Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)
Hi Amit, My rtp.conf has the stunaddr listed and icesupport set to yes. It looks like the issue is that the media isn't being sent from 192.168.3.150 to 192.168.3.131 (chrome browser to asteriskrtc.local). When using asteriskrtc.local to originate the call (make a call directly from sipml client to another number on asteriskrtc.local or to a number on another asterisk server) audio flows both ways with no issue, it's just when asteriskgary.local is originating the call that there is no audio flowing from chrome to asteriskrtc.local. I should probably rephrase the above though to say that on tshark I can actually see the packets flowing (tshark host 192.168.3.150): 2.384874 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.384925 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.385060 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:60175 2.385256 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:65021 2.394891 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.415195 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.434063 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.434121 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.434296 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:60175 2.434462 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:65021 2.435083 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.455310 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.475009 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 Thanks again for your time! Kind Regards, Gary Shergill - Original Message - From: Amit Patkar a...@avhan.com To: asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 4:55:57 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Please check rtp.conf Look for stunaddr setting. You can try with google STUN server stunaddr = stun.l.google.com:19302 Thanks Regards, Amit Patkar On 5/21/2014 9:13 PM, Gary Shergill wrote: Hi again, Just noticed this is being sent to the wrong thread... first time using a mailing list and I just replied to the mail sent by the mailing list for Amit's reply. Hope this time it works... Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side (I tested using the SIPml demo site and it worked, then realised I was missing a setting). However, the issue still remains where 1000 can not always hear 6901. As mentioned before, this works only SOMETIMES, and when it does work asteriskgary.local sees RTP packets coming FROM 192.168.3.131 (asteriskrtc.local). Unsure what would be causing this, because it does work sometimes and doesn't at others, with no obvious reason either way. Thanks again. Kind Regards, Gary Shergill - Original Message - From: Gary Shergill gsherg...@gltd.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 3:36:54 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Amit, ICE/STUN is configured correctly. The extension for the webrtc user is defined in sip.conf on the asteriskrtc.local server. The other user is defined in Elastix. I have directmedia=no set for the user on asteriskrtc.local. My exact setup/scenario is below: - asteriskgary.local has a route to dial extensions on my Elastix server. - asteriskgary.local has a route to dial extensions on asteriskrtc.local server. - The call is being originated from asteriskgary.local. The first party is an extension on asteriskgary.local, the destination party is an extension on my Elastix server. What's happening is as follows (this is a reverse of the previous case as 6901 is now dialling 1000): - User on asteriskgary.local places a call to 1000, his number is 6901 - 6901 answers on the web browser and begins to dial 1000 - 1000 answers and the call is established correctly - SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...) - 6901 can NEVER hear 1000 key: 192.168.3.127 - asteriskgary.local 192.168.3.131 - asteriskrtc.local 192.168.3.150 - machine running chrome browser where 6901 is logged on 192.168.3.100 - phone where 1000 is logged on (1000 can hear 6901) RTP TRACE ON asteriskrtc.local Got RTP packet from192.168.3.150:55148 (type 00, seq 014308, ts 2304496631, len 000160)
[asterisk-users] Queue is not working
Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone) and eyeBeam have six line and whenever a call comes into eyeBeam that call reserved by Line 1 suppose to 2nd call will come that call goes to Line 2(same extension used by Line 1) and 3rd call goes to 3rd line and so on. But i want to whenever 2nd call will come that call goes into different extentsion that call never hit into reserved extention. extenstion.conf [Queue_Test] exten = s,1,Answer ; Important, see notes exten = s,2,Queue(Queue_Test|tT|||300) ;dont set n option until really needed exten = s,3,Hangup() queues.conf [Queue_Test] music = default strategy = fewestcalls context = queue-out ; Here we go when the caller presses a single digit, while in the queue timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = yes joinempty = yes member = Sip/4001 member = Sip/4003 member = Sip/4004 member = Sip/4005 member = Sip/4006 member = Sip/4007 Regards Akhilesh Sent from Samsung Mobile-- _ -- 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] Queue is not working
On 22 May 2014 12:42, omakhileshchand omakhileshch...@gmail.com wrote: Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone) and eyeBeam have six line and whenever a call comes into eyeBeam that call reserved by Line 1 suppose to 2nd call will come that call goes to Line 2(same extension used by Line 1) and 3rd call goes to 3rd line and so on. But i want to whenever 2nd call will come that call goes into different extentsion that call never hit into reserved extention. extenstion.conf [Queue_Test] exten = s,1,Answer ; Important, see notes exten = s,2,Queue(Queue_Test|tT|||300) ;dont set n option until really needed exten = s,3,Hangup() queues.conf [Queue_Test] music = default strategy = fewestcalls context = queue-out ; Here we go when the caller presses a single digit, while in the queue timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = yes joinempty = yes member = Sip/4001 member = Sip/4003 member = Sip/4004 member = Sip/4005 member = Sip/4006 member = Sip/4007 Regards Akhilesh In your sip.conf have you got callcounter = yes set? What stats is queue show Queue_Test showing at various times? (this will give you an indication of how many calls each member has taken) What happens when you choose rrmemory as the stratergy? Have you read and fully understood the joinempty parameter? Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Queue is not working
I would research the ringinuse option as well. On 22 May 2014 13:42, omakhileshchand omakhileshch...@gmail.com wrote: Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone) and eyeBeam have six line and whenever a call comes into eyeBeam that call reserved by Line 1 suppose to 2nd call will come that call goes to Line 2(same extension used by Line 1) and 3rd call goes to 3rd line and so on. But i want to whenever 2nd call will come that call goes into different extentsion that call never hit into reserved extention. extenstion.conf [Queue_Test] exten = s,1,Answer ; Important, see notes exten = s,2,Queue(Queue_Test|tT|||300) ;dont set n option until really needed exten = s,3,Hangup() queues.conf [Queue_Test] music = default strategy = fewestcalls context = queue-out ; Here we go when the caller presses a single digit, while in the queue timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = yes joinempty = yes member = Sip/4001 member = Sip/4003 member = Sip/4004 member = Sip/4005 member = Sip/4006 member = Sip/4007 Regards Akhilesh Sent from Samsung Mobile -- _ -- 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
[asterisk-users] Interesting new hack attack
In the past little while, we've seen a wave of attacks on asterisk, via the provisioning. It goes something like this: A. scan for IP phones on the internet, either via spotting something on port 5060, or via the port 80 web interface for the phone. Or, use web sites that scan the internet, and classify the machines, to make your work shorter. B. Once you get into the web GUI, get the URL for provisioning. I haven't checked yet... do any phones actually allow you to set this, or do any display the current value? And, finally, how many phones publish their own MAC address in the GUI? Or, can you suck this out of the returned IP packets? C. Given the URL and the mac, fetch the phones provisioning info, including it's sip account info. Use to best advantage. D. Going further, set up a brute-force probe algorithm, to probe all possible mac addresses for a given phone manufacturer, via http requests. After all, those provisioning web servers are fast and efficient, aren't they? Collect all possible mac addresses and grab the provisioning, and now you have a LOT of sip accounts. Use to best advantage. And, professional hacking organizations seem to also follow these rules: a. wait several months for any history of the above activities to roll off the log files. Treat your phone systems like fine wine vintage. b. Use multiple (hundreds/thousands) of machines scattered over the earth to carry out the above probes, and also to use the accounts for generating international calls. In general, using the SIP account info gleaned from these kinds of efforts is a bit problematic. You see, to effectively use your phone system to place calls, they will have to set up their own phone system to act like a phone, and register to the phone system, and then initiate calls. Trouble is, your phone is usually already registered, but can be bumped off. Your phone will re-register at intervals and bump the hackers, who will again register and bump your phone. This little game of king of the hill may show up in your Asterisk logs. So, these defenses can be employed to stop/ameliorate such hacking efforts: 1. Keep your phones behind a firewall. Travellers, beware! Never leave the default login info of the phone at default! 2. Never use the default provisioning URL for the phone, with it's default URL or password. 3. Use fail2ban, ossec, whatever to stymie any brute force mac address searches. 4. Use your firewalls to restrict IP's that can access web, ftp, etc, for provisioning to just those IP's needed to allow your phones to provision. 5. Keep your logs for a couple years. 6. Change your phone SIP acct passwords now, if you haven't implemented the above precautions yet. If I missed a previous post on this, forgive me. Just thought you-all might appreciate a heads-up. murf -- Steve Murphy ParseTree Corporation 57 Lane 17 Cody, WY 82414 ✉ murf at parsetree dot com ☎ 307-899-5535 -- _ -- 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] Interesting new hack attack
On 5/22/2014 12:41 PM, Steve Murphy wrote: So, these defenses can be employed to stop/ameliorate such hacking efforts: 1. Keep your phones behind a firewall. Travellers, beware! Never leave the default login info of the phone at default! 2. Never use the default provisioning URL for the phone, with it's default URL or password. 3. Use fail2ban, ossec, whatever to stymie any brute force mac address searches. 4. Use your firewalls to restrict IP's that can access web, ftp, etc, for provisioning to just those IP's needed to allow your phones to provision. 5. Keep your logs for a couple years. 6. Change your phone SIP acct passwords now, if you haven't implemented the above precautions yet. If I missed a previous post on this, forgive me. Just thought you-all might appreciate a heads-up. Encrypt your provisioning system if the phone supports it. I had a cable/voip service provider who HTTPS provisioned by MAC without encryption and the provisioning URL was stored, unlocked, in the ATA. Had I been slightly more nefarious, I could have walked the the provisioning tree nice and slow and easily grabbed everyone's SIP credentials in the clear. No hacking or cracking was involved. The ATA doubled as the NAT router they handed out and gave the admin password out freely. -- _ -- 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] FollowMe reinvites
For a sip-only application, what exactly is required to ensure that calls completed via followme are reinvited? Can it at all? The code after outbound = findmeexec(targs, chan) calls ast_bridge_ call(). I don't see anything there which can cause a reinvite, yes? When the same peer is used for both the incoming and outgoing legs, it is a bit of a waste to proxy the rtp. And even when the legs are associated with different remotes, I'd prefer to proxy only when NATs a/or v4-v6 gatewaying are involved. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] issue installing voicemail imap support: imap_tk module missing
On Wed, 21 May 2014 23:09:28 +0200 Bart Remmerie remme...@gmail.com wrote: configure: *** The IMAP_TK installation appears to be missing or broken. [...] These are the steps I followed: sudo apt-get install libssl-dev libpam0g-dev cd ~/src/asterisk-complete mkdir third party cd third party wget ftp://ftp.cac.washington.edu/mail/imap.tar.Z $ tar zxvf imap.tar.Z cd imap-2007e make lnp EXTRACFLAGS=-fPIC -I/usr/include/openssl IP6=4 cd ~/src/asterisk-complete/asterisk/11 ./configure --with-imap=~/src/asterisk-complete/thirdparty/imap-2007e/ In your mkdir and cd lines, you have a space between third and party. That would make two directories, and then cd into 'third' (probably). Then the path you passed to configure didn't exist. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] maxsecs not working
Hello, We have servers running Asterisk 1.8.20.1 and 11.7.0, and on both setting maxsecs in voicemail.conf doesn't seem to have any effect. A voicemail keeps recording after the specified time, and when the caller hangs up the voicemail is saved in the mailbox. Are we doing something really silly? Here's the voicemail.conf. We have tried 'voicemail reload' and restarting Asterisk to make it take effect. [general] format = wav mailcmd = /bin/true review = no ; Maximum length of a voicemail message in seconds maxsecs=180 [zonemessages] #include /etc/asterisk/voicemail_zonemessages.conf [default] Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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
