SOLVED Can't get Audio with Skyetel trunks

lthown

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I'm assuming it has something to do with NAT but I have my external IP specified in the SIP settings, I have NAT=yes in all the trunks and I just can't figure out what needs to be done to get audio working

Code:
[2018-11-16 17:54:42] WARNING[1962]: chan_sip.c:4069 retrans_pkt: Retransmission timeout reached on transmission !!:RchKaGlKyGsXagjYR7k-acRLacapR7I0W7NbR5KYacxVaFN* for seqno 111103 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2018-11-16 17:54:42] WARNING[1962]: chan_sip.c:4093 retrans_pkt: Hanging up call !!:RchKaGlKyGsXagjYR7k-acRLacapR7I0W7NbR5KYacxVaFN* - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Channel SIP/Skyetel-SE-00000002 left 'simple_bridge' basic-bridge <9a5e72e1-4517-41c2-999c-bbd1d27f2d56>
  == Spawn extension (macro-dial-one, s, 52) exited non-zero on 'SIP/Skyetel-SE-00000002' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/Skyetel-SE-00000002' in macro 'exten-vm'
  == Spawn extension (ext-local, 701, 2) exited non-zero on 'SIP/Skyetel-SE-00000002'
    -- Executing [[email protected]:1] Macro("SIP/Skyetel-SE-00000002", "hangupcall,") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/Skyetel-SE-00000002", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Channel SIP/701-00000003 left 'simple_bridge' basic-bridge <9a5e72e1-4517-41c2-999c-bbd1d27f2d56>
    -- Executing [[email protected]:3] ExecIf("SIP/Skyetel-SE-00000002", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [[email protected]:4] Hangup("SIP/Skyetel-SE-00000002", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/Skyetel-SE-00000002' in macro 'hangupcall'
  == Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/Skyetel-SE-00000002'
 
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lthown

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The nice support guys at Skyetel sent me this debug log, looks like despite having the external IP set in the SIP general settings it's still sending the internal IP. Any ideas how to fix that?
 

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pnannery

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I am having issues with skyetel as well. same issue on SIP they dong get my external IP. so I use PJ_sip on port 5061 to receive calls however it won't work to make calls so I use chan_sip on port 5060 to make calls no one has be able to figure out why
 
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Nash Williams

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I seem to be having the same issue. I've been doing some searches to see if I can find some gap to fill, but so far, nothing.
 

lthown

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I think I have it solved now. Even though in FreePBX I have the external IP set, that only sets
external_media_address and external_signaling_address in pjsip.transports.conf

I just had to add externip=xx.xx.xx.xx to /etc/asterisk/sip_general_custom.conf and it's now working. I think if we were using pjsip instead of SIP may have just worked right away.
 

Nash Williams

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I tried your suggestion, Ithown, and it worked on my system as well. Thanks for sharing!
 

ws2000

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Any chance this fix could help my Obi/GV SIP trunk? Everything appeared to be working perfectly (incoming / outgoing calls) except for no audio.
 

lthown

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Any chance this fix could help my Obi/GV SIP trunk? Everything appeared to be working perfectly (incoming / outgoing calls) except for no audio.
entirely possible, you should give it a try!
 

dblair11

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Hello all,

I seem to be having this same issue, but I can't quite resolve it. I have added my external IP to my sip_general_custom.conf file and also have holes poked in my firewall on 5060 and 10000-20000 and NAT'd to my IncrediblePBX installation. The endpoints show as online and calls are going through, just no audio. When I dial outbound I get one ring on my handset then no audio but I can pick the call up on my cell, just no audio either direction.

I greatly appreciate any insight someone may have or any files that I should look at. Thanks!!
 

billsimon

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I have added my external IP to my sip_general_custom.conf file
No, don't do this. Use the web interface: Settings -> Asterisk SIP Settings

The more you veer off the path the more confusing it will get. Set all your SIP options in the web interface.
 

dblair11

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No, don't do this. Use the web interface: Settings -> Asterisk SIP Settings

The more you veer off the path the more confusing it will get. Set all your SIP options in the web interface.
I had tried doing that, but based on the previous replies with some luck I tried it. I have the same result with the line in or out of my .conf file :(

I'm oddly enough getting the same result with pjsip.
 

troysmoke

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I'll chime in.

I've done everything suggested so far:
Added Skyetel IPs in iptables
NAT/port redirection on my firewall for Skyetel IPs for 5060 and the whole world for 10000-20000
Made sure NAT config in SIP settings, and in the sip file

Is there something that needs to be done on the PBX to allow 10000-20000? Or are those ports already present and listening?

Any other thoughts on what to try? I'm a newbie, so I'd like to stick with the chan_sip implenetation.

Thanks.
troysmoke
 

dandy_don

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Well... I am having the exact issue. I can call in and out, but there is absolutely no audio either direction. Port 5060 is forwarded to the server. I have not forwarded or opened ports 10000-20000. The server is behind a router and the router's WAN address is configured in the general sip settings using the GUI.
When reviewing the configuration information from Skyetel, it indicates that one should edit sip_config.conf, but this is not possible to do in the GUI, and not sure if I should edit it directly via command line.
The instructions at https://nerdvittles.com/?p=21255 didn't indicate that this was necessary.

Any help is much appreciated!
Thanks,
Don
 

troysmoke

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Well... I am having the exact issue. I can call in and out, but there is absolutely no audio either direction. Port 5060 is forwarded to the server. I have not forwarded or opened ports 10000-20000. The server is behind a router and the router's WAN address is configured in the general sip settings using the GUI.
When reviewing the configuration information from Skyetel, it indicates that one should edit sip_config.conf, but this is not possible to do in the GUI, and not sure if I should edit it directly via command line.
The instructions at https://nerdvittles.com/?p=21255 didn't indicate that this was necessary.

Any help is much appreciated!
Thanks,
Don
Don,

You definitely have to forward ports 10000-20000 as well as 5060. I talked with Skyetel while troubleshooting mine and wound up forwarding 10000-50000 as my calls were using ports above 20000.
I have a dynamic IP, so I followed the tutorial for adding they dynamic IP to Skyetel and that also configured my external IP in sip_config.conf.

Hope that helps. Troy
 

dandy_don

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OK... This is a genuine "Emily Latella" Moment" (RIP Gilda Radner)...
I just forwarded UDP ports 10000-20000 and audio problems are gone.
"OH!, Never mind..."
Thanks,
Don
 
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wardmundy

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This is a router-specific problem. Some routers automatically open RTP ports for SIP traffic and some don't. You're probably safer forwarding UDP 10000-20000 to the private IP address of your Asterisk server.
 

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