search for: sipping

Hi Everyone, Like always, here are IPs from China that try to hack an Asterisk server. Can someone please explain what is happening or what the hacker is trying to reach: 02/10/2010 11:10 SIP/113.105.152.51-000000fb sip "sip" <sip> s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-000000fe sip "sip" <sip> s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-000000fc

I have about 20+ phones on a server, all set for registry expiry 1 min. But only this one, with 2 accounts, keeps re-registerting itself. All the time this is what I see on asterisk CLI and it is kind of annoying. What only this phone does this and no other. Its on a remote location. All phones are Grandstream GXP-2000. -- Registered SIP '502' at 64.101.221.250 port 18639 expires 60

Hey All: Question; when using a virtual IP on an Asterisk server, I am having trouble getting sip user to register to the ViP. They are able to register with the true IP, just not the virtual. It seems Asterisk is rejecting the SIP invite, register, etc (like it's not destined for this server) I've added all the IP's to the domain listing in sip.conf and in the Asterisk console a

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

Hello, i've recently configured my asterisk for internal sip calls. while testing, i noticed that 1 out of 10 calls works.. at first i thought my router dropping packets around the way as it were a bottle neck.. so i've added a switch. once i tested again same prob occurs... im using xlite as a softphone on clients pc and centos server on a dedicated machine. at times the phone call

Hi guys, sorry to be iterating this on the list once more, but I'm not able to get this stuff to work as I'd expect. So far, I've always managed to keep it out of NAT environments :-> My home LAN is NATed by a simple Draytek router. In the home LAN is an ATA186 with SIP. On the internet (public) is an Asterisk server. I have nat=yes in the sip.conf and the connectmode is set

Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one. My outgoing SIP and other channels work fine but the incoming/inbound SIP call goes straight to Broadvoice voicemail. I see that Broadvoice is registered when I look at the SIP registry. I have turned on SIP Debug and it is below. Anyone know

I've attached two SIP debugs in reference to bug #116. They are from today's CVS build. 1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the call. After which, SIP(2) rings for about 30 seconds then stops. 2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before the call is answered. SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

hi, let me explain in detail, what i have configured and what is happening now: 1st router w724v (Deutsche Telekom AG): - port forwarding, everything to destination port 51000-55999 to device with ip 192.168.2.50 (interface of 2nd router) 2nd router Bintec RS353j): - configured NAT, everything to port 51000-55999 to device 192.168.3.99 (same ports) other direction is totally open. I

I just updated one of my stable asterisk systems to head to test it out.. and I'm receiving a interesting log message now in asterisk.. Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce received from '<sip:3034585725@voip.livewirenet.com;user=phone>' (one line per registration) I'm using an AudioCodes mp108.. it worked fine with the latest stable..

I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer.

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

hi all when starting kphone, it tries to register with asterisk but fails after a while. The SIP entry in * for this user is below. This is identical to the other SIP entries. The other SIP clients are MSN messenger plus one snom. these work fine. See SIP debug output attached as 'screen-exchange' thanks roy [roy] type=friend ;insecure=yes username=roy ;secret=password host=dynamic

OK, here goes my next problem. I have puchased a DID which I can connect to via SIP I have been given the following details: Username: uka1xxxxxx Password: 1000xxxxxx Server: brxxxx.net:5160 My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT) The other end is a Cisco AS5300 (NO NAT) I can register with the Cisco with no problem. When I dial the DID it sends the call to my asterisk

Hello List- I'm having a problem getting snom 190 phones to transfer a call to another local extension. Here is the scenario: A call (call1) comes in from the PSTN to (exten1). (via pri, if that matters) Another call (call2) comes in to (exten1). (call1) is put on hold while (call2) is answered. (call2) is then transferred to (exten2) using the "Xfer" button on the snom phone. This

Hi all, I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my existing Cisco 7960G handset(s). I've tried multiple installs of asterisk 1.4.2 with multiple handsets and SIP will not authorize my phone. I'll include some verbose log messages below to show a VALID registration and one where I'm having difficulty registering the phone. Thanks to anyone who can lend

Hi all, Asterisk is great but I'm having issues with setting up realtime for our call center, which is needed for login integration with the rest of our applications (telephonists' web interface, etc.). I have reviewed a large number of previous posts to the mailing list and the voip-info wiki to no avail. Setup is as follows: Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ /