vicidial.org

Hi,

I want to logon SIP phone into vicidialnow server using internet. I have forwarded port 5060 into my myrouter to vicidialnow server. Now when I logon sip phone using IP of router, it log me in and I can see phone logged into asterisk CLI. When I ring from internal extension to this phone it rings and connects but no converstaion and remote phone is automatically disconnected after some time. When I try to ring from remote phone to any local extension it can't dial. Any help to do this?

Best regards,

you need to add ports for rtp from 10000 to 20000

Hi Op3r,

These ports are already opened and directed to VICIDIALNOW server.

Best regards,

Op3r wrote:you need to add ports for rtp from 10000 to 20000

um ... guess: codecs? have you watched your "sip debug" to see what is happening during these attempted calls?

Let see the output of your Asterisk CLI.

Hi Gardo,

During call CLI is

Code: Select all

    -- Executing [149@default:1] Dial("SIP/cc140-b70232f0", "SIP/cc149") in new stack
    -- Called cc149
    -- SIP/cc149-08639fd0 is ringing
    -- SIP/cc149-08639fd0 answered SIP/cc140-b70232f0
    -- Packet2Packet bridging SIP/cc140-b70232f0 and SIP/cc149-08639fd0


After some time around a minute call is automatically diconnected with this CLI output

Code: Select all

[Aug 15 20:58:47] NOTICE[2618]: chan_sip.c:15732 do_monitor: Disconnecting call 'SIP/cc149-08639fd0' for lack of RTP activity in 62 seconds
  == Spawn extension (default, 149, 1) exited non-zero on 'SIP/cc140-b70232f0'
    -- Executing [h@default:1] DeadAGI("SIP/cc140-b70232f0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----67-----62") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----67-----62 completed, returning 0


Best regards,

gardo wrote:Let see the output of your Asterisk CLI.

Hi William,

CLI with sip debug while call was made is

Code: Select all

<------------>
    -- Executing [149@default:1] Dial("SIP/cc140-086b1d70", "SIP/cc149") in new stack
Audio is at 192.168.1.250 port 15196
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 92.11.233.206:5060:
INVITE sip:cc149@92.11.233.206:40964 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK305137a4;rport
From: "Ext 140" <sip:cc140@192.168.1.250>;tag=as054375b1
To: <sip:cc149@92.11.233.206:40964>
Contact: <sip:cc140@192.168.1.250>
Call-ID: 121319a8183cbb0b10be03a27201afbc@192.168.1.250
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Ext 140" <sip:cc140@192.168.1.250>;privacy=off;screen=no
Date: Sat, 15 Aug 2009 21:09:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 2521 2521 IN IP4 192.168.1.250
s=session
c=IN IP4 192.168.1.250
t=0 0
m=audio 15196 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

----- Called cc149


And after about a minute call is auto disconnected and debug info is

Code: Select all

<------------->
[Aug 15 21:12:15] NOTICE[2618]: chan_sip.c:15732 do_monitor: Disconnecting call 'SIP/cc149-08695818' for lack of RTP activity in 61 seconds
Scheduling destruction of SIP dialog '5723140e5e15288753a00f6e65b32916@192.168.1.250' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:cc149@92.11.233.206:5060> for address/port to send to
set_destination: set destination to 92.11.233.206, port 5060
Reliably Transmitting (NAT) to 92.11.233.206:5060:
BYE sip:cc149@92.11.233.206:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK69230e1a;rport
From: "Ext 140" <sip:cc140@192.168.1.250>;tag=as54e387c5
To: <sip:cc149@92.11.233.206:40964>;tag=3401277448
Call-ID: 5723140e5e15288753a00f6e65b32916@192.168.1.250
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Ext 140" <sip:cc140@192.168.1.250>;privacy=off;screen=no
Content-Length: 0


---
  == Spawn extension (default, 149, 1) exited non-zero on 'SIP/cc140-086b1d70'
    -- Executing [h@default:1] DeadAGI("SIP/cc140-086b1d70", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----65-----61") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----65-----61 completed, returning 0
Scheduling destruction of SIP dialog '1800b72ede39f254' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:cc140@192.168.1.251:8268> for address/port to send to
set_destination: set destination to 192.168.1.251, port 8268
Reliably Transmitting (NAT) to 192.168.1.251:8268:
BYE sip:cc140@192.168.1.251:8268 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK09dd13f1;rport
From: <sip:149@192.168.1.250>;tag=as7290c3af
To: Ext 140<sip:cc140@192.168.1.250>;tag=d2772241
Call-ID: 1800b72ede39f254
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0



Best regards,

williamconley wrote:um ... guess: codecs? have you watched your "sip debug" to see what is happening during these attempted calls?

Checkout this link: http://www.asteriskguru.com/tutorials/s ... erisk.html . Looks like your problem is NAT related.

Hi Gardo,

As far as NAT problem is concerned, I have tested this on both ends. On the VICIDIAL end we are dialling local extensions as outbound numbers without any problem. For other end, It's my home, I have setup four voip accounts and they all are dialling and receiving calls without any problem. If these two setup independently working fine where the NAT is stopping them to communicate with each other?

Best regards,

gardo wrote:Checkout this link: http://www.asteriskguru.com/tutorials/s ... erisk.html . Looks like your problem is NAT related.

i think you answered your own question. it's not that the asterisk server is incapable of traversing the NAT ... it's simply not been told to for this connection.

Hi William,

What to try now. I did everything which I can. Could you please suggest anything to try?

Best regards,

williamconley wrote:i think you answered your own question. it's not that the asterisk server is incapable of traversing the NAT ... it's simply not been told to for this connection.