October 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

B-leg continuously Ringing
by Murugan Pandian 03 Oct '16

03 Oct '16

Hi, When i try to make outbound PSTN call using Gateway ,My phone ring continually i am getting ring until unless i disconnect call from a leg,if i disconnect call from my phone again i am getting ring to my phone . Bellow is my dial plan <extension name="outbound call"> <condition field="destination_number" expression="^919xxxxxxxx$"> <action application="bridge" data="sofia/gateway/flowroute/919xxxxxxxx"/> <action application="hangup"/> </condition> </extension> Regard's, Pandian

2 1

Bridging calls to other server
by Caio Assis 03 Oct '16

03 Oct '16

Good afternoon. The situation I'm facing is this: I have a server that has 2 network interfaces (eth0 and eth1), the former with an external static IP set, and the ladder with a internal IP set. I have one more server on my private network that bridges calls to my gsm gateway (which means it's set with a internal IP as well), and I have a VoIP account to bridge other calls. The problem I'm facing is: when I try to bridge a call to my gateway, in my private network, the IP freeswitch is sending is the external IP, so the call is denied by my gateway. With the VoIP I don't have any problem, since the IP external meets my voip's external IP. Just to give you an example, it's like this. Main Server (eth0: 192.168.1.50, eth1: 123.123.123.123) Main Server bridges calls to Gateway (192.168.1.255). The gateway recieves an request coming from 123.123.123.123, and it's set to decline external IP, so the call gets a forbidden signal. Main Server bridges calls to VoIP provider(177.177.177.177). VoIP accepts my calls, since the request it receives is coming from the external IP. I need Freeswitch to send the call to the gateway using the internal IP and voip using external IP. I use asterisk, and i set a parameter brindaddr=0.0.0.0 to accept request coming from both internal and external IPs. All of my asterisk configurations work fine. Thanks.

3 3

Registration Unauthorized then 481 Call Does Not Exist
by Tanim 03 Oct '16

03 Oct '16

Hi, When I try to registration its giving following log. Its only happen a specific dialer but that dialer also work other softswitch. It's getting problem only with freeswitch. My freeswitch version is 1.6.9 I can't figure out why its happen. How can I overcome this issue and is there any sip message issue or not? So that I can ask those dialer company to fix their issue. Thanks in advance. REGISTER sip:103.12.214.115:5060 SIP/2.0 Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea Max-Forwards: 70 From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060> Call-ID: smvpcaakieaaaaaaaaa CSeq: 1 REGISTER Contact: <sip:wmaeaccba@50.97.212.23:1310> Expires: 3600 Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe Call-ID: smvpcaakieaaaaaaaaa CSeq: 1 REGISTER User-Agent: WHaT Rocks! Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY Supported: precondition, 100rel, path, replaces WWW-Authenticate: Digest realm="103.12.214.115", nonce="2a9726d3-fac2-4d7b-9773-771b60a35e7a", algorithm=MD5, qop="auth" Content-Length: 0 ACK sip:103.12.214.115:5060 SIP/2.0 Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea Max-Forwards: 70 From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe Call-ID: smvpcaakieaaaaaaaaa CSeq: 1 ACK Contact: <sip:wmaeaccba@50.97.212.23:1310> Content-Length: 0 REGISTER sip:103.12.214.115:5060 SIP/2.0 Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea Max-Forwards: 70 From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe Call-ID: smvpcaakieaaaaaaaaa CSeq: 2 REGISTER Contact: <sip:wmaeaccba@50.97.212.23:1310> Expires: 3600 Authorization: Digest username="12345678",realm="103.12.214.115",nonce="2a9726d3-fac2-4d7b-9773-77 1b60a35e7a",uri="sip:103.12.214.115:5060",algorithm=MD5,response="73c3beac11 531911319f05807714c047" Content-Length: 0 SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe Call-ID: smvpcaakieaaaaaaaaa CSeq: 2 REGISTER User-Agent: WHaT Rocks! Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY Supported: precondition, 100rel, path, replaces Content-Length: 0 ACK sip:103.12.214.115:5060 SIP/2.0 Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea Max-Forwards: 70 From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe Call-ID: smvpcaakieaaaaaaaaa CSeq: 2 ACK Contact: <sip:wmaeaccba@50.97.212.23:1310> Content-Length: 0 Regards Minat

3 3

Are is it bug?
by Sergey Safarov 01 Oct '16

01 Oct '16

I cannot resolve IPv6 hostname via FS console. Are it is bug? [root@localhost6 ~]# nslookup -type=aaaa ipv6.cybernode.com Server: ::1 Address: ::1#53 Non-authoritative answer: ipv6.cybernode.com has AAAA address 2001:470:1:1b9::31 [root@localhost6 ~]# fs_cli -x "host_lookup ipv6.cybernode.com" -ERR no reply [root@localhost6 ~]#

4 8