search for: sip

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 sip "sip" <sip> s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-000000fd si...

...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 -- Registered SIP '7052823582' at 64.101.221.250 port 18641 expires 60 -- Registered SIP '502' at 64.101.221.250 port 18643 expires 60 -- Registered SIP '502' at 64.101.221.250 port 18647 expires 60 -- Reg...

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 canreinvite=yes context=from-sip callerid=Ronald Wiplinger <6003> dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p T...

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...

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 m...

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 to look for the Via header. Registration works like a charm, and if I dial in from the PSTN to the ATA the phone rings properly. However, it doesn't seem to be able to start an RTP s...

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 why even when SIP has registered I do not see incoming calls? Thanks,...

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...

-----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 = stun.exiga.net insecure=port,invite ; required for incoming ekiga.net calls /etc/asterisk/extensions.conf: [from-internal] ... exten => _8.,1,Dial(SI...

...op("My CalleID: <<<<<<<=".$ani); $agi->set_variable("CALLERID(num)","01133200274"); $ani = $agi->request['agi_callerid']; $agi->noop("My CalleID: <<<<<<<=".$ani); $agi-> exec('Dial',"SIP/00918885268942 at sip.trunk.gradwell.com,60,r"); //$agi-> exec('Dial',"SIP/00918885268942 at voipon,60,r"); ?> And CLI> == Using SIP RTP CoS mark 5 -- Executing [101 at outbound:1] Answer("SIP/2209-000026d3", "") in new stack -- Executi...

...: - 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 observed that all sip calls are closed exactly after 32s. call is disconnected on calling side as well... seems to be a timeout issue. here i have some debug logs. I see lot of requests from asterisk to sipgate.de, which are not answered. but communication is going fine in both directions (otherwise registration wou...

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.. registrations were ok, etc.. but now in head it's borked. verbose = 30 debug...

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 fam...

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...

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=dynami...

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 cal...

...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 results in dropping both calls. I've attached a sanitized sip trace from the snom phone for your perusal. Thanks for any help you can offer. Brian ### START SIP TRACE ### Sent to udp:192.168.0.129:5060 at 14/12/2004 18:21:29:500 (593 bytes): REGISTER sip:192.168.0.129 SIP/2.0 Via: SIP/2.0/UDP 192.168.102.70:5060;branch=z9hG4bK-wg4ok3zkt573;rport From: &q...

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 a helping hand with this matter or offer any insight on how to further debug. I've gone as far...

...r (linux) clients. On kphone the interface's register result is 'bad password', on zoiper registration continues indefinitely but after the first request it is ignored by asterisk due to being duplicate, after a time it fails silently. The debug log: [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: = No match Their Call ID: NDcxYjAyNTc4ZDQwZjZhMzM5OGE0MWYxYjg0YzZhZDk. Their Tag a9a71835 Our tag: as0a26e7a5 [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Allocating new SIP dialog for ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg. - REGISTER (No RTP) [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: **** Re...