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Summary:ASTERISK-29375: I can not make a call via zoiper (VOIP / SIP App)
Reporter:Karna Sai (Karna Sai)Labels:
Date Opened:2021-03-30 01:53:53Date Closed:2021-03-30 01:53:54
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_pjsip Channels/chan_sip/General
Versions:18.2.2 Frequency of
Occurrence
Related
Issues:
Environment:Windows Subsystem for Linux (UBUNTU)Attachments:
Description:5 or 6 day's working very well in inter calling purpose. But when i use to do as an out bounding call's purpose, then i don't know what happened on asterisk, it will not connect  in Zoiper Always says this

You have an error associated with this account
Request Timeout (code: 408)
Username 2
Account name2@192.168.1.5
Timestamp2021-03-30T11:57:38+05:30
Protocol5
Error Layer(3)
Error Code408
Error TextRequest Timeout

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CLI> module load chan_sip.so
Unable to load module chan_sip.so
Command 'module load chan_sip.so' failed.
[Mar 30 11:55:59] WARNING[341]: loader.c:1769 load_resource: Module 'chan_sip.so' already loaded and running.
BIMx*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
2                         (Unspecified)                            D  Auto (No)  No             0        Unmonitored
3                         (Unspecified)                            D  Auto (No)  No             0        Unmonitored
5                         (Unspecified)                            D  Auto (No)  No             0        Unmonitored
7001/7001                 (Unspecified)                            D  Auto (No)  No             0        Unmonitored
7004                      (Unspecified)                            D  Auto (No)  No             0        Unmonitored
5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 5 offline]


---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
$ sudo vim /etc/asterisk/sip.conf
[geiineral]
context=internal
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
alwaysauthreject=yes
srvlookup=yes
qualify=yes
disallow=all
allow=ulaw
session-timers=refuse
bindport=5060
localnet=192.168.1.0/255.255.255.0
nat=force_rport,comedia

[5]
type=friend
host=dynamic
disallow=all
secret=123
context=internal
allow=ulaw,alaw

[2]
type=friend
host=dynamic
secret=123
context=internal
disallow=all
allow=ulaw,alaw

[3]
type=friend
host=dynamic
secret=123
context=internal
disallow=all
allow=ulaw,alaw

[7004]
type=friend
host=dynamic
secret=123
context=internal
disallow=all
allow=ulaw,alaw

[7001]
type=friend
host=dynamic
secret=123
context=internal
disallow=all
allow=ulaw,alaw

---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
$ sudo vim /etc/asterisk/extensions.conf

[internal]
exten => 5,1,Answer()
exten => 5,2,Dial(SIP/5)
exten => 5,5,Hangup()
exten => 5,3,Playback(vm-nobodyavail)
exten => 5,4,VoiceMail(5@main)

exten => 2,1,Answer()
exten => 2,2,Dial(SIP/2,60)
exten => 2,3,Playback(vm-nobodyavail)
exten => 2,4,VoiceMail(2@main)
exten => 2,5,Hangup()

exten => 3,1,Answer()
exten => 3,2,Dial(SIP/3,60)
exten => 3,3,Playback(vm-nobodyavail)
exten => 3,4,VoiceMail(3@main)
exten => 3,5,Hangup()

exten => 7004,1,Answer()
exten => 7004,2,Dial(SIP/7004,60)
exten => 7004,3,Playback(vm-nobodyavail)
exten => 7004,4,VoiceMail(7004@main)
exten => 7004,5,Hangup()

exten => 7001,1,Answer()
exten => 7001,2,Dial(SIP/7001,60)
exten => 7001,3,Playback(vm-nobodyavail)
exten => 7001,4,VoiceMail(7001@main)
exten => 7001,5,Hangup()

exten => 8001,1,VoicemailMain(5@main)
exten => 8001,2,Hangup()

exten => 8002,1,VoicemailMain(2@main)
exten => 8002,2,Hangup()

exten => 8001,1,VoicemailMain(3@main)
exten => 8001,2,Hangup()

exten => 8001,1,VoicemailMain(7004@main)
exten => 8001,2,Hangup()

exten => 8001,1,VoicemailMain(7001@main)
exten => 8001,2,Hangup()
Comments:By: Asterisk Team (asteriskteam) 2021-03-30 01:53:54.364-0500

We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

If this issue is actually a bug please use the Bug issue type instead.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
By: Asterisk Team (asteriskteam) 2021-03-30 01:53:56.132-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/].