0

Remote Extensions: No Audio in Calls

Yamil Cavodeassi

September 21, 2017 22:04

None

Follow

Hello, 

We have two offices:

Office A: 

We own a s50 PBX with this setup, Service Provider Trunk.

6 Local extensions, all ander the same subnet 10.0.0.0/24

 

Everything is behind a linksys NAT router, with SIP/RTP ports forwarded to the pbx.

Office B:

19 Remote extensions, all in the same subnet 172.28.5.0/27

Behind a Fortinet in NAT mode, SIP ALG disabled.

Remotes SIP Registration is working flawless. 

 

Calls from extensions in Office A to extensions in Office A are OK.

Calls from extensions in Office A to External numbers are OK.

Calls from extensions in Office A to extensions in Office B are OK and vice versa. 

Calls from extensions in Office B to External numbers are OK.

Calls from extensions in Office B to extensions in Office B rings but NO AUDIO!

I dont know if its a NAT issue, but I can't find a fix for it.

Thank you very much, any help will be appreciated. 

 

• Facebook
• Twitter
• LinkedIn

17 comments

• 0
  
  Oscar Colka September 25, 2017 05:35

  Can you share your settings on SIP -> NAT settings in the S100?

  Also, if would be better if you can provide some pcap logs.

  Please go to S100 web GUI -> Maintenance -> Troubleshooting -> Ethernet capture tool to get some pacp logs

  

• 0
  
  Yamil Cavodeassi September 25, 2017 15:39

  These are my SIP -> NAT settings (It's a s50)

   

  

  The Local Network ID, is the Office A private subnet. Do I have to add the Office B private net too? 

  I've attached some pcaps logs.. from the s50 and a sniffer from the vlan on the other side (B office).https://ufile.io/89ano 

  Thank you for your time! 

• 0
  
  Oscar Colka September 26, 2017 01:52

  Hello Yamil,

  No need to add the office B prevate net.

  Please capture some more logs:

  Make sure there is not one make the calls, as the size of the logs in the S50 is limited and if there is more calls during you cpature the logs, you may not capture the expected logs successfully.

  1. Start capture the pcap logs on S50 packet capture tool

  2. Start capture the pcap logs on IP phone one in office B

  3. Start capture the pcap logs on IP phone two in office B

  4. Make a call from IP phone one to IP phone two

  5. Stop all the pcap logs capture

  6. Post the logs here;

• 0
  
  Yamil Cavodeassi September 28, 2017 19:33

  Hello Oscar,

  I've figured out that RTP ports were closed by fgt at my Office B side. Now calls between internals have audio and work as expected.

  My only concern is that incoming calls are being dropped after 25 seconds. I'll make a capture and post it here for further details.

  Again, thank you very much. 

• 0
  
  Yamil Cavodeassi September 28, 2017 21:08

  <--- Transmitting SIP request (458 bytes) to UDP:154.0.185.4:5060 --->

  BYE sip:1120840729@154.0.185.4:5060;user=phone;transport=udp SIP/2.0

  Via: SIP/2.0/UDP 200.80.198.198:5060;rport;branch=z9hG4bKPj989d6045-8b0e-406e-94fb-4a708e5722be

  From: <sip:1100@154.1.231.218;user=phone>;tag=a3e243d5-6f69-43e2-980c-787294a9f222

  To: <sip:1120840729@154.0.185.4;user=phone>;tag=ss216g49-CC-30

  Call-ID: 2baw23wbw2azw4sz241xs9x96xzbyg9a@SoftX3000

  CSeq: 3567 BYE

  Max-Forwards: 70

  User-Agent: Yeastar S50-30.5.0.8

  Content-Length:  0

   

   

   

  <--- Received SIP response (379 bytes) from UDP:154.0.185.4:5060 --->

  SIP/2.0 200 OK

  Via: SIP/2.0/UDP 200.80.198.198:5060;received=154.1.231.218;branch=z9hG4bKPj989d6045-8b0e-406e-94fb-4a708e5722be;rport=5060

  From: <sip:1100@154.1.231.218;user=phone>;tag=a3e243d5-6f69-43e2-980c-787294a9f222

  To: <sip:1120840729@154.0.185.4;user=phone>;tag=ss216g49-CC-30

  Call-ID: 2baw23wbw2azw4sz241xs9x96xzbyg9a@SoftX3000

  CSeq: 3567 BYE

  Content-Length: 0

   

   

   

      -- Channel PJSIP/trunk-Claro-endpoint-00000039 left 'simple_bridge' basic-bridge <158d78e0-54ba-4b25-a57d-3acb9ac3da59>

   

      -- Channel PJSIP/124-0000003a left 'simple_bridge' basic-bridge <158d78e0-54ba-4b25-a57d-3acb9ac3da59>

   

    == Spawn extension (macro-realstexten, s, 16) exited non-zero on 'PJSIP/trunk-Claro-endpoint-00000039' in macro 'realstexten'

  <--- Transmitting SIP request (415 bytes) to UDP:181.45.106.4:5060 --->

  BYE sip:124@181.45.106.4:5060 SIP/2.0

  Via: SIP/2.0/UDP 200.80.198.198:5060;rport;branch=z9hG4bKPj6941f4a7-ee79-4095-893b-5be22d678548

  From: <sip:1120840729@10.0.0.80>;tag=dbad0268-9f12-4d5b-9d72-6d15cb49fcbf

  To: <sip:124@181.45.106.4>;tag=2265150386

  Call-ID: 6ac4ce7a-7a08-4a87-98a2-e25b5f317293

  CSeq: 7602 BYE

  Reason: Q.850;cause=16

  Max-Forwards: 70

  User-Agent: Yeastar S50-30.5.0.8

  Content-Length:  0

   

   

   

    == Spawn extension (macro-stdexten-fromoutside, s, 4) exited non-zero on 'PJSIP/trunk-Claro-endpoint-00000039' in macro 'stdexten-fromoutside'

   

    == Spawn extension (ivr_6500, 124, 1) exited non-zero on 'PJSIP/trunk-Claro-endpoint-00000039'

   

  <--- Received SIP response (365 bytes) from UDP:181.45.106.4:5060 --->

  SIP/2.0 200 OK

  Via: SIP/2.0/UDP 200.80.198.198:5060;rport=5060;branch=z9hG4bKPj6941f4a7-ee79-4095-893b-5be22d678548

  From: <sip:1120840729@10.0.0.80>;tag=dbad0268-9f12-4d5b-9d72-6d15cb49fcbf

  To: <sip:124@181.45.106.4>;tag=2265150386

  Call-ID: 6ac4ce7a-7a08-4a87-98a2-e25b5f317293

  CSeq: 7602 BYE

  User-Agent: Yealink SIP-T21P_E2 52.81.0.110

  Content-Length: 0

• 0
  
  Oscar Colka September 29, 2017 01:42

  Hello Yamil,

  The SIP logs you post above is not the full logs, please try capture again the pcap logs.

   

  By the way, the incoming drop after 25 seconds happen on office A or on office B?

  Is it happen on incoming call from external number or happen between two offices?

• 0
  
  Yamil Cavodeassi September 29, 2017 02:03

  I'm sorry the calls drop after exactly 32 seconds. It happens only from external numbers, incoming calls drop on both offices. 

  This was the full pcap pbx capture. https://ufile.io/5dpon 

• 0
  
  Yamil Cavodeassi September 29, 2017 02:18

  https://ufile.io/65j3p 

  New caps from phone and pbx, thank you!

• 0
  
  Oscar Colka September 29, 2017 02:24

  The public IP address of S50 seems changed to 154.1.231.218, can you try set the NAT settings on S50 and change the external IP address to 154.1.231.218?

  

• 0
  
  Oscar Colka September 29, 2017 02:37

  
  
  

  The normal SIP call flow is 

  

   

  Below is what I found from the logs you post:

  

  The SIP trunk provider didn't send the ACK to S50 correctly, that caused the S50 hangup the call.

  The contact header that S50 sent to SIP trunk provider using IP address 200.80.198.198, it seems incorrect as the S50 public IP address seems changed to 154.1.231.218.

   

  

• 0
  
  Yamil Cavodeassi September 29, 2017 02:51

  The IP address 200.80.198.198 is from my ISP. Its set to register sip phones. 

  The IP address 154.1.231.218 is the address assigned by the service provider to the WAN port of the s50. 

  Its on dual mode: 

  If I set the 50s public address as external address Calls are not being drop anymore but I have no audio due to a NAT miss configuration.

• 0
  
  Oscar Colka September 29, 2017 03:03

  OK, I got you, try below:

  1. Change the External IP address back to 200.80.198.198;

  2. Keep the old localnet setting and add another local net on NAT settings page of S50

  154.1.231.218/255.255.255.255

  Check if above settings help.

   

• 0
  
  Yamil Cavodeassi September 29, 2017 03:18

  Did that without succeed. The calls keep dropping after 32 seconds. 

  

  Routes are:

  I have no idea how to solve this, I really appreciate your time. 

• 0
  
  Oscar Colka September 29, 2017 03:44

  Sorry, my mistake, we should add the SIP trunk IP address into the local net setting on NAT settings page of S50, try setup the localnet as:

  154.1.231.218/255.255.255.255 -- remove this one and add below one

  154.0.185.4/255.255.255.255

• 0
  
  Yamil Cavodeassi September 29, 2017 04:23

  That's did the trick, You did it!

  Thank you for taking the trouble to help me. I do appreciate it. 

  Bye, have a good day!

• 0
  
  Oscar Colka September 29, 2017 05:01

  Glad to know it works!  : )

• 0
  
  Kingsley October 20, 2017 06:05

  Hello Yamil, please how did you open the RTP ports to make calls go through?