstasterisk

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Ok, thanks maybe you can point me in a direction of why my sip client says it's a bad destination.
 

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