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Re: [Linphone-users] Unable to call local extension if server port is no

From: Tracy Briscoe
Subject: Re: [Linphone-users] Unable to call local extension if server port is not 5060
Date: Mon, 6 Feb 2017 20:02:11 +0000

Hi Walter


Our PABX uses a custom port for SIP extensions.  Below is a screenshot showing the setting I use on my machine.





Mr Tracy Briscoe

Network Engineer


St Peter's School





From: Linphone-users [mailto:linphone-users-bounces+address@hidden On Behalf Of Walter Stockinger
Sent: Friday, 3 February 2017 10:11 p.m.
To: address@hidden
Subject: [Linphone-users] Unable to call local extension if server port is not 5060




My problem:

I cannot call a local extension, because my server is set to listen on a port different from 5060 (I have changed the port since I registered a very large number of registration attempts from unknown computers)



Have several local and external clients registered on my asterisk server to call each other


My setup: 

DynDNS: mydomain.ab (functional)

Portforwarding on my router: 6950 to asterisk server local IP (and all other asterisk ports, as needed: tested and functional)

asterisk server (computer 1, raspberry) UDP listening port = 6950 (functional)

client 1 (computer 2, linphone 3.6.1, OS= ubuntu 16.10), extension 3000

client 2 (computer 3, linphone, OS=Android 5.0) extension 1000


Both clients register successfully with the asterisk server using

sip identity: sip:address@hidden

sip proxy: sip:mydomain.ab:6950


Working: Call from Android to Linux:

when i call from my android device extension '3000' (client 1) , i get a working connection


Not Working:

when I call from linux to Android I do not get a connection. Looking in the debug window I can see that name-resolution works (of course, I was also able to register with my asterisk server) BUT the call is placed to PORT 5060 (my IP and domain name are changed)


message: Flattened number is '1000'
message: Local interface to reach is
message: Notifying all friends that we are in status 5
message: Call 0x29e8820: moving from state LinphoneCallIdle to LinphoneCallOutgoingInit
message: Setting DSCP to 46 for audio stream.
warning: cannot set noise gate mode to [0] because no volume send
message: Setting DSCP to 46 for video stream.
message: ms_filter_link: MSRtpRecv:0x28fbc90,0-->MSVoidSink:0x288dd80,0
message: Priority used: 99
message: Audio MSTicker setpriority() failed: Permission denied, nevermind.
message: allocating transaction resource 16 344469379
message: allocating ICT context
message: eXosip_dnsutils_naptr_lookup: About to ask for ' NAPTR'
error: eXosip_dnsutils_naptr_lookup: res_query failed (' NAPTR')
message: Call 0x29e8820: moving from state LinphoneCallOutgoingInit to LinphoneCallOutgoingProgress
message: eXosip option set: dns cache used:mydomain.ab -> 123.456.789.012
message: DNS resolution with 123.456.789.012:5060
message: getaddrinfo returned: 123.456.789.012 port 5060
message: Message sent: (to dest=123.456.789.012:5060)


--->This seems wrong, it should use 123.456.789.012:6950

---> I believe my settings are ok, because registration with the server works


asterisk console: 'sip show peers:'


Name/username              Host                                    Dyn Forcerport ACL Port     Status     
1000/1000                  123.456.789.012                           D   N             55107    Unmonitored
3000/3000                  123.456.789.012                           D   N             60294    Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]



Many thanks for your help


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