similar to: Session Border Controller time...

Asteroid is a SIP denial of service attack tools which affected older versions of Asterisk the Open Source PBX and may affect other products running the SIP protocol. There are thousands of custom (mis)crafted SIP packets which were sent to a older versions of Asterisk that caused errors stopping Asterisk. The packets were crafted based on packetdumps from Wireshark with flags set for

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U)

Apologies for cross posting but some of us aren't on the other list (vice/versa) and thought both groups would benefit. For those familiar with the VoIP Abuse Project, no need to explain the gist of this. I got tired of parsing through the alerts (lists) I receive via email daily. They're long and sometimes I don't have the time to post them all. So for now, posting VoIP Abuse

I've noticed that Asterisk AGI scripts don't terminate when a call is answered. Does anyone know how to do this? I would think that this would be a very big problem, if the scripts stayed in memory, doing nothing, until the call terminates. Not only do you have to have a process for routing each call, but all the previous calls, that are still in progress, also have scripts running. It

As an example....if we have a call that: 1. originates via PSTN line to one of our local DID's in Seattle 2. comes into our Asterisk server in Los Angeles or Denver 3. is routed by Asterisk for termination back to a different Seattle PSTN ....and if our VOIP call termination provider requires (in order to get their best rate) all calls to go through their Nextone

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...

Our setup: We have a DS3 from Global Crossing terminating into a Adtran MX2800 M13 mux. From there, groups of 4 T1's run into T410P digium cards to 7 individual servers. Each trunk is configured as ISDN PRI, B8ZS/ESF, D-channel being chan 96 with B-channels of 1-95 (we're using NFAS). The D channel is up and there are no alarms. We see the connection on the console from the

All, We are in the process of trying to setup our network to use Verizon's SIP "trunking" product. They say that since Asterisk is not on their certified list of approved devices, we need to go through a field trial to get it approved before allowing us to use their service. Where we are at is getting the design approved. We are trying to watch our budget at the same time. We

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

Hello All, "safe_asterisk" just sent me an email saying "Asterisk on bill exited on signal 11. Might want to take a peek.". Looking at the /var/log/asterisk/message doesn't show me anything... This is a fresh installed Asterisk 1.6.2.6 on Ubuntu 9.10 (64-bit) and it is routing calls from Nextone MSW Softswitch to VPS Softswitch... Any reason why Asterisk died?

Hello, we're in process of testing an interconnection with a trans-european carrier. But the carrier wants the SIP signalling to server 1 and the RTP stream to server 2. How do I configure asterisk to work with that type of installation. It seems they are using NexTone as SIP Signaling and RTP servers. Can someone help me??? Regards, Marc -- CTO Marc Storck

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Hi all, I have the following setup: UAs ------------SER ------------------------ ASTERISK ---------------------GNUGK --------------- GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable?

Hey all. We have a DS3 circuit with GBLX split off into 7 systems with a 4 port sangoma card (A104D) in the first 2 systems, and digium T410P cards in the other 5. GBLX numbers their spans from 0 to 3 instead of 1-4 and we have a NFAS configuration with the d-channel on chan 96. All of our systems are running 1.0.7 for stability reasons (and no good time for maintaince, the entire platform

If anyone is looking at upgrading their Digium hardware, or has an excess of any of these cards, please let me know. I am not interested in non-Digium hardware, but thanks for asking :) E100P E400P TE410P I only need one card, in that preference order. Would also prefer if the card is already in Australia. I'm in Sydney/Australia if that matters to you. Regards, Adam -- Adam Goryachev

Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _________________________________________________________________ The

Greetings everybody; I have 2 questions today, but to put them in context, I'll describe the setup first. I am in the process of cnc'ing an old Sheldon metal lathe. These can make large amounts of flying metal chips & stringers when doing what they do, so its paramount that the electronics to run them are either far enough away, or put in a box thats well sealed, with fans in