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Re: [Openbts-discuss] Making phone calls in asterisk
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From: Kurtis H. <khe...@cs...> - 2013-04-22 02:45:58
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It's sorta the error log. If you don't know what it is, it's going to be hard to get you to generate it, I think. Maybe your OpenBTS log will be sufficient. On Sun, Apr 21, 2013 at 3:33 PM, Pat Kennedy <pat...@gm...> wrote: > Kurtis, > > Can you elaborate what you mean by the stack dump? > > Pat > > On Sun, Apr 21, 2013 at 12:35 AM, Kurtis Heimerl <khe...@cs... > > wrote: > >> Comments in line: >> >> >> On Sat, Apr 20, 2013 at 10:18 AM, Timothy < >> blu...@gm...> wrote: >> >>> Hey there Kurtis (and the rest of the mailing list), >>> >>> Thus far Pat and I have been able to establish: >>> >>> - SMS messages between cellphones. >>> - SMS messages from the Asterisk Console to individual phones. >>> - Cellphone to 'Application' connections (e.g. Playback, SayDigits) >>> - Asterisk Console to extra-network calls through our Skype Connect >>> Trunk (e.g. I can call landlines) >>> - Incoming calls from outside out network into Asterisk via Skype >>> Connect. (I currently have it read back the caller's phone number) >>> >>> So, we know that we have at least primitive connectivity between >>> OpenBTS, Asterisk, and Skype Connect. But what we don't have is phone to >>> phone connectivity. Thus far we have not been able to make cell to cell >>> calls, cell to outgoing call (Skype), nor incoming call (Skype) to cell. >>> 1) One a side note, when we go to dial some numbers on the cellphones we >>> see a message on the screens saying 'This number is unregister, and unable >>> to switch'. Is that an indicator of a specific problem? >>> >>> Pat and I also find it a bit odd that the "sip show peers" no longer >>> shows the cellphones as having addresses. Earlier, one of the phones had a >>> registered listing of IP address and port, but now neither of them show up >>> as being listed. >>> >> >> It depends on your setup. If you followed the wiki, you have OpenBTS >> pointing the registration traffic at sipauthserve, not Asterisk. As such, >> asterisk won't know anything about the phones. >> >> >>> >>> 2) You mentioned 'do[ing] the Asterisk install' from the wiki. Are you >>> talking about simply reinstalling Asterisk given a set of instructions? If >>> so, could you specify the link to the instructions? I'm having difficulty >>> finding them. Also, are you saying that something was not installed/linked >>> correctly with our previous install of Asterisk? >>> >> >> https://wush.net/trac/rangepublic/wiki/asteriskConfig pointing to >> http://www.voip-info.org/wiki/view/Asterisk+RealTime >> >> >>> >>> 3) And on your suggestion of hard coding the 'host' and 'port', we >>> attempted that per your suggestion. While it does cause the 'sip show >>> peers' to indicate the set values for ip address and port, OpenBTS appears >>> to crash each time we attempt to then call one of the phones. >>> >> >> That's not right. Can you provide a stack dump of that behavior? >> >> >>> >>> 4) I have been working to find the log that is outputted from OpenBTS, >>> but have had no luck thus far. It just sits there silent, seemingly keeping >>> whatever problems it has to itself, until it crashes or we close it. What >>> is the standard method of catching the logs of OpenBTS, sipauthserv, and >>> smqueue for the public (non-commericial) release of OpenBTS? >>> >>> >> It should log to either /var/log/OpenBTS.log or /var/log/syslog if you >> have not configured syslogd. >> >> >>> 5) Is there a nice way to shut down OpenBTS and transceiver? Thus far >>> we've either been using 'exit' from the OpenBTS CLI, but we then have to >>> "sudo pkill transceiver" to allow for OpenBTS to be started again. >>> >>> >> That's a bug, if you actually just cntl-c OpenBTS it'll clean up right. >> >> >>> If any additional information would help in diagnosing our issue(s), we >>> would be glad to provide it. >>> >>> Thank you in advance, >>> Timothy >>> >>> >>> On Apr 19, 2013, at 6:34 PM, Kurtis Heimerl wrote: >>> >>> Yeah, I'm pretty sure that the users are "offline" as you don't register >>> them with Asterisk. You could do the asterisk install on the wiki (which >>> will resolve this) or, I think, you can just dial them directly by adding >>> the IP (127.0.0.1) and port (5062) into the dial command. Then I think >>> Asterisk won't care that they're not online. >>> >>> >>> On Fri, Apr 19, 2013 at 3:27 PM, Pat Kennedy <pat...@gm...>wrote: >>> >>>> OpenBTS-discuss, >>>> >>>> We are trying to use OpenBTS and asterisk work so two phones within our >>>> network can call each other. >>>> >>>> This is what we added to sip.conf: >>>> >>>> [IMSI001011052191591] >>>> callerid=8885555 >>>> canreinvite=no >>>> type=friend >>>> context=blackphone >>>> allow=gsm >>>> host=dynamic >>>> >>>> >>>> [IMSI001014027170835] >>>> callerid=9876543 >>>> canreinvite=no >>>> type=friend >>>> context=redphone >>>> allow=gsm >>>> host=dynamic >>>> >>>> This is what we added to extensions.conf: >>>> >>>> [macro-dialGSM] >>>> exten => s,1,Dial(SIP/${ARG1}) >>>> exten => s,2,Goto(s-${DIALSTATUS},1) >>>> exten => s-CANCEL,1,Hangup >>>> exten => s-NOANSWER,1,Hangup >>>> exten => s-BUSY,1,Busy(30) >>>> exten => s-CONGESTION,1,Congestion(30) >>>> exten => s-CHANUNAVAIL,1,playback(ss-noservice) >>>> exten => s-CANCEL,1,Hangup >>>> >>>> [redphone] >>>> >>>> exten => 8885555,1,Macro(dialGSM,IMSI001011052191591) >>>> >>>> [blackphone] >>>> >>>> >>>> exten => 9876543,1,Macro(dialGSM,IMSI001014027170835) >>>> >>>> >>>> sip show peers followed by an attempt to call (verbosity is set to 3): >>>> >>>> openbts*CLI> sip show peers >>>> Name/username Host Dyn >>>> Forcerport ACL Port Status >>>> IMSI001011052191591 (Unspecified) D >>>> N 0 Unmonitored >>>> IMSI001014027170835 (Unspecified) D >>>> N 0 Unmonitored >>>> skype/99051000208998 63.209.144.201 >>>> N 5060 OK (44 ms) >>>> 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 2 >>>> offline] >>>> == Using SIP RTP CoS mark 5 >>>> -- Executing [8885555@redphone:1] >>>> Macro("SIP/IMSI001014027170835-00000000", "dialGSM,IMSI001011052191591") in >>>> new stack >>>> -- Executing [s@macro-dialGSM:1] >>>> Dial("SIP/IMSI001014027170835-00000000", "SIP/IMSI001011052191591") in new >>>> stack >>>> [Apr 19 18:22:31] WARNING[3829]: app_dial.c:2274 dial_exec_full: Unable >>>> to create channel of type 'SIP' (cause 20 - Unknown) >>>> == Everyone is busy/congested at this time (1:0/0/1) >>>> -- Executing [s@macro-dialGSM:2] >>>> Goto("SIP/IMSI001014027170835-00000000", "s-CHANUNAVAIL,1") in new stack >>>> -- Goto (macro-dialGSM,s-CHANUNAVAIL,1) >>>> -- Executing [s-CHANUNAVAIL@macro-dialGSM:1] >>>> Playback("SIP/IMSI001014027170835-00000000", "ss-noservice") in new stack >>>> -- <SIP/IMSI001014027170835-00000000> Playing 'ss-noservice.gsm' >>>> (language 'en') >>>> -- Auto fallthrough, channel 'SIP/IMSI001014027170835-00000000' >>>> status is 'CHANUNAVAIL' >>>> >>>> >>>> >>>> Any help would be greatly appreciated. I understand somewhat that >>>> asterisk is looking to the above contexts when I try to make a call. I >>>> suspect it has something to do with registration but I am not sure what to >>>> do about it. >>>> >>>> Pat >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Precog is a next-generation analytics platform capable of advanced >>>> analytics on semi-structured data. The platform includes APIs for >>>> building >>>> apps and a phenomenal toolset for data science. Developers can use >>>> our toolset for easy data analysis & visualization. Get a free account! >>>> http://www2.precog.com/precogplatform/slashdotnewsletter >>>> _______________________________________________ >>>> Openbts-discuss mailing list >>>> Ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/openbts-discuss >>>> >>>> >>> >>> ------------------------------------------------------------------------------ >>> Precog is a next-generation analytics platform capable of advanced >>> analytics on semi-structured data. The platform includes APIs for >>> building >>> apps and a phenomenal toolset for data science. Developers can use >>> our toolset for easy data analysis & visualization. Get a free account! >>> >>> http://www2.precog.com/precogplatform/slashdotnewsletter_______________________________________________ >>> Openbts-discuss mailing list >>> Ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/openbts-discuss >>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Precog is a next-generation analytics platform capable of advanced >>> analytics on semi-structured data. The platform includes APIs for >>> building >>> apps and a phenomenal toolset for data science. Developers can use >>> our toolset for easy data analysis & visualization. Get a free account! >>> http://www2.precog.com/precogplatform/slashdotnewsletter >>> _______________________________________________ >>> Openbts-discuss mailing list >>> Ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/openbts-discuss >>> >>> >> >> >> ------------------------------------------------------------------------------ >> Precog is a next-generation analytics platform capable of advanced >> analytics on semi-structured data. The platform includes APIs for building >> apps and a phenomenal toolset for data science. Developers can use >> our toolset for easy data analysis & visualization. Get a free account! >> http://www2.precog.com/precogplatform/slashdotnewsletter >> _______________________________________________ >> Openbts-discuss mailing list >> Ope...@li... >> https://lists.sourceforge.net/lists/listinfo/openbts-discuss >> >> > |
