[asterisk-bugs] [JIRA] (ASTERISK-22206) No audio on Asterisk 11 when calling from Chrome to PSTN and calls go to voicemail/another call leg

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22206?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=208347#comment-208347 ] 

Rusty Newton edited comment on ASTERISK-22206 at 7/29/13 4:06 PM:
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Thanks for the report.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

No Audio is often not a bug, but just mis-configuration for the complexities of networking. We'll need more information. Not to mention you posted the logs in a format that is not convenient for the devs.

* Please attach your debug files in flat text format with a .txt extension (grab the Asterisk log file itself from /var/log/asterisk/ and just slap a .txt on it). Be sure the debug files include both VERBOSE and DEBUG , and if you capture from the CLI rather than grabbing the log files already generated for you in /var/log/asterisk, then turn up VERBOSE and DEBUG both to 5.
* We'll need a packet capture corresponding to each log file (capture with tcpdump or wireshark) it should include SIP and RTP, and all inbound/outbound traffic to or from Asterisk during those calls.
* For the examples of unexpected or failing behavior please indicate where the actual issue occurs (point to a particular timestamp or frame) in each file.
* For both logs and PCAPs include the entire call from the initial INVITE to the final SIP packet whatever that may be.
                
      was (Author: rnewton):
    Thanks for the report.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

No Audio is often not a bug, but just mis-configuration for the complexities of networking. We'll need more information. Not to mention you posted the logs in a format that is not convenient for the devs.

* Please attach your debug files in flat text format with a .txt extension (grab the Asterisk log file itself from /var/log/asterisk/ and just slap a .txt on it.
* We'll need a packet capture corresponding to each log file (capture with tcpdump or wireshark) it should include SIP and RTP, and all inbound/outbound traffic to or from Asterisk during those calls.
* For the examples of unexpected or failing behavior please indicate where the actual issue occurs (point to a particular timestamp or frame) in each file.
* For both logs and PCAPs include the entire call from the initial INVITE to the final SIP packet whatever that may be.
                  
> No audio on Asterisk 11 when calling from Chrome to PSTN and calls go to voicemail/another call leg
> ---------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-22206
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22206
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>         Environment: Server: Ubuntu 12.10
> root at ip-10-188-135-200:/opt# uname -a
> Linux ip-10-188-135-200 3.5.0-21-generic #32-Ubuntu SMP Tue Dec 11 18:51:59 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux
>            Reporter: James Mortensen
>            Assignee: James Mortensen
>         Attachments: badCallToVM.rtf, realGoodCall.rtf
>
>
> When making a call from Google Chrome to Asterisk 11.5 to the PSTN, we normally get two way audio. However, if the call goes to voicemail, or if no answer call forwarding is enabled, the call has no audio either direction, but Asterisk does receive audio from the PSTN gateway and it passes it onto Chrome. 
> Here are the following scenarios I test:
> - Call to cellphone. When I answer it, I verify two way audio.
> - Call to cellphone. Let it go to VM. I don't hear the VM message and it doesn't hear me. We only see RTP flowing from PSTN to Chrome in the Asterisk RTP logs.
> - On Verizon cellphone, set no answer call forwarding *71 followed by the forwarding number. Make call to cellphone and let it forward to the other number. Answer the call and verify no audio.
> I looked at the 200 OK's received from the gateway and both of them look normal. Yet we consistently get no audio when calls transfer or when they go to voicemail. This happens regardless of the PSTN carrier.  What's more, the transfers also happen deep in the stack at the carrier level. For instance, Verizon handles the switch to VM; this should be a black box to us.
> I've attached the Asterisk 11.5 logs from the 200 OK and beyond for a good call and for a bad call where it went to VM.  If I call from another PSTN number, I can of course verify that the voicemail message is indeed played.

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