SOLVED Can't get Audio with Skyetel trunks

Discussion in 'Open Discussion' started by lthown, Nov 16, 2018.

  1. lthown

    lthown New Member

    Joined:
    Nov 15, 2007
    Messages:
    16
    Likes Received:
    8
    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'
    
     
    Nash Williams likes this.
  2. lthown

    lthown New Member

    Joined:
    Nov 15, 2007
    Messages:
    16
    Likes Received:
    8
    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?
     

    Attached Files:

    Nash Williams likes this.
  3. pnannery

    pnannery New Member

    Joined:
    Apr 5, 2010
    Messages:
    10
    Likes Received:
    2
    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
     
    Nash Williams likes this.
  4. Nash Williams

    Nash Williams Good...bad...I'm the guy with the gun.

    Joined:
    Nov 18, 2018
    Messages:
    5
    Likes Received:
    0
    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.
     
  5. lthown

    lthown New Member

    Joined:
    Nov 15, 2007
    Messages:
    16
    Likes Received:
    8
    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 and wardmundy like this.
  6. Nash Williams

    Nash Williams Good...bad...I'm the guy with the gun.

    Joined:
    Nov 18, 2018
    Messages:
    5
    Likes Received:
    0
    I tried your suggestion, Ithown, and it worked on my system as well. Thanks for sharing!
     
  7. Nash Williams

    Nash Williams Good...bad...I'm the guy with the gun.

    Joined:
    Nov 18, 2018
    Messages:
    5
    Likes Received:
    0
  8. ws2000

    ws2000 New Member

    Joined:
    Nov 17, 2010
    Messages:
    10
    Likes Received:
    0
    Any chance this fix could help my Obi/GV SIP trunk? Everything appeared to be working perfectly (incoming / outgoing calls) except for no audio.
     
  9. lthown

    lthown New Member

    Joined:
    Nov 15, 2007
    Messages:
    16
    Likes Received:
    8
    entirely possible, you should give it a try!
     
  10. dblair11

    dblair11 New Member

    Joined:
    Oct 12, 2018
    Messages:
    2
    Likes Received:
    0
    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!!
     
  11. billsimon

    billsimon Experienced in Asterisk, FreePBX, and SIP

    Joined:
    Jan 2, 2011
    Messages:
    906
    Likes Received:
    274
    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.
     
  12. dblair11

    dblair11 New Member

    Joined:
    Oct 12, 2018
    Messages:
    2
    Likes Received:
    0
    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.