[asterisk-bugs] [JIRA] (ASTERISK-26354) Cannot make outgoing calls from SIP account

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26354?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232237#comment-232237 ] 

Asterisk Team commented on ASTERISK-26354:
------------------------------------------

The severity of this issue has been automatically downgraded from "Blocker" to "Major". The "Blocker" severity is reserved for issues which have been determined to block the next release of Asterisk. This severity can only be set by privileged users. If this issue is deemed to block the next release it will be updated accordingly during the triage process.

> Cannot make outgoing calls from SIP account
> -------------------------------------------
>
>                 Key: ASTERISK-26354
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26354
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: Core/RTP
>    Affects Versions: 1.8.12.2
>         Environment: Ubuntu 12.04 LTS
>            Reporter: kristian slavov
>
> Hello,
> A client of mine cannot make outgoing(incoming works) calls via his SIP account, here is the log:
> 7-sept.-2016 13:42:26.468   [MS105000] C:1.2: No RTP packets were received:remoteAddr=217.16.11.101:18420,extAddr=0.0.0.0:0,localAddr=196.207.239.214:9000
> 07-sept.-2016 13:42:25.885   Leg L:1.1[Extn:200] is terminated: Cause: BYE from PBX
> 07-sept.-2016 13:42:25.884   [CM503008]: Call(C:1): Call is terminated
> 07-sept.-2016 13:42:25.883   [CM503023]: Call(C:1): Call recording is stopped, audio file: C:\ProgramData\3CX\Instance1\Data\Recordings\200\[HELIX]_200-0033248256987_20160907134224(1).wav
> 07-sept.-2016 13:42:25.871   Leg L:1.2[Line:10009>>0248256987] is terminated: Cause: BYE from 217.16.11.101:5060
> 07-sept.-2016 13:42:25.006   [CM503007]: Call(C:1): Line:10009>>0248256987 has joined, contact <sip:0033214001840 at 217.16.11.101:5060>
> 07-sept.-2016 13:42:25.005   [CM503007]: Call(C:1): Extn:200 has joined, contact <sip:200 at 192.168.221.52:5060>
> 07-sept.-2016 13:42:25.004   [CM503022]: Call(C:1): Call recording is started, audio file: C:\ProgramData\3CX\Instance1\Data\Recordings\200\[HELIX]_200-0033248256987_20160907134224(1).wav
> 07-sept.-2016 13:42:24.827   L:1.2[Line:10009>>0248256987] has joined to L:1.1[Extn:200]
> 07-sept.-2016 13:42:24.827   NAT/ALG check:L:1.2[Line:10009>>0248256987] RESPONSE 200 on 'INVITE' - basic check passed. No information for extended checks
> 07-sept.-2016 13:42:24.826   [CM505003]: Provider:[SIPAccountMoh] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:0033214001840 at 196.207.239.214:5060]
> 07-sept.-2016 13:42:21.733   [CM503026]: Call(C:1): Route 2 to PSTNlineSONATEL is not active (Busy/Not registered)
> 07-sept.-2016 13:42:21.733   [CM503025]: Call(C:1): Calling T:Line:10009>>0248256987@[Dev:sip:0033214001840 at 217.16.11.101:5060] for L:1.1[Extn:200]
> Do you have any suggestions?



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