search

tmakkonen / sipcmd

sipcmd
327 stars 108 forks source link

Can't call using freevoipdeal.com #22

Open acubino opened 10 years ago

acubino commented 10 years ago

Hi.

I just create a new account with credit on freevoipdeal.com. I test on to my android phone using the native sip configuration, and work ok. But i can't make calls using sipcmd. ./sipcmd -P sip -u "myuser" -c "mypass" -w "sip.freevoipdeal.com:5060" -x "c0034999999999" -o log.log

I also make some test, putting and remove the port. And the execute options. Also, i change the destination phone including 00349999999 +349999999 a phone never ring.

No protocol specified xcb_connection_has_error() returned true Starting sipcmd in debug mode Manager Init initialising SIP endpoint... TestChanAudio TestChanAudio Listening for SIP signalling on 0.0.0.0:5060 SIP listener up registered as sip:username@sip.freevoipdeal.com:5060 Created LocalEndPoint Main

Call

TestPhone::Main: calling "00349999999" using gateway "sip.freevoipdeal.com:5060" at Sat Mar 15 13:33:03 2014

Setting up a call to: sip:00349999999@sip.freevoipdeal.com:5060 LocalEndpoint::MakeConnection LocalEndpointCreateConnection LocalConnection OnIncomingConnection: token=Lcdb4886a2 connection set up to sip:00349999999@sip.freevoipdeal.com:5060 TestPhone::Main: calling "sip:00349999999@sip.freevoipdeal.com:5060" for 0 ... TestPhone::Main: calling "sip:00349999999@sip.freevoipdeal.com:5060" for 10 Problem running command sequence ("c00349999999"): Call: Dial timed out TestPhone::Main: shutting down OnReleased: reason: EndedByLocalUser OnReleased: reason: EndedByLocalUser OnClearedCall ~LocalConnection TestPhone::Main: exiting... Exiting... ~Manager

Because log file is a little big, i upload into pastebin. http://pastebin.com/Z2wuhtdE

Any help?

edholland commented 10 years ago

Hi,

This could be a problem with NAT. You seem to be using a 192.168.x.x address (an internal subnet). Your SIP provider may well not support NAT Proxying. Try using a public IP address and see if you get better results.

Ed

acubino commented 10 years ago

Hi.

Yes, i'm on internal subnet, but no firewall is installed. I can redirect some ports if is needed.

Also, i test from PC using the linux linphone, and works ok and can make the call without problem.

This is the complete config from my voip provider. SIP port : 5060 Registrar : sip.freevoipdeal.com Proxy server : sip.freevoipdeal.com Outbound proxy server : leave empty Account name : your FreeVoipDeal username Password : your FreeVoipDeal password Display name/number : your FreeVoipDeal username or voipnumber Stunserver (option) : stun.freevoipdeal.com

But with linphone i only use the username, password en host.

Need to put anyone on proxy server?.

jaromrax commented 10 years ago

Hi, this is a very interesting piece of software, but with an "account with credit" there may be some catch somewhere, hard to say what level of difficulty it might be. I just want to encourage the creators.

edholland commented 10 years ago

Hi,

I would suggest ensuring that port 5060 is forwarded to your internal IP, but even better i would suggest placing your phone into a DMZ for testing. If we could rule out NAT as a cause of this problem it would allow us to examine some other less likely possibilities.

Ed

On 17 March 2014 07:45, jaromrax notifications@github.com wrote:

Hi, this is a very interesting piece of software, but with an "account with credit" there may be some catch somewhere, hard to say what level of difficulty it might be. I just want to encourage the creators.

Reply to this email directly or view it on GitHubhttps://github.com/tmakkonen/sipcmd/issues/22#issuecomment-37791011 .

acubino commented 10 years ago

Hi.

I will try to test today from my bussiness server, that have public IP and don't have NAT into a datacenter.

If not, i can give my account details for testing, no problem for me, i only make the account with 10 Euro for try this software. I'm trying to make a confirmation CODE for customers when make orders on my page, and this software are easy to make this, only I need to make work.

I will put more info today.

acubino commented 10 years ago

Hi.

I just try with one server connected directly to the net, eth0 have external IP address. Before, i open the port 5060 on the arno-ip-tables, i open UDP and also the TCP.

The error are the same.

I will send in private message to edholland and jaromrax my sip data for testing. I try in 3 pcs, all use Debian. I don't know if the problem are the ports or some on the sip account. With linphone can call using the sip account without problems in one PC, also with the mobile phone.

acubino commented 10 years ago

I see that no are option for send private message in github.

You can send me email to acubino +++ necostek.com and i will reply you with the details.

knireis commented 10 years ago

Anyone got this working? I used several voip providers but no luck sofar.

acubino commented 10 years ago

Hi.

I don't get lucky too. Please tell me if you can use.

knireis commented 10 years ago

I went for another solution: linphone-nogtk package more info: http://www.linphone.org/eng/documentation/guide/linphonecsh-control.html

fralbo commented 5 years ago

Hello, I think I have the same problem using OVH. Did you solve your issue?