OpenWrt Forum Archive

Topic: Asterisk on WRT54GS/OpenWRT with 2 phones on a PAP2

The content of this topic has been archived on 13 Apr 2018. There are no obvious gaps in this topic, but there may still be some posts missing at the end.

Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other.

I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is all on the lan side of my gateway/router WRT54G 192.168.1.1


BusyBox v1.00 (2006.11.07-01:40+0000) Built-in shell (ash)


---------------------------------------------------
root@OpenWrt:~# asterisk -r
Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <markster@digium.com>
=========================================================================
Connected to Asterisk 1.2.1 currently running on OpenWrt (pid = 3353)
OpenWrt*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
phone2                     192.168.1.135               5060     Unmonitored
phone1                     192.168.1.135               5060     Unmonitored
2 sip peers [2 online , 0 offline]
OpenWrt*CLI>                                     *****Looks like astersisk ports are both 5060******
                                                                          instead of 5060/5061



Here is some of the errors I am getting


Jan  1 05:10:33 NOTICE[3363]: chan_sip.c:10817 handle_request_register: Registration from 'phone1 <sip:5560@192.168.1.130
>' failed for '192.168.1.135' - Not a local SIP domain

Jan  1 05:11:03 NOTICE[3363]: chan_sip.c:10817 handle_request_register: Registration from 'phone2 <sip:5561@192.168.1.130
>' failed for '192.168.1.135' - Not a local SIP domain



(Not a local SIP domain)  so asterisks sees the extensions but they are not under the sip domain??
My extensions.conf file is blank as I don't know how to set up extensions for PAP2 with 2 phones

in my sip.conf file

autodomain=yes


OpenWrt*CLI> sip show settings

Global Settings:
----------------
  SIP Port:               5060
  Bindaddress:            0.0.0.0
  Videosupport:           No
  AutoCreatePeer:         No
  Allow unknown access:   Yes
  Promsic. redir:         No
  SIP domain support:     Yes
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm          asterisk
  Realm. auth:            No
  User Agent:             Asterisk PBX
  MWI checking interval:  10 secs
  Reg. context:           (not set)
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  IP ToS:                 0x0
  OSP Support:            No
  SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
  Codecs:                 none
  Relax DTMF:             No
  Compact SIP headers:    No
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   No
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes

Default Settings:
-----------------
  Context:                default
  Nat:                    RFC3581
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               (Defaults to English)
  Musicclass:             default
  Voice Mail Extension:   asterisk

----
OpenWrt*CLI>

Gazoo2 wrote:

Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other.

Well, one thing's for sure: this is not an issue relating to the development of the OpenWrt platform itself.

If you have a working installation of Asterisk - and I'd recommend zandbelt's recent packages - then this is a user's question about Asterisk.

Jan  1 05:10:33 NOTICE[3363]: chan_sip.c:10817 handle_request_register: Registration from 'phone1 <sip:5560@192.168.1.130>' failed for '192.168.1.135' - Not a local SIP domain

Jan  1 05:11:03 NOTICE[3363]: chan_sip.c:10817 handle_request_register: Registration from 'phone2 <sip:5561@192.168.1.130>' failed for '192.168.1.135' - Not a local SIP domain

I suggest you post those errors, plus your entire sip.conf file, to the asterisk-users@lists.digium.com mailing list.

ok well thank you for pointing me in the right direction, I will try the mailing list and other forums on asterisks, looks like the asterisk 1.4 release has a gui that might be working with it too.


Thank you

The discussion might have continued from here.