Re: [asterisk-users] Portech MV-374 does not register behind NAT
Jared, the Portech SIMbox is registering to a DNS name. The firewall is off and the NAT is a Zyxel NBG-419 router. No mather what port I set, It is not working : --- SIP read from my_public_ip:5070 --- REGISTER sip:sip.sipserver.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.1.25:5070;branch=z9hG4bK4465928ede From: SIM 1-1 sip:sims...@sip.sipserver.tld;tag=6672a4c3 To: SIM 1-1 sip:sims...@sip.sipserver.tld Call-ID: 277210db31b0ae184675c3245d5e9...@192.168.1.25 Contact: sip:sims...@192.168.1.25:5070 CSeq: 56 REGISTER Max-Forwards: 70 Expires: 60 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE User-Agent: Mv-37x (904290) Content-Length: 0 - [May 2 10:38:51] --- (12 headers 0 lines) --- [May 2 10:38:51] Using latest REGISTER request as basis request [May 2 10:38:51] Sending to 192.168.1.25 : 5070 (no NAT) [May 2 10:38:51] --- Transmitting (NAT) to my_public_ip:5070 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.25:5070;branch=z9hG4bK4465928ede;received=my_public_ip From: SIM 1-1 sip:sims...@sip.sipserver.tld;tag=6672a4c3 To: SIM 1-1 sip:sims...@sip.sipserver.tld Call-ID: 277210db31b0ae184675c3245d5e9...@192.168.1.25 CSeq: 56 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 [May 2 10:38:51] --- Transmitting (NAT) to my_public_ip:5070 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.25:5070;branch=z9hG4bK4465928ede;received=my_public_ip From: SIM 1-1 sip:sims...@sip.sipserver.tld;tag=6672a4c3 To: SIM 1-1 sip:sims...@sip.sipserver.tld;tag=as12309a5f Call-ID: 277210db31b0ae184675c3245d5e9...@192.168.1.25 CSeq: 56 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=103001vc, nonce=7c2508cc Content-Length: 0 Is it normal that there is a Via-header sent from the SIMbox with its local IP-address in it ?? Is it normal that SIP read from my_public_ip:5070 has the same port number as the SIP-account (simsim1) 192.168.1.25:5070 ?? Could it be that NAT is not working correctly in my router ?? Jonas. On 04/22/2010 05:53 PM, Jared Smith wrote: Is the device registering to an IP address, or do a DNS name? What type of NAT firewall are you using? This reminds me of a problem I had years ago with a Cisco PIX firewall, where it would rewrite IP addresses in the SIP Request URI, causing the authentication to fail. One solution was to have it register to a fully-qualified domain name instead of an IP address, so that the Request URI wouldn't get overwritten. It's certainly worth a shot... -- Jared Smith Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
my advise check your internet connection on the remote location and keep a ping from that network to your server running all the time to check for time outs. How can i log a continuous ping test to a file and include the date and time of each ping? I've found this bash code but it only logs once the tests have all finished. If I set it to continuous and then kill the task when I want to view the pings, it doesn't record the data. #!/bin/sh NOW=$(date +%T %m/%d/%Y) PING=$(ping -qc 5 example.com | grep '5 packets') echo $NOW: $PING /home/matt/ping.log exit 0 Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote: snip How can i log a continuous ping test to a file and include the date and time of each ping? Try this: #!/bin/sh for (( ; ; )) do NOW=$(date +%T %m/%d/%Y) PING=$(ping -qc 1 example.com) echo $NOW: $PING pinger.log done exit 0 You can then monitor the log file using: $ tail -f pinger.log You will need to use ^C to kill the script. Hope this helps. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
On 05/02/2010 02:59 AM, James Lamanna wrote: It seems that the PAP2T does support TFTP and an XML-based config for deployments... I've used both the Grandstream 286 and the Linksys PAP2T. I have been able to get some limited faxing to work using T30 with a PAP2T. Configuration and provisioning of the Linksys is very easy through either the web GUI or XML configuration files, which can be transferred through TFTP or HTTP. I can only hope that Cisco will update the firmware of the PAP2T to support T38 one day... They've made it pretty clear that isn't going to happen. I suspect its a lack of resources - memory and processing power - rather than being bloody minded and trying to maintain some differentiation for the SPA2102. They can only do one port of G.729, and I assume that is more than just saving on per port patent licence fees. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security tests
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Steve. On Fri, Apr 23, 2010 at 22:38:49 -0300, Steve Totaro wrote: Perhaps it was not very clear, but yes, I was talking about this. I believe that I found the cause of the problem. The cause by which I was not seeing VoIP traffic between 10.1.0.38 and 10.1.0.65 is because there is no direct traffic among them but that is between each party and the Asterisk server :-) So using ettercap with de IP of Asterisk server and 10.1.0.65 I can now capture and play calls from this IP to 10.1.0.38 or vice versa. But I'm noticing that playing from Wireshark it can be heard delayed. Is it normal to happen? On the other hand, I had to change the order of preference of the codecs in the sip.conf so that G711 is preferred over GSM, because it was configured in a reverse order of preference and I see that the RTP player of Wireshark does not support GSM. Do you know any way to play GSM directly from the captured packets? How did you place your virtual listening machine into the network, is it connected to an old hub, or a switch, or the mirroring port of a switch, or does it use the same NIC (and computer) as the softphone? You will first need to get in between the two endpoints in order to be able to capture that point-to-point RTP traffic - there are normal and malicious ways to achieve that. I have a switch that connects to the phone (10.1.0.38), PC with softphone (10.1.0.65), the Asterisk server and a VMHost that has the virtual machine where I use ettercap and tcpdump. Check out *Cain* *Abel* http://www.oxid.it/ and OrecX http://www.orecx.com/web/products-orekagpl.php. Oreca will run just fine on your Asterisk box. I had read something about Cain Abel. I will try reproducing the capture in an equipment with Windows using Cain Abel because here, in my house, I only have GNU/Linux and OpenBSD. About the delayed reproduction on Wireshark, is it something that also you have experimented? I am not sure what kind of security audit you are trying to do. What you propose is simple and simply the way things work, it is not security. This is initially for an presentation about security in the course of Distributed Systems. My idea was to speak on attacks and countermeasures in VoIP. On the other hand, they are asking to me to make a practical demonstration of the countermeasures. Although a direct form to avoid this is using VLANs, it seems that the idea is to demonstrate the countermeasures with some software. Then I was thinking about trying, for example, SRTP or SIP over TCP/TLS. Do you have implemented it on Asterisk 1.4? In such case, could you recommend some good document on this matter? I'm using at the moment Asterisk 1.4.24.1. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvdgVkACgkQZpa/GxTmHTfukwCgg3hf2mBvZHXqiEjk2JAvI1dW +6sAoI/bDWWfEeWvY9InSO1Pi0381uNu =hHoH -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
Hi Bob, Thanks for that. Is there any way I can make the task run in the background and free up the console? Also so that I can disconnect my ssh session without losing the task. Thanks Dan Sent from my Windows Mobile® phone. -Original Message- From: Bob Smither smit...@c-c-i.com Sent: 02 May 2010 14:04 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Calls Dropping On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote: snip How can i log a continuous ping test to a file and include the date and time of each ping? Try this: #!/bin/sh for (( ; ; )) do NOW=$(date +%T %m/%d/%Y) PING=$(ping -qc 1 example.com) echo $NOW: $PING pinger.log done exit 0 You can then monitor the log file using: $ tail -f pinger.log You will need to use ^C to kill the script. Hope this helps. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Saturday, May 01, 2010 9:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286 It seems that the PAP2T does support TFTP and an XML-based config for deployments... I've used both the Grandstream 286 and the Linksys PAP2T. I have been able to get some limited faxing to work using T30 with a PAP2T. Configuration and provisioning of the Linksys is very easy through either the web GUI or XML configuration files, which can be transferred through TFTP or HTTP. I can only hope that Cisco will update the firmware of the PAP2T to support T38 one day... Related to this: what is a good ATA with a reasonable price that works correct for sending and receiving a fax? The faxes we have (on multiple locations) require a analog connection. We have sending a fax working with the PAP2T and incoming faxes are send to an email address (as a PDF). Regards, Mark -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
Hi, try using screen : http://www.rackaid.com/resources/linux-screen-tutorial-and-how-to/ I think it's the best way of doing this. Regards, Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ Le 2 mai 10 à 15:52, Dan Journo a écrit : Hi Bob, Thanks for that. Is there any way I can make the task run in the background and free up the console? Also so that I can disconnect my ssh session without losing the task. Thanks Dan Sent from my Windows Mobile® phone. -Original Message- From: Bob Smither smit...@c-c-i.com Sent: 02 May 2010 14:04 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Calls Dropping On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote: snip How can i log a continuous ping test to a file and include the date and time of each ping? Try this: #!/bin/sh for (( ; ; )) do NOW=$(date +%T %m/%d/%Y) PING=$(ping -qc 1 example.com) echo $NOW: $PING pinger.log done exit 0 You can then monitor the log file using: $ tail -f pinger.log You will need to use ^C to kill the script. Hope this helps. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI == DeadAGI
Hi Luki: 于 2010年05月01日 06:03, Luki 写道: The good news is, we run tens of thousands of calls every day through this box and about half of them spit out this warning, but it never caused any problems for over a year. Thus this warning is probably safe to ignore. We run tens of thousands of call every day too. Call is controlled by AGI , and the asterisk version is 1.2.24. I find memory leak in asterisk. After serveral weeks, the memory used by asterisk will reach 1.2 GB or higher. Each time I have to restart to asterisk, and the memory leak will repeat. I wonder if you have the memory leak problem? Which version asterisk you use? Thanks for reply. Regards, Chen Xueqin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI == DeadAGI
We run tens of thousands of call every day too. Call is controlled by AGI , and the asterisk version is 1.2.24. I find memory leak in asterisk. After serveral weeks, the memory used by asterisk will reach 1.2 GB or higher. Each time I have to restart to asterisk, and the memory leak will repeat. I wonder if you have the memory leak problem? Which version asterisk you use? Thanks for reply. No, no memory leak here. Memory usage: 58 MB after: System uptime: 25 weeks, 5 days, 12 hours, 39 minutes, 52 seconds. Asterisk version 1.4.23.1 (with about 25 custom, in-house patches). This particular box only handles signaling from a dozen static peers. No registration, no media (directrtpsetup=yes), no NAT, no transcoding, no MOH... but it does use realtime for SIP and IAX, and AGI and DeadAGI for routing. Luki -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] working example of t38 fax w/ 1.6.2?
I can't get a test T.38 fax between 2 1.6.2 machines, using app
_fax and spandsp pre17 and 20100501. The machines can't seem to get
connected.
send side extensions.conf:
[fax-tx-test]
exten=s,1,NoOp(Context fax-tx-test)
exten=s,n,SendFAX(${FaxFile}.tif)
exten=s,n,HangUp()
exten=h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR} FAXMODE:
${FAXMODE})
Channel:SIP/side-sip-fax
Context:fax-tx-test
Extension:s
Priority:1
Set:FaxFile=/var/spool/asterisk/fax/20091113_1455
receive side:
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
exten = s,n,ReceiveFAX(${FAXFILE}.tif)
exten = s,n,Hangup()
There's a bunch more stuff at https://issues.asterisk.org/view.php?id=17105
But does anyone have a setup that Just Works? I'd love to find a setup
that works for someone else and just copy it.
Thanks,
sean
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Re: [asterisk-users] AGI == DeadAGI
CHEN XUEQIN wrote: Hi Luki: 于 2010年05月01日 06:03, Luki 写道: The good news is, we run tens of thousands of calls every day through this box and about half of them spit out this warning, but it never caused any problems for over a year. Thus this warning is probably safe to ignore. We run tens of thousands of call every day too. Call is controlled by AGI , and the asterisk version is 1.2.24. I find memory leak in asterisk. After serveral weeks, the memory used by asterisk will reach 1.2 GB or higher. Each time I have to restart to asterisk, and the memory leak will repeat. I wonder if you have the memory leak problem? Which version asterisk you use? Thanks for reply. Regards, Chen Xueqin Did you really mean 1.2.24? or was that a typo? You probably should, at the least up to the last version of 1.2, or, 1.2 Luddites not withstanding, move to a late 1.4 version If all else fails, automatically reboot once a week at some quiet time JMO John Novack -- Dog is my co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI == DeadAGI
On Sat, 1 May 2010, SIP wrote: [snip] We run DeadAGI for a considerable number of calls since it has the ability to run post-hangup cleanup no matter which side hangs up (unlike AGI). [snip] When a channel hangs up, Asterisk sends a SIGHUP signal to the AGI. If the AGI did not establish a handler for the SIGHUP, the AGI exits. If the AGI established a handler, the AGI can choose to ignore the signal or execute appropriate code -- like clean up files, write a CDR to the database, etc. If the AGI is started when the channel is live, you should use agi() and catch signals appropriately. If the AGI is started when the channel is dead, you should use deadagi(). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI == DeadAGI
On 5/2/2010 4:52 PM, Steve Edwards wrote: On Sat, 1 May 2010, SIP wrote: [snip] We run DeadAGI for a considerable number of calls since it has the ability to run post-hangup cleanup no matter which side hangs up (unlike AGI). [snip] When a channel hangs up, Asterisk sends a SIGHUP signal to the AGI. If the AGI did not establish a handler for the SIGHUP, the AGI exits. If the AGI established a handler, the AGI can choose to ignore the signal or execute appropriate code -- like clean up files, write a CDR to the database, etc. If the AGI is started when the channel is live, you should use agi() and catch signals appropriately. If the AGI is started when the channel is dead, you should use deadagi(). Right. That's the way it works in theory, with the nice separation of AGI on live channels and DeadAGI on dead channels. But with our scripts, we use DeadAGI because the channel will redial different gateways after a live connection is made if there's a problem, and we've been unable to figure out how to get that from AGI, since, once the channel is hung up, it won't let us redial again. I'm sure it's a matter of just some little collection of things we're doing wrong, but for the moment, DeadAGI works swimmingly, so we haven't delved too deeply. We've never run into one of the supposed problems with running DeadAGI on a live channel. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calling a RESTful Web service from Dialplan????
Dear All, Last Week i tried and goggling more on how to call RESTful webservice from Dialplan? i found *CURL* function but while i tried to use it ,it 's not supported HTTPS request and we cannot set headers while send a request. also without HTTPS . i get result it will return a string means whole xml,json request is represented in string format, how can i parse that request? my question is that is there any best utility in asterisk that support calling a webservie from Dialplan? i am also comfortable with C, or PERL based AGI. please guide me as i am new to this Webservice part... regards Dhaval -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
