*CLI> jabber show connections
Jabber Users and their status:
[asterisk] r...@gmail.com - Connected
----
Number of users: 1On 1/22/13 2:14 PM, Danny Nicholas wrote:
What about "jabber show channels"? -----Original Message----- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI> core show help gtalk gtalk show channels Show GoogleTalk channels *CLI> gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage="Ohai from Asterisk" timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote:Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -----Original Message----- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote:If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -----Original Message----- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote:Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-20000 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -----Original Message----- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has "CONNECTED" status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote:Do a "netstat -anp" during the call. This will (hopefully) show you where the out of range condition is occurring. -----Original Message----- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI> -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose("Gtalk/+1xxxxxxxxxx-2310", "0, Incoming gtalk from "+1xxxxxxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" <>") in new stack Incoming gtalk from "+xxxxxxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" <> -- Executing [r...@gmail.com@gtalk_incoming:2] Answer("Gtalk/+xxxxxxxxxx-2310", "") in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait("Gtalk/+xxxxxxxxxx-2310", "2") in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial("Gtalk/+xxxxxxxxxx-2310", "SIP/D70") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI> *CLI> -- SIP/D70-00000006 is ringing *CLI> -- SIP/D70-00000006 answered Gtalk/+xxxxxxxxxx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xxxxxxxxxx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote:You are obviously getting the call connected, so the subnet issue ismoot.What this sounds like (pardon the pun) to me is an rtp skip issue. The "working" calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (10000-20000 by default) are open in the firewall. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-CommercialDiscussionSubject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote:On Mon, Jan 21, 2013 at 9:59 PM, Frank <fr...@efirehouse.com <mailto:fr...@efirehouse.com>> wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com <http://voice.google.com> actually resolves to a range of IPaddresses.When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.36 voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.46 voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.33 voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.32 voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.41 voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.38 voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.35 voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.39 voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.40 voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.34 voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.37 (ie 74.125.225.32-41 and 74.125.225.46) Since these are short TTL values (the 300 means 5 minutes) there may be a brief period where your devices and your firewall agree, before one or both change their mind about the IP address behind thathostname.I just tried out of the blue calling from D70 through GoogleVoiceto a cell phone, and it worked. I hung up, redial, and no audio atall.On 1/21/13 10:38 PM, Frank wrote: Greetings all, I was reading the documentation tonight, and decided totryGoogle voice with my asterisk. I was able to setup iksemel, connect to google using jabber,andconnect to google voice using gtalk. Here is my physical configuration: Digium D70 <-- private network 192.168.1.x --> Airport express<-->Internet <--> Asterisk with public IP My asterisk has the following ports open: 5060 tcp/udp from my Airport Express public IP and from voice.google.com <http://voice.google.com> 10,000:20,000 from my Airport Express public IP and from voice.google.com <http://voice.google.com> My issue is that when I place a call with google voice, Ihaveno audio path at all in both way. When a call is received on google voice (and sent to theD70),if I pick up, nothing happen, and the caller still hear the ringingtone.My D70 is setup as follow in the sip.conf: [D70] type=friend nat=yes qualify=yes directmedia=no host=dynamic secret=takapoum disallow=all allow=ulaw context=LocalSets mailbox=D70@default my gtalk.conf is setup as follow: [general] bindaddr=0.0.0.0 allowguest=yes [guest] disallow=all allow=ulaw context=gtalk_incoming connection=asterisk and finally, the interesting parts in my extensions.confaresetup as follow: ;Dialing out on google voice: exten =>_1zxxzxxxxxx,1,Dial(Gtalk/__asterisk/+${EXTEN}@voice.__google.com<mailto:exten...@voice.google.com>)same => n,Hangup() ;Google voice incoming [gtalk_incoming] exten => r...@gmail.com<mailto:r...@gmail.com>,1,Verbose(0,Incoming gtalk from ${CALLERID(all)}) same => n,Answer() same => n,Wait(2) same => n,Dial(SIP/D70) same => Hangup() I would appreciate if anyone could give me a hint abouttheaudio path. This is a project that we I will try to setup in a smallfiredepartment, and before I try it, I would like to make sure thatmyDigium phones will be able to get full audio path behindprivatenetworks. Thanks a ton for the help ! ---- __________________________________________________________________ _ _ _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar everyThurs: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 everyThurs: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
